diff options
Diffstat (limited to 'sound')
272 files changed, 15912 insertions, 5898 deletions
diff --git a/sound/Kconfig b/sound/Kconfig index 200aca1..1eceb85 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -60,6 +60,8 @@ source "sound/aoa/Kconfig" source "sound/arm/Kconfig" +source "sound/atmel/Kconfig" + source "sound/spi/Kconfig" source "sound/mips/Kconfig" diff --git a/sound/Makefile b/sound/Makefile index c76d707..ec467de 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -6,7 +6,7 @@ obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \ - sparc/ spi/ parisc/ pcmcia/ mips/ soc/ + sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff --git a/sound/aoa/aoa-gpio.h b/sound/aoa/aoa-gpio.h index ee64f5d..6065b03 100644 --- a/sound/aoa/aoa-gpio.h +++ b/sound/aoa/aoa-gpio.h @@ -34,10 +34,12 @@ struct gpio_methods { void (*set_headphone)(struct gpio_runtime *rt, int on); void (*set_speakers)(struct gpio_runtime *rt, int on); void (*set_lineout)(struct gpio_runtime *rt, int on); + void (*set_master)(struct gpio_runtime *rt, int on); int (*get_headphone)(struct gpio_runtime *rt); int (*get_speakers)(struct gpio_runtime *rt); int (*get_lineout)(struct gpio_runtime *rt); + int (*get_master)(struct gpio_runtime *rt); void (*set_hw_reset)(struct gpio_runtime *rt, int on); diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index c93ad5d..de8e03a 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -14,7 +14,7 @@ #include <linux/interrupt.h> #include "../aoa.h" -/* TODO: these are 20 global variables +/* TODO: these are lots of global variables * that aren't used on most machines... * Move them into a dynamically allocated * structure and use that. @@ -23,6 +23,7 @@ /* these are the GPIO numbers (register addresses as offsets into * the GPIO space) */ static int headphone_mute_gpio; +static int master_mute_gpio; static int amp_mute_gpio; static int lineout_mute_gpio; static int hw_reset_gpio; @@ -32,6 +33,7 @@ static int linein_detect_gpio; /* see the SWITCH_GPIO macro */ static int headphone_mute_gpio_activestate; +static int master_mute_gpio_activestate; static int amp_mute_gpio_activestate; static int lineout_mute_gpio_activestate; static int hw_reset_gpio_activestate; @@ -156,6 +158,7 @@ static int ftr_gpio_get_##name(struct gpio_runtime *rt) \ FTR_GPIO(headphone, 0); FTR_GPIO(amp, 1); FTR_GPIO(lineout, 2); +FTR_GPIO(master, 3); static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on) { @@ -172,6 +175,8 @@ static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on) hw_reset_gpio, v); } +static struct gpio_methods methods; + static void ftr_gpio_all_amps_off(struct gpio_runtime *rt) { int saved; @@ -181,6 +186,8 @@ static void ftr_gpio_all_amps_off(struct gpio_runtime *rt) ftr_gpio_set_headphone(rt, 0); ftr_gpio_set_amp(rt, 0); ftr_gpio_set_lineout(rt, 0); + if (methods.set_master) + ftr_gpio_set_master(rt, 0); rt->implementation_private = saved; } @@ -193,6 +200,8 @@ static void ftr_gpio_all_amps_restore(struct gpio_runtime *rt) ftr_gpio_set_headphone(rt, (s>>0)&1); ftr_gpio_set_amp(rt, (s>>1)&1); ftr_gpio_set_lineout(rt, (s>>2)&1); + if (methods.set_master) + ftr_gpio_set_master(rt, (s>>3)&1); } static void ftr_handle_notify(struct work_struct *work) @@ -231,6 +240,12 @@ static void ftr_gpio_init(struct gpio_runtime *rt) get_gpio("hw-reset", "audio-hw-reset", &hw_reset_gpio, &hw_reset_gpio_activestate); + if (get_gpio("master-mute", NULL, + &master_mute_gpio, + &master_mute_gpio_activestate)) { + methods.set_master = ftr_gpio_set_master; + methods.get_master = ftr_gpio_get_master; + } headphone_detect_node = get_gpio("headphone-detect", NULL, &headphone_detect_gpio, diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index ad60f5d..fbf5c93 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -1,16 +1,14 @@ /* - * Apple Onboard Audio driver -- layout fabric + * Apple Onboard Audio driver -- layout/machine id fabric * - * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * Copyright 2006-2008 Johannes Berg <johannes@sipsolutions.net> * * GPL v2, can be found in COPYING. * * - * This fabric module looks for sound codecs - * based on the layout-id property in the device tree. - * + * This fabric module looks for sound codecs based on the + * layout-id or device-id property in the device tree. */ - #include <asm/prom.h> #include <linux/list.h> #include <linux/module.h> @@ -63,7 +61,7 @@ struct codec_connect_info { #define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF (1<<0) struct layout { - unsigned int layout_id; + unsigned int layout_id, device_id; struct codec_connect_info codecs[MAX_CODECS_PER_BUS]; int flags; @@ -111,6 +109,10 @@ MODULE_ALIAS("sound-layout-96"); MODULE_ALIAS("sound-layout-98"); MODULE_ALIAS("sound-layout-100"); +MODULE_ALIAS("aoa-device-id-14"); +MODULE_ALIAS("aoa-device-id-22"); +MODULE_ALIAS("aoa-device-id-35"); + /* onyx with all but microphone connected */ static struct codec_connection onyx_connections_nomic[] = { { @@ -518,6 +520,27 @@ static struct layout layouts[] = { .connections = onyx_connections_noheadphones, }, }, + /* PowerMac3,4 */ + { .device_id = 14, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_noline, + }, + }, + /* PowerMac3,6 */ + { .device_id = 22, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, + /* PowerBook5,2 */ + { .device_id = 35, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, {} }; @@ -526,7 +549,7 @@ static struct layout *find_layout_by_id(unsigned int id) struct layout *l; l = layouts; - while (l->layout_id) { + while (l->codecs[0].name) { if (l->layout_id == id) return l; l++; @@ -534,6 +557,19 @@ static struct layout *find_layout_by_id(unsigned int id) return NULL; } +static struct layout *find_layout_by_device(unsigned int id) +{ + struct layout *l; + + l = layouts; + while (l->codecs[0].name) { + if (l->device_id == id) + return l; + l++; + } + return NULL; +} + static void use_layout(struct layout *l) { int i; @@ -564,6 +600,7 @@ struct layout_dev { struct snd_kcontrol *headphone_ctrl; struct snd_kcontrol *lineout_ctrl; struct snd_kcontrol *speaker_ctrl; + struct snd_kcontrol *master_ctrl; struct snd_kcontrol *headphone_detected_ctrl; struct snd_kcontrol *lineout_detected_ctrl; @@ -615,6 +652,7 @@ static struct snd_kcontrol_new n##_ctl = { \ AMP_CONTROL(headphone, "Headphone Switch"); AMP_CONTROL(speakers, "Speakers Switch"); AMP_CONTROL(lineout, "Line-Out Switch"); +AMP_CONTROL(master, "Master Switch"); static int detect_choice_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -855,6 +893,11 @@ static void layout_attached_codec(struct aoa_codec *codec) lineout = codec->gpio->methods->get_detect(codec->gpio, AOA_NOTIFY_LINE_OUT); + if (codec->gpio->methods->set_master) { + ctl = snd_ctl_new1(&master_ctl, codec->gpio); + ldev->master_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } while (cc->connected) { if (cc->connected & CC_SPEAKERS) { if (headphones <= 0 && lineout <= 0) @@ -938,8 +981,8 @@ static struct aoa_fabric layout_fabric = { static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) { struct device_node *sound = NULL; - const unsigned int *layout_id; - struct layout *layout; + const unsigned int *id; + struct layout *layout = NULL; struct layout_dev *ldev = NULL; int err; @@ -952,15 +995,18 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) if (sound->type && strcasecmp(sound->type, "soundchip") == 0) break; } - if (!sound) return -ENODEV; + if (!sound) + return -ENODEV; - layout_id = of_get_property(sound, "layout-id", NULL); - if (!layout_id) - goto outnodev; - printk(KERN_INFO "snd-aoa-fabric-layout: found bus with layout %d\n", - *layout_id); + id = of_get_property(sound, "layout-id", NULL); + if (id) { + layout = find_layout_by_id(*id); + } else { + id = of_get_property(sound, "device-id", NULL); + if (id) + layout = find_layout_by_device(*id); + } - layout = find_layout_by_id(*layout_id); if (!layout) { printk(KERN_ERR "snd-aoa-fabric-layout: unknown layout\n"); goto outnodev; @@ -976,6 +1022,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) ldev->layout = layout; ldev->gpio.node = sound->parent; switch (layout->layout_id) { + case 0: /* anything with device_id, not layout_id */ case 41: /* that unknown machine no one seems to have */ case 51: /* PowerBook5,4 */ case 58: /* Mac Mini */ diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index be468edf..418c84c 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -1,7 +1,7 @@ /* * i2sbus driver * - * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * Copyright 2006-2008 Johannes Berg <johannes@sipsolutions.net> * * GPL v2, can be found in COPYING. */ @@ -186,13 +186,25 @@ static int i2sbus_add_dev(struct macio_dev *macio, } } if (i == 1) { - const u32 *layout_id = - of_get_property(sound, "layout-id", NULL); - if (layout_id) { - layout = *layout_id; + const u32 *id = of_get_property(sound, "layout-id", NULL); + + if (id) { + layout = *id; snprintf(dev->sound.modalias, 32, "sound-layout-%d", layout); ok = 1; + } else { + id = of_get_property(sound, "device-id", NULL); + /* + * We probably cannot handle all device-id machines, + * so restrict to those we do handle for now. + */ + if (id && (*id == 22 || *id == 14 || *id == 35)) { + snprintf(dev->sound.modalias, 32, + "aoa-device-id-%d", *id); + ok = 1; + layout = -1; + } } } /* for the time being, until we can handle non-layout-id diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index f8e6de4..885683a 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -11,17 +11,6 @@ menuconfig SND_ARM if SND_ARM -config SND_SA11XX_UDA1341 - tristate "SA11xx UDA1341TS driver (iPaq H3600)" - depends on ARCH_SA1100 && L3 - select SND_PCM - help - Say Y here if you have a Compaq iPaq H3x00 handheld computer - and want to use its Philips UDA 1341 audio chip. - - To compile this driver as a module, choose M here: the module - will be called snd-sa11xx-uda1341. - config SND_ARMAACI tristate "ARM PrimeCell PL041 AC Link support" depends on ARM_AMBA diff --git a/sound/arm/Makefile b/sound/arm/Makefile index 2054de1..5a549ed 100644 --- a/sound/arm/Makefile +++ b/sound/arm/Makefile @@ -2,9 +2,6 @@ # Makefile for ALSA # -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-sa11xx-uda1341.o -snd-sa11xx-uda1341-objs := sa11xx-uda1341.o - obj-$(CONFIG_SND_ARMAACI) += snd-aaci.o snd-aaci-objs := aaci.o devdma.o diff --git a/sound/arm/aaci.c b/sound/arm/aaci.c index 7d39aac..7fbd68f 100644 --- a/sound/arm/aaci.c +++ b/sound/arm/aaci.c @@ -90,7 +90,7 @@ static void aaci_ac97_write(struct snd_ac97 *ac97, unsigned short reg, */ do { v = readl(aaci->base + AACI_SLFR); - } while ((v & (SLFR_1TXB|SLFR_2TXB)) && timeout--); + } while ((v & (SLFR_1TXB|SLFR_2TXB)) && --timeout); if (!timeout) dev_err(&aaci->dev->dev, @@ -126,7 +126,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) */ do { v = readl(aaci->base + AACI_SLFR); - } while ((v & SLFR_1TXB) && timeout--); + } while ((v & SLFR_1TXB) && --timeout); if (!timeout) { dev_err(&aaci->dev->dev, "timeout on slot 1 TX busy\n"); @@ -147,7 +147,7 @@ static unsigned short aaci_ac97_read(struct snd_ac97 *ac97, unsigned short reg) do { cond_resched(); v = readl(aaci->base + AACI_SLFR) & (SLFR_1RXV|SLFR_2RXV); - } while ((v != (SLFR_1RXV|SLFR_2RXV)) && timeout--); + } while ((v != (SLFR_1RXV|SLFR_2RXV)) && --timeout); if (!timeout) { dev_err(&aaci->dev->dev, "timeout on RX valid\n"); diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 71bef45..0afd1a8 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -21,7 +21,6 @@ #include <sound/pxa2xx-lib.h> #include <asm/irq.h> -#include <mach/hardware.h> #include <mach/regs-ac97.h> #include <mach/pxa2xx-gpio.h> #include <mach/audio.h> diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 7ed100c..c570ebd 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -20,8 +20,6 @@ #include <sound/initval.h> #include <sound/pxa2xx-lib.h> -#include <mach/hardware.h> -#include <mach/pxa-regs.h> #include <mach/regs-ac97.h> #include <mach/audio.h> diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 75a0d74..108b643 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -12,8 +12,7 @@ #include <sound/pcm_params.h> #include <sound/pxa2xx-lib.h> -#include <asm/dma.h> -#include <mach/pxa-regs.h> +#include <mach/dma.h> #include "pxa2xx-pcm.h" diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c deleted file mode 100644 index 51d708c..0000000 --- a/sound/arm/sa11xx-uda1341.c +++ /dev/null @@ -1,984 +0,0 @@ -/* - * Driver for Philips UDA1341TS on Compaq iPAQ H3600 soundcard - * Copyright (C) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek initial release - based on h3600-uda1341.c from OSS - * 2002-03-20 Tomas Kasparek playback over ALSA is working - * 2002-03-28 Tomas Kasparek playback over OSS emulation is working - * 2002-03-29 Tomas Kasparek basic capture is working (native ALSA) - * 2002-03-29 Tomas Kasparek capture is working (OSS emulation) - * 2002-04-04 Tomas Kasparek better rates handling (allow non-standard rates) - * 2003-02-14 Brian Avery fixed full duplex mode, other updates - * 2003-02-20 Tomas Kasparek merged updates by Brian (except HAL) - * 2003-04-19 Jaroslav Kysela recoded DMA stuff to follow 2.4.18rmk3-hh24 kernel - * working suspend and resume - * 2003-04-28 Tomas Kasparek updated work by Jaroslav to compile it under 2.5.x again - * merged HAL layer (patches from Brian) - */ - -/*************************************************************************************************** -* -* To understand what Alsa Drivers should be doing look at "Writing an Alsa Driver" by Takashi Iwai -* available in the Alsa doc section on the website -* -* A few notes to make things clearer. The UDA1341 is hooked up to Serial port 4 on the SA1100. -* We are using SSP mode to talk to the UDA1341. The UDA1341 bit & wordselect clocks are generated -* by this UART. Unfortunately, the clock only runs if the transmit buffer has something in it. -* So, if we are just recording, we feed the transmit DMA stream a bunch of 0x0000 so that the -* transmit buffer is full and the clock keeps going. The zeroes come from FLUSH_BASE_PHYS which -* is a mem loc that always decodes to 0's w/ no off chip access. -* -* Some alsa terminology: -* frame => num_channels * sample_size e.g stereo 16 bit is 2 * 16 = 32 bytes -* period => the least number of bytes that will generate an interrupt e.g. we have a 1024 byte -* buffer and 4 periods in the runtime structure this means we'll get an int every 256 -* bytes or 4 times per buffer. -* A number of the sizes are in frames rather than bytes, use frames_to_bytes and -* bytes_to_frames to convert. The easiest way to tell the units is to look at the -* type i.e. runtime-> buffer_size is in frames and its type is snd_pcm_uframes_t -* -* Notes about the pointer fxn: -* The pointer fxn needs to return the offset into the dma buffer in frames. -* Interrupts must be blocked before calling the dma_get_pos fxn to avoid race with interrupts. -* -* Notes about pause/resume -* Implementing this would be complicated so it's skipped. The problem case is: -* A full duplex connection is going, then play is paused. At this point you need to start xmitting -* 0's to keep the record active which means you cant just freeze the dma and resume it later you'd -* need to save off the dma info, and restore it properly on a resume. Yeach! -* -* Notes about transfer methods: -* The async write calls fail. I probably need to implement something else to support them? -* -***************************************************************************************************/ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/err.h> -#include <linux/platform_device.h> -#include <linux/errno.h> -#include <linux/ioctl.h> -#include <linux/delay.h> -#include <linux/slab.h> - -#ifdef CONFIG_PM -#include <linux/pm.h> -#endif - -#include <mach/hardware.h> -#include <mach/h3600.h> -#include <asm/mach-types.h> -#include <asm/dma.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/initval.h> - -#include <linux/l3/l3.h> - -#undef DEBUG_MODE -#undef DEBUG_FUNCTION_NAMES -#include <sound/uda1341.h> - -/* - * FIXME: Is this enough as autodetection of 2.4.X-rmkY-hhZ kernels? - * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this - * module for Familiar 0.6.1 - */ - -/* {{{ Type definitions */ - -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>"); -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("SA1100/SA1111 + UDA1341TS driver for ALSA"); -MODULE_SUPPORTED_DEVICE("{{UDA1341,iPAQ H3600 UDA1341TS}}"); - -static char *id; /* ID for this card */ - -module_param(id, charp, 0444); -MODULE_PARM_DESC(id, "ID string for SA1100/SA1111 + UDA1341TS soundcard."); - -struct audio_stream { - char *id; /* identification string */ - int stream_id; /* numeric identification */ - dma_device_t dma_dev; /* device identifier for DMA */ -#ifdef HH_VERSION - dmach_t dmach; /* dma channel identification */ -#else - dma_regs_t *dma_regs; /* points to our DMA registers */ -#endif - unsigned int active:1; /* we are using this stream for transfer now */ - int period; /* current transfer period */ - int periods; /* current count of periods registerd in the DMA engine */ - int tx_spin; /* are we recoding - flag used to do DMA trans. for sync */ - unsigned int old_offset; - spinlock_t dma_lock; /* for locking in DMA operations (see dma-sa1100.c in the kernel) */ - struct snd_pcm_substream *stream; -}; - -struct sa11xx_uda1341 { - struct snd_card *card; - struct l3_client *uda1341; - struct snd_pcm *pcm; - long samplerate; - struct audio_stream s[2]; /* playback & capture */ -}; - -static unsigned int rates[] = { - 8000, 10666, 10985, 14647, - 16000, 21970, 22050, 24000, - 29400, 32000, 44100, 48000, -}; - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = ARRAY_SIZE(rates), - .list = rates, - .mask = 0, -}; - -static struct platform_device *device; - -/* }}} */ - -/* {{{ Clock and sample rate stuff */ - -/* - * Stop-gap solution until rest of hh.org HAL stuff is merged. - */ -#define GPIO_H3600_CLK_SET0 GPIO_GPIO (12) -#define GPIO_H3600_CLK_SET1 GPIO_GPIO (13) - -#ifdef CONFIG_SA1100_H3XXX -#define clr_sa11xx_uda1341_egpio(x) clr_h3600_egpio(x) -#define set_sa11xx_uda1341_egpio(x) set_h3600_egpio(x) -#else -#error This driver could serve H3x00 handhelds only! -#endif - -static void sa11xx_uda1341_set_audio_clock(long val) -{ - switch (val) { - case 24000: case 32000: case 48000: /* 00: 12.288 MHz */ - GPCR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; - break; - - case 22050: case 29400: case 44100: /* 01: 11.2896 MHz */ - GPSR = GPIO_H3600_CLK_SET0; - GPCR = GPIO_H3600_CLK_SET1; - break; - - case 8000: case 10666: case 16000: /* 10: 4.096 MHz */ - GPCR = GPIO_H3600_CLK_SET0; - GPSR = GPIO_H3600_CLK_SET1; - break; - - case 10985: case 14647: case 21970: /* 11: 5.6245 MHz */ - GPSR = GPIO_H3600_CLK_SET0 | GPIO_H3600_CLK_SET1; - break; - } -} - -static void sa11xx_uda1341_set_samplerate(struct sa11xx_uda1341 *sa11xx_uda1341, long rate) -{ - int clk_div = 0; - int clk=0; - - /* We don't want to mess with clocks when frames are in flight */ - Ser4SSCR0 &= ~SSCR0_SSE; - /* wait for any frame to complete */ - udelay(125); - - /* - * We have the following clock sources: - * 4.096 MHz, 5.6245 MHz, 11.2896 MHz, 12.288 MHz - * Those can be divided either by 256, 384 or 512. - * This makes up 12 combinations for the following samplerates... - */ - if (rate >= 48000) - rate = 48000; - else if (rate >= 44100) - rate = 44100; - else if (rate >= 32000) - rate = 32000; - else if (rate >= 29400) - rate = 29400; - else if (rate >= 24000) - rate = 24000; - else if (rate >= 22050) - rate = 22050; - else if (rate >= 21970) - rate = 21970; - else if (rate >= 16000) - rate = 16000; - else if (rate >= 14647) - rate = 14647; - else if (rate >= 10985) - rate = 10985; - else if (rate >= 10666) - rate = 10666; - else - rate = 8000; - - /* Set the external clock generator */ - - sa11xx_uda1341_set_audio_clock(rate); - - /* Select the clock divisor */ - switch (rate) { - case 8000: - case 10985: - case 22050: - case 24000: - clk = F512; - clk_div = SSCR0_SerClkDiv(16); - break; - case 16000: - case 21970: - case 44100: - case 48000: - clk = F256; - clk_div = SSCR0_SerClkDiv(8); - break; - case 10666: - case 14647: - case 29400: - case 32000: - clk = F384; - clk_div = SSCR0_SerClkDiv(12); - break; - } - - /* FMT setting should be moved away when other FMTs are added (FIXME) */ - l3_command(sa11xx_uda1341->uda1341, CMD_FORMAT, (void *)LSB16); - - l3_command(sa11xx_uda1341->uda1341, CMD_FS, (void *)clk); - Ser4SSCR0 = (Ser4SSCR0 & ~0xff00) + clk_div + SSCR0_SSE; - sa11xx_uda1341->samplerate = rate; -} - -/* }}} */ - -/* {{{ HW init and shutdown */ - -static void sa11xx_uda1341_audio_init(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - unsigned long flags; - - /* Setup DMA stuff */ - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].id = "UDA1341 out"; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].stream_id = SNDRV_PCM_STREAM_PLAYBACK; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK].dma_dev = DMA_Ser4SSPWr; - - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].id = "UDA1341 in"; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].stream_id = SNDRV_PCM_STREAM_CAPTURE; - sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE].dma_dev = DMA_Ser4SSPRd; - - /* Initialize the UDA1341 internal state */ - - /* Setup the uarts */ - local_irq_save(flags); - GAFR |= (GPIO_SSP_CLK); - GPDR &= ~(GPIO_SSP_CLK); - Ser4SSCR0 = 0; - Ser4SSCR0 = SSCR0_DataSize(16) + SSCR0_TI + SSCR0_SerClkDiv(8); - Ser4SSCR1 = SSCR1_SClkIactL + SSCR1_SClk1P + SSCR1_ExtClk; - Ser4SSCR0 |= SSCR0_SSE; - local_irq_restore(flags); - - /* Enable the audio power */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* Wait for the UDA1341 to wake up */ - mdelay(1); //FIXME - was removed by Perex - Why? - - /* Initialize the UDA1341 internal state */ - l3_open(sa11xx_uda1341->uda1341); - - /* external clock configuration (after l3_open - regs must be initialized */ - sa11xx_uda1341_set_samplerate(sa11xx_uda1341, sa11xx_uda1341->samplerate); - - /* Wait for the UDA1341 to wake up */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - mdelay(1); - - /* make the left and right channels unswapped (flip the WS latch) */ - Ser4SSDR = 0; - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) -{ - /* mute on */ - set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); - - /* disable the audio power and all signals leading to the audio chip */ - l3_close(sa11xx_uda1341->uda1341); - Ser4SSCR0 = 0; - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); - - /* power off and mute off */ - /* FIXME - is muting off necesary??? */ - - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -} - -/* }}} */ - -/* {{{ DMA staff */ - -/* - * these are the address and sizes used to fill the xmit buffer - * so we can get a clock in record only mode - */ -#define FORCE_CLOCK_ADDR (dma_addr_t)FLUSH_BASE_PHYS -#define FORCE_CLOCK_SIZE 4096 // was 2048 - -// FIXME Why this value exactly - wrote comment -#define DMA_BUF_SIZE 8176 /* <= MAX_DMA_SIZE from asm/arch-sa1100/dma.h */ - -#ifdef HH_VERSION - -static int audio_dma_request(struct audio_stream *s, void (*callback)(void *, int)) -{ - int ret; - - ret = sa1100_request_dma(&s->dmach, s->id, s->dma_dev); - if (ret < 0) { - printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); - return ret; - } - sa1100_dma_set_callback(s->dmach, callback); - return 0; -} - -static inline void audio_dma_free(struct audio_stream *s) -{ - sa1100_free_dma(s->dmach); - s->dmach = -1; -} - -#else - -static int audio_dma_request(struct audio_stream *s, void (*callback)(void *)) -{ - int ret; - - ret = sa1100_request_dma(s->dma_dev, s->id, callback, s, &s->dma_regs); - if (ret < 0) - printk(KERN_ERR "unable to grab audio dma 0x%x\n", s->dma_dev); - return ret; -} - -static void audio_dma_free(struct audio_stream *s) -{ - sa1100_free_dma(s->dma_regs); - s->dma_regs = 0; -} - -#endif - -static u_int audio_get_dma_pos(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int offset; - unsigned long flags; - dma_addr_t addr; - - // this must be called w/ interrupts locked out see dma-sa1100.c in the kernel - spin_lock_irqsave(&s->dma_lock, flags); -#ifdef HH_VERSION - sa1100_dma_get_current(s->dmach, NULL, &addr); -#else - addr = sa1100_get_dma_pos((s)->dma_regs); -#endif - offset = addr - runtime->dma_addr; - spin_unlock_irqrestore(&s->dma_lock, flags); - - offset = bytes_to_frames(runtime,offset); - if (offset >= runtime->buffer_size) - offset = 0; - - return offset; -} - -/* - * this stops the dma and clears the dma ptrs - */ -static void audio_stop_dma(struct audio_stream *s) -{ - unsigned long flags; - - spin_lock_irqsave(&s->dma_lock, flags); - s->active = 0; - s->period = 0; - /* this stops the dma channel and clears the buffer ptrs */ -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - sa1100_clear_dma(s->dma_regs); -#endif - spin_unlock_irqrestore(&s->dma_lock, flags); -} - -static void audio_process_dma(struct audio_stream *s) -{ - struct snd_pcm_substream *substream = s->stream; - struct snd_pcm_runtime *runtime; - unsigned int dma_size; - unsigned int offset; - int ret; - - /* we are requested to process synchronization DMA transfer */ - if (s->tx_spin) { - if (snd_BUG_ON(s->stream_id != SNDRV_PCM_STREAM_PLAYBACK)) - return; - /* fill the xmit dma buffers and return */ -#ifdef HH_VERSION - sa1100_dma_set_spin(s->dmach, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); -#else - while (1) { - ret = sa1100_start_dma(s->dma_regs, FORCE_CLOCK_ADDR, FORCE_CLOCK_SIZE); - if (ret) - return; - } -#endif - return; - } - - /* must be set here - only valid for running streams, not for forced_clock dma fills */ - runtime = substream->runtime; - while (s->active && s->periods < runtime->periods) { - dma_size = frames_to_bytes(runtime, runtime->period_size); - if (s->old_offset) { - /* a little trick, we need resume from old position */ - offset = frames_to_bytes(runtime, s->old_offset - 1); - s->old_offset = 0; - s->periods = 0; - s->period = offset / dma_size; - offset %= dma_size; - dma_size = dma_size - offset; - if (!dma_size) - continue; /* special case */ - } else { - offset = dma_size * s->period; - snd_BUG_ON(dma_size > DMA_BUF_SIZE); - } -#ifdef HH_VERSION - ret = sa1100_dma_queue_buffer(s->dmach, s, runtime->dma_addr + offset, dma_size); - if (ret) - return; //FIXME -#else - ret = sa1100_start_dma((s)->dma_regs, runtime->dma_addr + offset, dma_size); - if (ret) { - printk(KERN_ERR "audio_process_dma: cannot queue DMA buffer (%i)\n", ret); - return; - } -#endif - - s->period++; - s->period %= runtime->periods; - s->periods++; - } -} - -#ifdef HH_VERSION -static void audio_dma_callback(void *data, int size) -#else -static void audio_dma_callback(void *data) -#endif -{ - struct audio_stream *s = data; - - /* - * If we are getting a callback for an active stream then we inform - * the PCM middle layer we've finished a period - */ - if (s->active) - snd_pcm_period_elapsed(s->stream); - - spin_lock(&s->dma_lock); - if (!s->tx_spin && s->periods > 0) - s->periods--; - audio_process_dma(s); - spin_unlock(&s->dma_lock); -} - -/* }}} */ - -/* {{{ PCM setting */ - -/* {{{ trigger & timer */ - -static int snd_sa11xx_uda1341_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - int stream_id = substream->pstr->stream; - struct audio_stream *s = &chip->s[stream_id]; - struct audio_stream *s1 = &chip->s[stream_id ^ 1]; - int err = 0; - - /* note local interrupts are already disabled in the midlevel code */ - spin_lock(&s->dma_lock); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - /* now we need to make sure a record only stream has a clock */ - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - /* we need to force fill the xmit DMA with zeros */ - s1->tx_spin = 1; - audio_process_dma(s1); - } - /* this case is when you were recording then you turn on a - * playback stream so we stop (also clears it) the dma first, - * clear the sync flag and then we let it turned on - */ - else { - s->tx_spin = 0; - } - - /* requested stream startup */ - s->active = 1; - audio_process_dma(s); - break; - case SNDRV_PCM_TRIGGER_STOP: - /* requested stream shutdown */ - audio_stop_dma(s); - - /* - * now we need to make sure a record only stream has a clock - * so if we're stopping a playback with an active capture - * we need to turn the 0 fill dma on for the xmit side - */ - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK && s1->active) { - /* we need to force fill the xmit DMA with zeros */ - s->tx_spin = 1; - audio_process_dma(s); - } - /* - * we killed a capture only stream, so we should also kill - * the zero fill transmit - */ - else { - if (s1->tx_spin) { - s1->tx_spin = 0; - audio_stop_dma(s1); - } - } - - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - s->active = 0; -#ifdef HH_VERSION - sa1100_dma_stop(s->dmach); -#else - //FIXME - DMA API -#endif - s->old_offset = audio_get_dma_pos(s) + 1; -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - //FIXME - DMA API -#endif - s->periods = 0; - break; - case SNDRV_PCM_TRIGGER_RESUME: - s->active = 1; - s->tx_spin = 0; - audio_process_dma(s); - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - s1->tx_spin = 1; - audio_process_dma(s1); - } - break; - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -#ifdef HH_VERSION - sa1100_dma_stop(s->dmach); -#else - //FIXME - DMA API -#endif - s->active = 0; - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) { - if (s1->active) { - s->tx_spin = 1; - s->old_offset = audio_get_dma_pos(s) + 1; -#ifdef HH_VERSION - sa1100_dma_flush_all(s->dmach); -#else - //FIXME - DMA API -#endif - audio_process_dma(s); - } - } else { - if (s1->tx_spin) { - s1->tx_spin = 0; -#ifdef HH_VERSION - sa1100_dma_flush_all(s1->dmach); -#else - //FIXME - DMA API -#endif - } - } - break; - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - s->active = 1; - if (s->old_offset) { - s->tx_spin = 0; - audio_process_dma(s); - break; - } - if (stream_id == SNDRV_PCM_STREAM_CAPTURE && !s1->active) { - s1->tx_spin = 1; - audio_process_dma(s1); - } -#ifdef HH_VERSION - sa1100_dma_resume(s->dmach); -#else - //FIXME - DMA API -#endif - break; - default: - err = -EINVAL; - break; - } - spin_unlock(&s->dma_lock); - return err; -} - -static int snd_sa11xx_uda1341_prepare(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct audio_stream *s = &chip->s[substream->pstr->stream]; - - /* set requested samplerate */ - sa11xx_uda1341_set_samplerate(chip, runtime->rate); - - /* set requestd format when available */ - /* set FMT here !!! FIXME */ - - s->period = 0; - s->periods = 0; - - return 0; -} - -static snd_pcm_uframes_t snd_sa11xx_uda1341_pointer(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - return audio_get_dma_pos(&chip->s[substream->pstr->stream]); -} - -/* }}} */ - -static struct snd_pcm_hardware snd_sa11xx_uda1341_capture = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_KNOT), - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 64*1024, - .period_bytes_min = 64, - .period_bytes_max = DMA_BUF_SIZE, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_sa11xx_uda1341_playback = -{ - .info = (SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), - .formats = SNDRV_PCM_FMTBIT_S16_LE, - .rates = (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ - SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 |\ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\ - SNDRV_PCM_RATE_KNOT), - .rate_min = 8000, - .rate_max = 48000, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = 64*1024, - .period_bytes_min = 64, - .period_bytes_max = DMA_BUF_SIZE, - .periods_min = 2, - .periods_max = 255, - .fifo_size = 0, -}; - -static int snd_card_sa11xx_uda1341_open(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - int stream_id = substream->pstr->stream; - int err; - - chip->s[stream_id].stream = substream; - - if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) - runtime->hw = snd_sa11xx_uda1341_playback; - else - runtime->hw = snd_sa11xx_uda1341_capture; - if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) - return err; - if ((err = snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates)) < 0) - return err; - - return 0; -} - -static int snd_card_sa11xx_uda1341_close(struct snd_pcm_substream *substream) -{ - struct sa11xx_uda1341 *chip = snd_pcm_substream_chip(substream); - - chip->s[substream->pstr->stream].stream = NULL; - return 0; -} - -/* {{{ HW params & free */ - -static int snd_sa11xx_uda1341_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -} - -static int snd_sa11xx_uda1341_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - -/* }}} */ - -static struct snd_pcm_ops snd_card_sa11xx_uda1341_playback_ops = { - .open = snd_card_sa11xx_uda1341_open, - .close = snd_card_sa11xx_uda1341_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_sa11xx_uda1341_hw_params, - .hw_free = snd_sa11xx_uda1341_hw_free, - .prepare = snd_sa11xx_uda1341_prepare, - .trigger = snd_sa11xx_uda1341_trigger, - .pointer = snd_sa11xx_uda1341_pointer, -}; - -static struct snd_pcm_ops snd_card_sa11xx_uda1341_capture_ops = { - .open = snd_card_sa11xx_uda1341_open, - .close = snd_card_sa11xx_uda1341_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_sa11xx_uda1341_hw_params, - .hw_free = snd_sa11xx_uda1341_hw_free, - .prepare = snd_sa11xx_uda1341_prepare, - .trigger = snd_sa11xx_uda1341_trigger, - .pointer = snd_sa11xx_uda1341_pointer, -}; - -static int __init snd_card_sa11xx_uda1341_pcm(struct sa11xx_uda1341 *sa11xx_uda1341, int device) -{ - struct snd_pcm *pcm; - int err; - - if ((err = snd_pcm_new(sa11xx_uda1341->card, "UDA1341 PCM", device, 1, 1, &pcm)) < 0) - return err; - - /* - * this sets up our initial buffers and sets the dma_type to isa. - * isa works but I'm not sure why (or if) it's the right choice - * this may be too large, trying it for now - */ - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, 64*1024); - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_card_sa11xx_uda1341_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_card_sa11xx_uda1341_capture_ops); - pcm->private_data = sa11xx_uda1341; - pcm->info_flags = 0; - strcpy(pcm->name, "UDA1341 PCM"); - - sa11xx_uda1341_audio_init(sa11xx_uda1341); - - /* setup DMA controller */ - audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_PLAYBACK], audio_dma_callback); - audio_dma_request(&sa11xx_uda1341->s[SNDRV_PCM_STREAM_CAPTURE], audio_dma_callback); - - sa11xx_uda1341->pcm = pcm; - - return 0; -} - -/* }}} */ - -/* {{{ module init & exit */ - -#ifdef CONFIG_PM - -static int snd_sa11xx_uda1341_suspend(struct platform_device *devptr, - pm_message_t state) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - snd_pcm_suspend_all(chip->pcm); -#ifdef HH_VERSION - sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); - sa1100_dma_sleep(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); -#else - //FIXME -#endif - l3_command(chip->uda1341, CMD_SUSPEND, NULL); - sa11xx_uda1341_audio_shutdown(chip); - - return 0; -} - -static int snd_sa11xx_uda1341_resume(struct platform_device *devptr) -{ - struct snd_card *card = platform_get_drvdata(devptr); - struct sa11xx_uda1341 *chip = card->private_data; - - sa11xx_uda1341_audio_init(chip); - l3_command(chip->uda1341, CMD_RESUME, NULL); -#ifdef HH_VERSION - sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_PLAYBACK].dmach); - sa1100_dma_wakeup(chip->s[SNDRV_PCM_STREAM_CAPTURE].dmach); -#else - //FIXME -#endif - snd_power_change_state(card, SNDRV_CTL_POWER_D0); - return 0; -} -#endif /* COMFIG_PM */ - -void snd_sa11xx_uda1341_free(struct snd_card *card) -{ - struct sa11xx_uda1341 *chip = card->private_data; - - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_PLAYBACK]); - audio_dma_free(&chip->s[SNDRV_PCM_STREAM_CAPTURE]); -} - -static int __devinit sa11xx_uda1341_probe(struct platform_device *devptr) -{ - int err; - struct snd_card *card; - struct sa11xx_uda1341 *chip; - - /* register the soundcard */ - err = snd_card_create(-1, id, THIS_MODULE, - sizeof(struct sa11xx_uda1341), &card); - if (err < 0) - return err; - - chip = card->private_data; - spin_lock_init(&chip->s[0].dma_lock); - spin_lock_init(&chip->s[1].dma_lock); - - card->private_free = snd_sa11xx_uda1341_free; - chip->card = card; - chip->samplerate = AUDIO_RATE_DEFAULT; - - // mixer - if ((err = snd_chip_uda1341_mixer_new(card, &chip->uda1341))) - goto nodev; - - // PCM - if ((err = snd_card_sa11xx_uda1341_pcm(chip, 0)) < 0) - goto nodev; - - strcpy(card->driver, "UDA1341"); - strcpy(card->shortname, "H3600 UDA1341TS"); - sprintf(card->longname, "Compaq iPAQ H3600 with Philips UDA1341TS"); - - snd_card_set_dev(card, &devptr->dev); - - if ((err = snd_card_register(card)) == 0) { - printk( KERN_INFO "iPAQ audio support initialized\n" ); - platform_set_drvdata(devptr, card); - return 0; - } - - nodev: - snd_card_free(card); - return err; -} - -static int __devexit sa11xx_uda1341_remove(struct platform_device *devptr) -{ - snd_card_free(platform_get_drvdata(devptr)); - platform_set_drvdata(devptr, NULL); - return 0; -} - -#define SA11XX_UDA1341_DRIVER "sa11xx_uda1341" - -static struct platform_driver sa11xx_uda1341_driver = { - .probe = sa11xx_uda1341_probe, - .remove = __devexit_p(sa11xx_uda1341_remove), -#ifdef CONFIG_PM - .suspend = snd_sa11xx_uda1341_suspend, - .resume = snd_sa11xx_uda1341_resume, -#endif - .driver = { - .name = SA11XX_UDA1341_DRIVER, - }, -}; - -static int __init sa11xx_uda1341_init(void) -{ - int err; - - if (!machine_is_h3xxx()) - return -ENODEV; - if ((err = platform_driver_register(&sa11xx_uda1341_driver)) < 0) - return err; - device = platform_device_register_simple(SA11XX_UDA1341_DRIVER, -1, NULL, 0); - if (!IS_ERR(device)) { - if (platform_get_drvdata(device)) - return 0; - platform_device_unregister(device); - err = -ENODEV; - } else - err = PTR_ERR(device); - platform_driver_unregister(&sa11xx_uda1341_driver); - return err; -} - -static void __exit sa11xx_uda1341_exit(void) -{ - platform_device_unregister(device); - platform_driver_unregister(&sa11xx_uda1341_driver); -} - -module_init(sa11xx_uda1341_init); -module_exit(sa11xx_uda1341_exit); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ diff --git a/sound/atmel/Kconfig b/sound/atmel/Kconfig new file mode 100644 index 0000000..6c228a9 --- /dev/null +++ b/sound/atmel/Kconfig @@ -0,0 +1,19 @@ +menu "Atmel devices (AVR32 and AT91)" + depends on AVR32 || ARCH_AT91 + +config SND_ATMEL_ABDAC + tristate "Atmel Audio Bitstream DAC (ABDAC) driver" + select SND_PCM + depends on DW_DMAC && AVR32 + help + ALSA sound driver for the Atmel Audio Bitstream DAC (ABDAC). + +config SND_ATMEL_AC97C + tristate "Atmel AC97 Controller (AC97C) driver" + select SND_PCM + select SND_AC97_CODEC + depends on DW_DMAC && AVR32 + help + ALSA sound driver for the Atmel AC97 controller. + +endmenu diff --git a/sound/atmel/Makefile b/sound/atmel/Makefile new file mode 100644 index 0000000..219dcfa --- /dev/null +++ b/sound/atmel/Makefile @@ -0,0 +1,5 @@ +snd-atmel-abdac-objs := abdac.o +snd-atmel-ac97c-objs := ac97c.o + +obj-$(CONFIG_SND_ATMEL_ABDAC) += snd-atmel-abdac.o +obj-$(CONFIG_SND_ATMEL_AC97C) += snd-atmel-ac97c.o diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c new file mode 100644 index 0000000..f2f41c8 --- /dev/null +++ b/sound/atmel/abdac.c @@ -0,0 +1,602 @@ +/* + * Driver for the Atmel on-chip Audio Bitstream DAC (ABDAC) + * + * Copyright (C) 2006-2009 Atmel Corporation + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published by + * the Free Software Foundation. + */ +#include <linux/clk.h> +#include <linux/bitmap.h> +#include <linux/dw_dmac.h> +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/atmel-abdac.h> + +/* DAC register offsets */ +#define DAC_DATA 0x0000 +#define DAC_CTRL 0x0008 +#define DAC_INT_MASK 0x000c +#define DAC_INT_EN 0x0010 +#define DAC_INT_DIS 0x0014 +#define DAC_INT_CLR 0x0018 +#define DAC_INT_STATUS 0x001c + +/* Bitfields in CTRL */ +#define DAC_SWAP_OFFSET 30 +#define DAC_SWAP_SIZE 1 +#define DAC_EN_OFFSET 31 +#define DAC_EN_SIZE 1 + +/* Bitfields in INT_MASK/INT_EN/INT_DIS/INT_STATUS/INT_CLR */ +#define DAC_UNDERRUN_OFFSET 28 +#define DAC_UNDERRUN_SIZE 1 +#define DAC_TX_READY_OFFSET 29 +#define DAC_TX_READY_SIZE 1 + +/* Bit manipulation macros */ +#define DAC_BIT(name) \ + (1 << DAC_##name##_OFFSET) +#define DAC_BF(name, value) \ + (((value) & ((1 << DAC_##name##_SIZE) - 1)) \ + << DAC_##name##_OFFSET) +#define DAC_BFEXT(name, value) \ + (((value) >> DAC_##name##_OFFSET) \ + & ((1 << DAC_##name##_SIZE) - 1)) +#define DAC_BFINS(name, value, old) \ + (((old) & ~(((1 << DAC_##name##_SIZE) - 1) \ + << DAC_##name##_OFFSET)) \ + | DAC_BF(name, value)) + +/* Register access macros */ +#define dac_readl(port, reg) \ + __raw_readl((port)->regs + DAC_##reg) +#define dac_writel(port, reg, value) \ + __raw_writel((value), (port)->regs + DAC_##reg) + +/* + * ABDAC supports a maximum of 6 different rates from a generic clock. The + * generic clock has a power of two divider, which gives 6 steps from 192 kHz + * to 5112 Hz. + */ +#define MAX_NUM_RATES 6 +/* ALSA seems to use rates between 192000 Hz and 5112 Hz. */ +#define RATE_MAX 192000 +#define RATE_MIN 5112 + +enum { + DMA_READY = 0, +}; + +struct atmel_abdac_dma { + struct dma_chan *chan; + struct dw_cyclic_desc *cdesc; +}; + +struct atmel_abdac { + struct clk *pclk; + struct clk *sample_clk; + struct platform_device *pdev; + struct atmel_abdac_dma dma; + + struct snd_pcm_hw_constraint_list constraints_rates; + struct snd_pcm_substream *substream; + struct snd_card *card; + struct snd_pcm *pcm; + + void __iomem *regs; + unsigned long flags; + unsigned int rates[MAX_NUM_RATES]; + unsigned int rates_num; + int irq; +}; + +#define get_dac(card) ((struct atmel_abdac *)(card)->private_data) + +/* This function is called by the DMA driver. */ +static void atmel_abdac_dma_period_done(void *arg) +{ + struct atmel_abdac *dac = arg; + snd_pcm_period_elapsed(dac->substream); +} + +static int atmel_abdac_prepare_dma(struct atmel_abdac *dac, + struct snd_pcm_substream *substream, + enum dma_data_direction direction) +{ + struct dma_chan *chan = dac->dma.chan; + struct dw_cyclic_desc *cdesc; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long buffer_len, period_len; + + /* + * We don't do DMA on "complex" transfers, i.e. with + * non-halfword-aligned buffers or lengths. + */ + if (runtime->dma_addr & 1 || runtime->buffer_size & 1) { + dev_dbg(&dac->pdev->dev, "too complex transfer\n"); + return -EINVAL; + } + + buffer_len = frames_to_bytes(runtime, runtime->buffer_size); + period_len = frames_to_bytes(runtime, runtime->period_size); + + cdesc = dw_dma_cyclic_prep(chan, runtime->dma_addr, buffer_len, + period_len, DMA_TO_DEVICE); + if (IS_ERR(cdesc)) { + dev_dbg(&dac->pdev->dev, "could not prepare cyclic DMA\n"); + return PTR_ERR(cdesc); + } + + cdesc->period_callback = atmel_abdac_dma_period_done; + cdesc->period_callback_param = dac; + + dac->dma.cdesc = cdesc; + + set_bit(DMA_READY, &dac->flags); + + return 0; +} + +static struct snd_pcm_hardware atmel_abdac_hw = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_BLOCK_TRANSFER + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_BE), + .rates = (SNDRV_PCM_RATE_KNOT), + .rate_min = RATE_MIN, + .rate_max = RATE_MAX, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64 * 4096, + .period_bytes_min = 4096, + .period_bytes_max = 4096, + .periods_min = 6, + .periods_max = 64, +}; + +static int atmel_abdac_open(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + + dac->substream = substream; + atmel_abdac_hw.rate_max = dac->rates[dac->rates_num - 1]; + atmel_abdac_hw.rate_min = dac->rates[0]; + substream->runtime->hw = atmel_abdac_hw; + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &dac->constraints_rates); +} + +static int atmel_abdac_close(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + dac->substream = NULL; + return 0; +} + +static int atmel_abdac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + int retval; + + retval = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_READY, &dac->flags)) + dw_dma_cyclic_free(dac->dma.chan); + + return retval; +} + +static int atmel_abdac_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + if (test_and_clear_bit(DMA_READY, &dac->flags)) + dw_dma_cyclic_free(dac->dma.chan); + return snd_pcm_lib_free_pages(substream); +} + +static int atmel_abdac_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + int retval; + + retval = clk_set_rate(dac->sample_clk, 256 * substream->runtime->rate); + if (retval) + return retval; + + if (!test_bit(DMA_READY, &dac->flags)) + retval = atmel_abdac_prepare_dma(dac, substream, DMA_TO_DEVICE); + + return retval; +} + +static int atmel_abdac_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + int retval = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ + case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_START: + clk_enable(dac->sample_clk); + retval = dw_dma_cyclic_start(dac->dma.chan); + if (retval) + goto out; + dac_writel(dac, CTRL, DAC_BIT(EN)); + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ + case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_STOP: + dw_dma_cyclic_stop(dac->dma.chan); + dac_writel(dac, DATA, 0); + dac_writel(dac, CTRL, 0); + clk_disable(dac->sample_clk); + break; + default: + retval = -EINVAL; + break; + } +out: + return retval; +} + +static snd_pcm_uframes_t +atmel_abdac_pointer(struct snd_pcm_substream *substream) +{ + struct atmel_abdac *dac = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t frames; + unsigned long bytes; + + bytes = dw_dma_get_src_addr(dac->dma.chan); + bytes -= runtime->dma_addr; + + frames = bytes_to_frames(runtime, bytes); + if (frames >= runtime->buffer_size) + frames -= runtime->buffer_size; + + return frames; +} + +static irqreturn_t abdac_interrupt(int irq, void *dev_id) +{ + struct atmel_abdac *dac = dev_id; + u32 status; + + status = dac_readl(dac, INT_STATUS); + if (status & DAC_BIT(UNDERRUN)) { + dev_err(&dac->pdev->dev, "underrun detected\n"); + dac_writel(dac, INT_CLR, DAC_BIT(UNDERRUN)); + } else { + dev_err(&dac->pdev->dev, "spurious interrupt (status=0x%x)\n", + status); + dac_writel(dac, INT_CLR, status); + } + + return IRQ_HANDLED; +} + +static struct snd_pcm_ops atmel_abdac_ops = { + .open = atmel_abdac_open, + .close = atmel_abdac_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_abdac_hw_params, + .hw_free = atmel_abdac_hw_free, + .prepare = atmel_abdac_prepare, + .trigger = atmel_abdac_trigger, + .pointer = atmel_abdac_pointer, +}; + +static int __devinit atmel_abdac_pcm_new(struct atmel_abdac *dac) +{ + struct snd_pcm_hardware hw = atmel_abdac_hw; + struct snd_pcm *pcm; + int retval; + + retval = snd_pcm_new(dac->card, dac->card->shortname, + dac->pdev->id, 1, 0, &pcm); + if (retval) + return retval; + + strcpy(pcm->name, dac->card->shortname); + pcm->private_data = dac; + pcm->info_flags = 0; + dac->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &atmel_abdac_ops); + + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &dac->pdev->dev, hw.periods_min * hw.period_bytes_min, + hw.buffer_bytes_max); + + return retval; +} + +static bool filter(struct dma_chan *chan, void *slave) +{ + struct dw_dma_slave *dws = slave; + + if (dws->dma_dev == chan->device->dev) { + chan->private = dws; + return true; + } else + return false; +} + +static int set_sample_rates(struct atmel_abdac *dac) +{ + long new_rate = RATE_MAX; + int retval = -EINVAL; + int index = 0; + + /* we start at 192 kHz and work our way down to 5112 Hz */ + while (new_rate >= RATE_MIN && index < (MAX_NUM_RATES + 1)) { + new_rate = clk_round_rate(dac->sample_clk, 256 * new_rate); + if (new_rate < 0) + break; + /* make sure we are below the ABDAC clock */ + if (new_rate <= clk_get_rate(dac->pclk)) { + dac->rates[index] = new_rate / 256; + index++; + } + /* divide by 256 and then by two to get next rate */ + new_rate /= 256 * 2; + } + + if (index) { + int i; + + /* reverse array, smallest go first */ + for (i = 0; i < (index / 2); i++) { + unsigned int tmp = dac->rates[index - 1 - i]; + dac->rates[index - 1 - i] = dac->rates[i]; + dac->rates[i] = tmp; + } + + dac->constraints_rates.count = index; + dac->constraints_rates.list = dac->rates; + dac->constraints_rates.mask = 0; + dac->rates_num = index; + + retval = 0; + } + + return retval; +} + +static int __devinit atmel_abdac_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct atmel_abdac *dac; + struct resource *regs; + struct atmel_abdac_pdata *pdata; + struct clk *pclk; + struct clk *sample_clk; + int retval; + int irq; + + regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!regs) { + dev_dbg(&pdev->dev, "no memory resource\n"); + return -ENXIO; + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_dbg(&pdev->dev, "could not get IRQ number\n"); + return irq; + } + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_dbg(&pdev->dev, "no platform data\n"); + return -ENXIO; + } + + pclk = clk_get(&pdev->dev, "pclk"); + if (IS_ERR(pclk)) { + dev_dbg(&pdev->dev, "no peripheral clock\n"); + return PTR_ERR(pclk); + } + sample_clk = clk_get(&pdev->dev, "sample_clk"); + if (IS_ERR(pclk)) { + dev_dbg(&pdev->dev, "no sample clock\n"); + retval = PTR_ERR(pclk); + goto out_put_pclk; + } + clk_enable(pclk); + + retval = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, sizeof(struct atmel_abdac), &card); + if (retval) { + dev_dbg(&pdev->dev, "could not create sound card device\n"); + goto out_put_sample_clk; + } + + dac = get_dac(card); + + dac->irq = irq; + dac->card = card; + dac->pclk = pclk; + dac->sample_clk = sample_clk; + dac->pdev = pdev; + + retval = set_sample_rates(dac); + if (retval < 0) { + dev_dbg(&pdev->dev, "could not set supported rates\n"); + goto out_free_card; + } + + dac->regs = ioremap(regs->start, regs->end - regs->start + 1); + if (!dac->regs) { + dev_dbg(&pdev->dev, "could not remap register memory\n"); + goto out_free_card; + } + + /* make sure the DAC is silent and disabled */ + dac_writel(dac, DATA, 0); + dac_writel(dac, CTRL, 0); + + retval = request_irq(irq, abdac_interrupt, 0, "abdac", dac); + if (retval) { + dev_dbg(&pdev->dev, "could not request irq\n"); + goto out_unmap_regs; + } + + snd_card_set_dev(card, &pdev->dev); + + if (pdata->dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->dws; + dma_cap_mask_t mask; + + dws->tx_reg = regs->start + DAC_DATA; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + dac->dma.chan = dma_request_channel(mask, filter, dws); + } + if (!pdata->dws.dma_dev || !dac->dma.chan) { + dev_dbg(&pdev->dev, "DMA not available\n"); + retval = -ENODEV; + goto out_unset_card_dev; + } + + strcpy(card->driver, "Atmel ABDAC"); + strcpy(card->shortname, "Atmel ABDAC"); + sprintf(card->longname, "Atmel Audio Bitstream DAC"); + + retval = atmel_abdac_pcm_new(dac); + if (retval) { + dev_dbg(&pdev->dev, "could not register ABDAC pcm device\n"); + goto out_release_dma; + } + + retval = snd_card_register(card); + if (retval) { + dev_dbg(&pdev->dev, "could not register sound card\n"); + goto out_release_dma; + } + + platform_set_drvdata(pdev, card); + + dev_info(&pdev->dev, "Atmel ABDAC at 0x%p using %s\n", + dac->regs, dev_name(&dac->dma.chan->dev->device)); + + return retval; + +out_release_dma: + dma_release_channel(dac->dma.chan); + dac->dma.chan = NULL; +out_unset_card_dev: + snd_card_set_dev(card, NULL); + free_irq(irq, dac); +out_unmap_regs: + iounmap(dac->regs); +out_free_card: + snd_card_free(card); +out_put_sample_clk: + clk_put(sample_clk); + clk_disable(pclk); +out_put_pclk: + clk_put(pclk); + return retval; +} + +#ifdef CONFIG_PM +static int atmel_abdac_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_abdac *dac = card->private_data; + + dw_dma_cyclic_stop(dac->dma.chan); + clk_disable(dac->sample_clk); + clk_disable(dac->pclk); + + return 0; +} + +static int atmel_abdac_resume(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_abdac *dac = card->private_data; + + clk_enable(dac->pclk); + clk_enable(dac->sample_clk); + if (test_bit(DMA_READY, &dac->flags)) + dw_dma_cyclic_start(dac->dma.chan); + + return 0; +} +#else +#define atmel_abdac_suspend NULL +#define atmel_abdac_resume NULL +#endif + +static int __devexit atmel_abdac_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_abdac *dac = get_dac(card); + + clk_put(dac->sample_clk); + clk_disable(dac->pclk); + clk_put(dac->pclk); + + dma_release_channel(dac->dma.chan); + dac->dma.chan = NULL; + snd_card_set_dev(card, NULL); + iounmap(dac->regs); + free_irq(dac->irq, dac); + snd_card_free(card); + + platform_set_drvdata(pdev, NULL); + + return 0; +} + +static struct platform_driver atmel_abdac_driver = { + .remove = __devexit_p(atmel_abdac_remove), + .driver = { + .name = "atmel_abdac", + }, + .suspend = atmel_abdac_suspend, + .resume = atmel_abdac_resume, +}; + +static int __init atmel_abdac_init(void) +{ + return platform_driver_probe(&atmel_abdac_driver, + atmel_abdac_probe); +} +module_init(atmel_abdac_init); + +static void __exit atmel_abdac_exit(void) +{ + platform_driver_unregister(&atmel_abdac_driver); +} +module_exit(atmel_abdac_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Driver for Atmel Audio Bitstream DAC (ABDAC)"); +MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>"); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c new file mode 100644 index 0000000..0c0f877 --- /dev/null +++ b/sound/atmel/ac97c.c @@ -0,0 +1,1022 @@ +/* + * Driver for Atmel AC97C + * + * Copyright (C) 2005-2009 Atmel Corporation + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published by + * the Free Software Foundation. + */ +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/bitmap.h> +#include <linux/device.h> +#include <linux/dmaengine.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/mutex.h> +#include <linux/gpio.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/ac97_codec.h> +#include <sound/atmel-ac97c.h> +#include <sound/memalloc.h> + +#include <linux/dw_dmac.h> + +#include "ac97c.h" + +enum { + DMA_TX_READY = 0, + DMA_RX_READY, + DMA_TX_CHAN_PRESENT, + DMA_RX_CHAN_PRESENT, +}; + +/* Serialize access to opened variable */ +static DEFINE_MUTEX(opened_mutex); + +struct atmel_ac97c_dma { + struct dma_chan *rx_chan; + struct dma_chan *tx_chan; +}; + +struct atmel_ac97c { + struct clk *pclk; + struct platform_device *pdev; + struct atmel_ac97c_dma dma; + + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_ac97 *ac97; + struct snd_ac97_bus *ac97_bus; + + u64 cur_format; + unsigned int cur_rate; + unsigned long flags; + /* Serialize access to opened variable */ + spinlock_t lock; + void __iomem *regs; + int irq; + int opened; + int reset_pin; +}; + +#define get_chip(card) ((struct atmel_ac97c *)(card)->private_data) + +#define ac97c_writel(chip, reg, val) \ + __raw_writel((val), (chip)->regs + AC97C_##reg) +#define ac97c_readl(chip, reg) \ + __raw_readl((chip)->regs + AC97C_##reg) + +/* This function is called by the DMA driver. */ +static void atmel_ac97c_dma_playback_period_done(void *arg) +{ + struct atmel_ac97c *chip = arg; + snd_pcm_period_elapsed(chip->playback_substream); +} + +static void atmel_ac97c_dma_capture_period_done(void *arg) +{ + struct atmel_ac97c *chip = arg; + snd_pcm_period_elapsed(chip->capture_substream); +} + +static int atmel_ac97c_prepare_dma(struct atmel_ac97c *chip, + struct snd_pcm_substream *substream, + enum dma_data_direction direction) +{ + struct dma_chan *chan; + struct dw_cyclic_desc *cdesc; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long buffer_len, period_len; + + /* + * We don't do DMA on "complex" transfers, i.e. with + * non-halfword-aligned buffers or lengths. + */ + if (runtime->dma_addr & 1 || runtime->buffer_size & 1) { + dev_dbg(&chip->pdev->dev, "too complex transfer\n"); + return -EINVAL; + } + + if (direction == DMA_TO_DEVICE) + chan = chip->dma.tx_chan; + else + chan = chip->dma.rx_chan; + + buffer_len = frames_to_bytes(runtime, runtime->buffer_size); + period_len = frames_to_bytes(runtime, runtime->period_size); + + cdesc = dw_dma_cyclic_prep(chan, runtime->dma_addr, buffer_len, + period_len, direction); + if (IS_ERR(cdesc)) { + dev_dbg(&chip->pdev->dev, "could not prepare cyclic DMA\n"); + return PTR_ERR(cdesc); + } + + if (direction == DMA_TO_DEVICE) { + cdesc->period_callback = atmel_ac97c_dma_playback_period_done; + set_bit(DMA_TX_READY, &chip->flags); + } else { + cdesc->period_callback = atmel_ac97c_dma_capture_period_done; + set_bit(DMA_RX_READY, &chip->flags); + } + + cdesc->period_callback_param = chip; + + return 0; +} + +static struct snd_pcm_hardware atmel_ac97c_hw = { + .info = (SNDRV_PCM_INFO_MMAP + | SNDRV_PCM_INFO_MMAP_VALID + | SNDRV_PCM_INFO_INTERLEAVED + | SNDRV_PCM_INFO_BLOCK_TRANSFER + | SNDRV_PCM_INFO_JOINT_DUPLEX + | SNDRV_PCM_INFO_RESUME + | SNDRV_PCM_INFO_PAUSE), + .formats = (SNDRV_PCM_FMTBIT_S16_BE + | SNDRV_PCM_FMTBIT_S16_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS), + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 2 * 2 * 64 * 2048, + .period_bytes_min = 4096, + .period_bytes_max = 4096, + .periods_min = 6, + .periods_max = 64, +}; + +static int atmel_ac97c_playback_open(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + mutex_lock(&opened_mutex); + chip->opened++; + runtime->hw = atmel_ac97c_hw; + if (chip->cur_rate) { + runtime->hw.rate_min = chip->cur_rate; + runtime->hw.rate_max = chip->cur_rate; + } + if (chip->cur_format) + runtime->hw.formats = (1ULL << chip->cur_format); + mutex_unlock(&opened_mutex); + chip->playback_substream = substream; + return 0; +} + +static int atmel_ac97c_capture_open(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + mutex_lock(&opened_mutex); + chip->opened++; + runtime->hw = atmel_ac97c_hw; + if (chip->cur_rate) { + runtime->hw.rate_min = chip->cur_rate; + runtime->hw.rate_max = chip->cur_rate; + } + if (chip->cur_format) + runtime->hw.formats = (1ULL << chip->cur_format); + mutex_unlock(&opened_mutex); + chip->capture_substream = substream; + return 0; +} + +static int atmel_ac97c_playback_close(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + + mutex_lock(&opened_mutex); + chip->opened--; + if (!chip->opened) { + chip->cur_rate = 0; + chip->cur_format = 0; + } + mutex_unlock(&opened_mutex); + + chip->playback_substream = NULL; + + return 0; +} + +static int atmel_ac97c_capture_close(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + + mutex_lock(&opened_mutex); + chip->opened--; + if (!chip->opened) { + chip->cur_rate = 0; + chip->cur_format = 0; + } + mutex_unlock(&opened_mutex); + + chip->capture_substream = NULL; + + return 0; +} + +static int atmel_ac97c_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + int retval; + + retval = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + + /* Set restrictions to params. */ + mutex_lock(&opened_mutex); + chip->cur_rate = params_rate(hw_params); + chip->cur_format = params_format(hw_params); + mutex_unlock(&opened_mutex); + + return retval; +} + +static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + int retval; + + retval = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + + /* Set restrictions to params. */ + mutex_lock(&opened_mutex); + chip->cur_rate = params_rate(hw_params); + chip->cur_format = params_format(hw_params); + mutex_unlock(&opened_mutex); + + return retval; +} + +static int atmel_ac97c_playback_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + return snd_pcm_lib_free_pages(substream); +} + +static int atmel_ac97c_capture_hw_free(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + return snd_pcm_lib_free_pages(substream); +} + +static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long word = ac97c_readl(chip, OCA); + int retval; + + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + + /* assign channels to AC97C channel A */ + switch (runtime->channels) { + case 1: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A); + break; + case 2: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A) + | AC97C_CH_ASSIGN(PCM_RIGHT, A); + break; + default: + /* TODO: support more than two channels */ + return -EINVAL; + } + ac97c_writel(chip, OCA, word); + + /* configure sample format and size */ + word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; + + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + word |= AC97C_CMR_CEM_LITTLE; + break; + case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ + word &= ~(AC97C_CMR_CEM_LITTLE); + break; + default: + word = ac97c_readl(chip, OCA); + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + ac97c_writel(chip, OCA, word); + return -EINVAL; + } + + /* Enable underrun interrupt on channel A */ + word |= AC97C_CSR_UNRUN; + + ac97c_writel(chip, CAMR, word); + + /* Enable channel A event interrupt */ + word = ac97c_readl(chip, IMR); + word |= AC97C_SR_CAEVT; + ac97c_writel(chip, IER, word); + + /* set variable rate if needed */ + if (runtime->rate != 48000) { + word = ac97c_readl(chip, MR); + word |= AC97C_MR_VRA; + ac97c_writel(chip, MR, word); + } else { + word = ac97c_readl(chip, MR); + word &= ~(AC97C_MR_VRA); + ac97c_writel(chip, MR, word); + } + + retval = snd_ac97_set_rate(chip->ac97, AC97_PCM_FRONT_DAC_RATE, + runtime->rate); + if (retval) + dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", + runtime->rate); + + if (!test_bit(DMA_TX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_TO_DEVICE); + + return retval; +} + +static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long word = ac97c_readl(chip, ICA); + int retval; + + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + + /* assign channels to AC97C channel A */ + switch (runtime->channels) { + case 1: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A); + break; + case 2: + word |= AC97C_CH_ASSIGN(PCM_LEFT, A) + | AC97C_CH_ASSIGN(PCM_RIGHT, A); + break; + default: + /* TODO: support more than two channels */ + return -EINVAL; + } + ac97c_writel(chip, ICA, word); + + /* configure sample format and size */ + word = AC97C_CMR_DMAEN | AC97C_CMR_SIZE_16; + + switch (runtime->format) { + case SNDRV_PCM_FORMAT_S16_LE: + word |= AC97C_CMR_CEM_LITTLE; + break; + case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ + word &= ~(AC97C_CMR_CEM_LITTLE); + break; + default: + word = ac97c_readl(chip, ICA); + word &= ~(AC97C_CH_MASK(PCM_LEFT) | AC97C_CH_MASK(PCM_RIGHT)); + ac97c_writel(chip, ICA, word); + return -EINVAL; + } + + /* Enable overrun interrupt on channel A */ + word |= AC97C_CSR_OVRUN; + + ac97c_writel(chip, CAMR, word); + + /* Enable channel A event interrupt */ + word = ac97c_readl(chip, IMR); + word |= AC97C_SR_CAEVT; + ac97c_writel(chip, IER, word); + + /* set variable rate if needed */ + if (runtime->rate != 48000) { + word = ac97c_readl(chip, MR); + word |= AC97C_MR_VRA; + ac97c_writel(chip, MR, word); + } else { + word = ac97c_readl(chip, MR); + word &= ~(AC97C_MR_VRA); + ac97c_writel(chip, MR, word); + } + + retval = snd_ac97_set_rate(chip->ac97, AC97_PCM_LR_ADC_RATE, + runtime->rate); + if (retval) + dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", + runtime->rate); + + if (!test_bit(DMA_RX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_FROM_DEVICE); + + return retval; +} + +static int +atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + unsigned long camr; + int retval = 0; + + camr = ac97c_readl(chip, CAMR); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ + case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_START: + retval = dw_dma_cyclic_start(chip->dma.tx_chan); + if (retval) + goto out; + camr |= AC97C_CMR_CENA; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ + case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_STOP: + dw_dma_cyclic_stop(chip->dma.tx_chan); + if (chip->opened <= 1) + camr &= ~AC97C_CMR_CENA; + break; + default: + retval = -EINVAL; + goto out; + } + + ac97c_writel(chip, CAMR, camr); +out: + return retval; +} + +static int +atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + unsigned long camr; + int retval = 0; + + camr = ac97c_readl(chip, CAMR); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ + case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ + case SNDRV_PCM_TRIGGER_START: + retval = dw_dma_cyclic_start(chip->dma.rx_chan); + if (retval) + goto out; + camr |= AC97C_CMR_CENA; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ + case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ + case SNDRV_PCM_TRIGGER_STOP: + dw_dma_cyclic_stop(chip->dma.rx_chan); + if (chip->opened <= 1) + camr &= ~AC97C_CMR_CENA; + break; + default: + retval = -EINVAL; + break; + } + + ac97c_writel(chip, CAMR, camr); +out: + return retval; +} + +static snd_pcm_uframes_t +atmel_ac97c_playback_pointer(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t frames; + unsigned long bytes; + + bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + bytes -= runtime->dma_addr; + + frames = bytes_to_frames(runtime, bytes); + if (frames >= runtime->buffer_size) + frames -= runtime->buffer_size; + return frames; +} + +static snd_pcm_uframes_t +atmel_ac97c_capture_pointer(struct snd_pcm_substream *substream) +{ + struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t frames; + unsigned long bytes; + + bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + bytes -= runtime->dma_addr; + + frames = bytes_to_frames(runtime, bytes); + if (frames >= runtime->buffer_size) + frames -= runtime->buffer_size; + return frames; +} + +static struct snd_pcm_ops atmel_ac97_playback_ops = { + .open = atmel_ac97c_playback_open, + .close = atmel_ac97c_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_ac97c_playback_hw_params, + .hw_free = atmel_ac97c_playback_hw_free, + .prepare = atmel_ac97c_playback_prepare, + .trigger = atmel_ac97c_playback_trigger, + .pointer = atmel_ac97c_playback_pointer, +}; + +static struct snd_pcm_ops atmel_ac97_capture_ops = { + .open = atmel_ac97c_capture_open, + .close = atmel_ac97c_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = atmel_ac97c_capture_hw_params, + .hw_free = atmel_ac97c_capture_hw_free, + .prepare = atmel_ac97c_capture_prepare, + .trigger = atmel_ac97c_capture_trigger, + .pointer = atmel_ac97c_capture_pointer, +}; + +static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) +{ + struct atmel_ac97c *chip = (struct atmel_ac97c *)dev; + irqreturn_t retval = IRQ_NONE; + u32 sr = ac97c_readl(chip, SR); + u32 casr = ac97c_readl(chip, CASR); + u32 cosr = ac97c_readl(chip, COSR); + + if (sr & AC97C_SR_CAEVT) { + dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", + casr & AC97C_CSR_OVRUN ? " OVRUN" : "", + casr & AC97C_CSR_RXRDY ? " RXRDY" : "", + casr & AC97C_CSR_UNRUN ? " UNRUN" : "", + casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", + casr & AC97C_CSR_TXRDY ? " TXRDY" : "", + !casr ? " NONE" : ""); + retval = IRQ_HANDLED; + } + + if (sr & AC97C_SR_COEVT) { + dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n", + cosr & AC97C_CSR_OVRUN ? " OVRUN" : "", + cosr & AC97C_CSR_RXRDY ? " RXRDY" : "", + cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", + cosr & AC97C_CSR_TXRDY ? " TXRDY" : "", + !cosr ? " NONE" : ""); + retval = IRQ_HANDLED; + } + + if (retval == IRQ_NONE) { + dev_err(&chip->pdev->dev, "spurious interrupt sr 0x%08x " + "casr 0x%08x cosr 0x%08x\n", sr, casr, cosr); + } + + return retval; +} + +static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip) +{ + struct snd_pcm *pcm; + struct snd_pcm_hardware hw = atmel_ac97c_hw; + int capture, playback, retval; + + capture = test_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + playback = test_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + + retval = snd_pcm_new(chip->card, chip->card->shortname, + chip->pdev->id, playback, capture, &pcm); + if (retval) + return retval; + + if (capture) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &atmel_ac97_capture_ops); + if (playback) + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &atmel_ac97_playback_ops); + + retval = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + &chip->pdev->dev, hw.periods_min * hw.period_bytes_min, + hw.buffer_bytes_max); + if (retval) + return retval; + + pcm->private_data = chip; + pcm->info_flags = 0; + strcpy(pcm->name, chip->card->shortname); + chip->pcm = pcm; + + return 0; +} + +static int atmel_ac97c_mixer_new(struct atmel_ac97c *chip) +{ + struct snd_ac97_template template; + memset(&template, 0, sizeof(template)); + template.private_data = chip; + return snd_ac97_mixer(chip->ac97_bus, &template, &chip->ac97); +} + +static void atmel_ac97c_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct atmel_ac97c *chip = get_chip(ac97); + unsigned long word; + int timeout = 40; + + word = (reg & 0x7f) << 16 | val; + + do { + if (ac97c_readl(chip, COSR) & AC97C_CSR_TXRDY) { + ac97c_writel(chip, COTHR, word); + return; + } + udelay(1); + } while (--timeout); + + dev_dbg(&chip->pdev->dev, "codec write timeout\n"); +} + +static unsigned short atmel_ac97c_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct atmel_ac97c *chip = get_chip(ac97); + unsigned long word; + int timeout = 40; + int write = 10; + + word = (0x80 | (reg & 0x7f)) << 16; + + if ((ac97c_readl(chip, COSR) & AC97C_CSR_RXRDY) != 0) + ac97c_readl(chip, CORHR); + +retry_write: + timeout = 40; + + do { + if ((ac97c_readl(chip, COSR) & AC97C_CSR_TXRDY) != 0) { + ac97c_writel(chip, COTHR, word); + goto read_reg; + } + udelay(10); + } while (--timeout); + + if (!--write) + goto timed_out; + goto retry_write; + +read_reg: + do { + if ((ac97c_readl(chip, COSR) & AC97C_CSR_RXRDY) != 0) { + unsigned short val = ac97c_readl(chip, CORHR); + return val; + } + udelay(10); + } while (--timeout); + + if (!--write) + goto timed_out; + goto retry_write; + +timed_out: + dev_dbg(&chip->pdev->dev, "codec read timeout\n"); + return 0xffff; +} + +static bool filter(struct dma_chan *chan, void *slave) +{ + struct dw_dma_slave *dws = slave; + + if (dws->dma_dev == chan->device->dev) { + chan->private = dws; + return true; + } else + return false; +} + +static void atmel_ac97c_reset(struct atmel_ac97c *chip) +{ + ac97c_writel(chip, MR, 0); + ac97c_writel(chip, MR, AC97C_MR_ENA); + ac97c_writel(chip, CAMR, 0); + ac97c_writel(chip, COMR, 0); + + if (gpio_is_valid(chip->reset_pin)) { + gpio_set_value(chip->reset_pin, 0); + /* AC97 v2.2 specifications says minimum 1 us. */ + udelay(2); + gpio_set_value(chip->reset_pin, 1); + } +} + +static int __devinit atmel_ac97c_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct atmel_ac97c *chip; + struct resource *regs; + struct ac97c_platform_data *pdata; + struct clk *pclk; + static struct snd_ac97_bus_ops ops = { + .write = atmel_ac97c_write, + .read = atmel_ac97c_read, + }; + int retval; + int irq; + + regs = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!regs) { + dev_dbg(&pdev->dev, "no memory resource\n"); + return -ENXIO; + } + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_dbg(&pdev->dev, "no platform data\n"); + return -ENXIO; + } + + irq = platform_get_irq(pdev, 0); + if (irq < 0) { + dev_dbg(&pdev->dev, "could not get irq\n"); + return -ENXIO; + } + + pclk = clk_get(&pdev->dev, "pclk"); + if (IS_ERR(pclk)) { + dev_dbg(&pdev->dev, "no peripheral clock\n"); + return PTR_ERR(pclk); + } + clk_enable(pclk); + + retval = snd_card_create(SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, sizeof(struct atmel_ac97c), &card); + if (retval) { + dev_dbg(&pdev->dev, "could not create sound card device\n"); + goto err_snd_card_new; + } + + chip = get_chip(card); + + retval = request_irq(irq, atmel_ac97c_interrupt, 0, "AC97C", chip); + if (retval) { + dev_dbg(&pdev->dev, "unable to request irq %d\n", irq); + goto err_request_irq; + } + chip->irq = irq; + + spin_lock_init(&chip->lock); + + strcpy(card->driver, "Atmel AC97C"); + strcpy(card->shortname, "Atmel AC97C"); + sprintf(card->longname, "Atmel AC97 controller"); + + chip->card = card; + chip->pclk = pclk; + chip->pdev = pdev; + chip->regs = ioremap(regs->start, regs->end - regs->start + 1); + + if (!chip->regs) { + dev_dbg(&pdev->dev, "could not remap register memory\n"); + goto err_ioremap; + } + + if (gpio_is_valid(pdata->reset_pin)) { + if (gpio_request(pdata->reset_pin, "reset_pin")) { + dev_dbg(&pdev->dev, "reset pin not available\n"); + chip->reset_pin = -ENODEV; + } else { + gpio_direction_output(pdata->reset_pin, 1); + chip->reset_pin = pdata->reset_pin; + } + } + + snd_card_set_dev(card, &pdev->dev); + + atmel_ac97c_reset(chip); + + /* Enable overrun interrupt from codec channel */ + ac97c_writel(chip, COMR, AC97C_CSR_OVRUN); + ac97c_writel(chip, IER, ac97c_readl(chip, IMR) | AC97C_SR_COEVT); + + retval = snd_ac97_bus(card, 0, &ops, chip, &chip->ac97_bus); + if (retval) { + dev_dbg(&pdev->dev, "could not register on ac97 bus\n"); + goto err_ac97_bus; + } + + retval = atmel_ac97c_mixer_new(chip); + if (retval) { + dev_dbg(&pdev->dev, "could not register ac97 mixer\n"); + goto err_ac97_bus; + } + + if (pdata->rx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->rx_dws; + dma_cap_mask_t mask; + + dws->rx_reg = regs->start + AC97C_CARHR + 2; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + chip->dma.rx_chan = dma_request_channel(mask, filter, dws); + + dev_info(&chip->pdev->dev, "using %s for DMA RX\n", + dev_name(&chip->dma.rx_chan->dev->device)); + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + } + + if (pdata->tx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->tx_dws; + dma_cap_mask_t mask; + + dws->tx_reg = regs->start + AC97C_CATHR + 2; + + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); + + chip->dma.tx_chan = dma_request_channel(mask, filter, dws); + + dev_info(&chip->pdev->dev, "using %s for DMA TX\n", + dev_name(&chip->dma.tx_chan->dev->device)); + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + } + + if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && + !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { + dev_dbg(&pdev->dev, "DMA not available\n"); + retval = -ENODEV; + goto err_dma; + } + + retval = atmel_ac97c_pcm_new(chip); + if (retval) { + dev_dbg(&pdev->dev, "could not register ac97 pcm device\n"); + goto err_dma; + } + + retval = snd_card_register(card); + if (retval) { + dev_dbg(&pdev->dev, "could not register sound card\n"); + goto err_dma; + } + + platform_set_drvdata(pdev, card); + + dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p\n", + chip->regs); + + return 0; + +err_dma: + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; +err_ac97_bus: + snd_card_set_dev(card, NULL); + + if (gpio_is_valid(chip->reset_pin)) + gpio_free(chip->reset_pin); + + iounmap(chip->regs); +err_ioremap: + free_irq(irq, chip); +err_request_irq: + snd_card_free(card); +err_snd_card_new: + clk_disable(pclk); + clk_put(pclk); + return retval; +} + +#ifdef CONFIG_PM +static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_ac97c *chip = card->private_data; + + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.tx_chan); + clk_disable(chip->pclk); + + return 0; +} + +static int atmel_ac97c_resume(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_ac97c *chip = card->private_data; + + clk_enable(chip->pclk); + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.tx_chan); + + return 0; +} +#else +#define atmel_ac97c_suspend NULL +#define atmel_ac97c_resume NULL +#endif + +static int __devexit atmel_ac97c_remove(struct platform_device *pdev) +{ + struct snd_card *card = platform_get_drvdata(pdev); + struct atmel_ac97c *chip = get_chip(card); + + if (gpio_is_valid(chip->reset_pin)) + gpio_free(chip->reset_pin); + + ac97c_writel(chip, CAMR, 0); + ac97c_writel(chip, COMR, 0); + ac97c_writel(chip, MR, 0); + + clk_disable(chip->pclk); + clk_put(chip->pclk); + iounmap(chip->regs); + free_irq(chip->irq, chip); + + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + + snd_card_set_dev(card, NULL); + snd_card_free(card); + + platform_set_drvdata(pdev, NULL); + + return 0; +} + +static struct platform_driver atmel_ac97c_driver = { + .remove = __devexit_p(atmel_ac97c_remove), + .driver = { + .name = "atmel_ac97c", + }, + .suspend = atmel_ac97c_suspend, + .resume = atmel_ac97c_resume, +}; + +static int __init atmel_ac97c_init(void) +{ + return platform_driver_probe(&atmel_ac97c_driver, + atmel_ac97c_probe); +} +module_init(atmel_ac97c_init); + +static void __exit atmel_ac97c_exit(void) +{ + platform_driver_unregister(&atmel_ac97c_driver); +} +module_exit(atmel_ac97c_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Driver for Atmel AC97 controller"); +MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>"); diff --git a/sound/atmel/ac97c.h b/sound/atmel/ac97c.h new file mode 100644 index 0000000..ecbba50 --- /dev/null +++ b/sound/atmel/ac97c.h @@ -0,0 +1,73 @@ +/* + * Register definitions for Atmel AC97C + * + * Copyright (C) 2005-2009 Atmel Corporation + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ +#ifndef __SOUND_ATMEL_AC97C_H +#define __SOUND_ATMEL_AC97C_H + +#define AC97C_MR 0x08 +#define AC97C_ICA 0x10 +#define AC97C_OCA 0x14 +#define AC97C_CARHR 0x20 +#define AC97C_CATHR 0x24 +#define AC97C_CASR 0x28 +#define AC97C_CAMR 0x2c +#define AC97C_CORHR 0x40 +#define AC97C_COTHR 0x44 +#define AC97C_COSR 0x48 +#define AC97C_COMR 0x4c +#define AC97C_SR 0x50 +#define AC97C_IER 0x54 +#define AC97C_IDR 0x58 +#define AC97C_IMR 0x5c +#define AC97C_VERSION 0xfc + +#define AC97C_CATPR PDC_TPR +#define AC97C_CATCR PDC_TCR +#define AC97C_CATNPR PDC_TNPR +#define AC97C_CATNCR PDC_TNCR +#define AC97C_CARPR PDC_RPR +#define AC97C_CARCR PDC_RCR +#define AC97C_CARNPR PDC_RNPR +#define AC97C_CARNCR PDC_RNCR +#define AC97C_PTCR PDC_PTCR + +#define AC97C_MR_ENA (1 << 0) +#define AC97C_MR_WRST (1 << 1) +#define AC97C_MR_VRA (1 << 2) + +#define AC97C_CSR_TXRDY (1 << 0) +#define AC97C_CSR_TXEMPTY (1 << 1) +#define AC97C_CSR_UNRUN (1 << 2) +#define AC97C_CSR_RXRDY (1 << 4) +#define AC97C_CSR_OVRUN (1 << 5) +#define AC97C_CSR_ENDTX (1 << 10) +#define AC97C_CSR_ENDRX (1 << 14) + +#define AC97C_CMR_SIZE_20 (0 << 16) +#define AC97C_CMR_SIZE_18 (1 << 16) +#define AC97C_CMR_SIZE_16 (2 << 16) +#define AC97C_CMR_SIZE_10 (3 << 16) +#define AC97C_CMR_CEM_LITTLE (1 << 18) +#define AC97C_CMR_CEM_BIG (0 << 18) +#define AC97C_CMR_CENA (1 << 21) +#define AC97C_CMR_DMAEN (1 << 22) + +#define AC97C_SR_CAEVT (1 << 3) +#define AC97C_SR_COEVT (1 << 2) +#define AC97C_SR_WKUP (1 << 1) +#define AC97C_SR_SOF (1 << 0) + +#define AC97C_CH_MASK(slot) \ + (0x7 << (3 * (AC97_SLOT_##slot - 3))) +#define AC97C_CH_ASSIGN(slot, channel) \ + (AC97C_CHANNEL_##channel << (3 * (AC97_SLOT_##slot - 3))) +#define AC97C_CHANNEL_NONE 0x0 +#define AC97C_CHANNEL_A 0x1 + +#endif /* __SOUND_ATMEL_AC97C_H */ diff --git a/sound/core/control.c b/sound/core/control.c index 636b3b5..4b20fa2 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -1373,12 +1373,9 @@ EXPORT_SYMBOL(snd_ctl_unregister_ioctl_compat); static int snd_ctl_fasync(int fd, struct file * file, int on) { struct snd_ctl_file *ctl; - int err; + ctl = file->private_data; - err = fasync_helper(fd, file, on, &ctl->fasync); - if (err < 0) - return err; - return 0; + return fasync_helper(fd, file, on, &ctl->fasync); } /* diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 195cafc..a70ee7f 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -99,9 +99,6 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) if (hw == NULL) return -ENODEV; - if (!hw->ops.open) - return -ENXIO; - if (!try_module_get(hw->card->module)) return -EFAULT; @@ -113,6 +110,10 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) err = -EBUSY; break; } + if (!hw->ops.open) { + err = 0; + break; + } err = hw->ops.open(hw, file); if (err >= 0) break; @@ -151,7 +152,7 @@ static int snd_hwdep_open(struct inode *inode, struct file * file) static int snd_hwdep_release(struct inode *inode, struct file * file) { - int err = -ENXIO; + int err = 0; struct snd_hwdep *hw = file->private_data; struct module *mod = hw->card->module; diff --git a/sound/core/info.c b/sound/core/info.c index 70fa871..35df614 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -154,11 +154,6 @@ EXPORT_SYMBOL(snd_seq_root); struct snd_info_entry *snd_oss_root; #endif -static inline void snd_info_entry_prepare(struct proc_dir_entry *de) -{ - de->owner = THIS_MODULE; -} - static void snd_remove_proc_entry(struct proc_dir_entry *parent, struct proc_dir_entry *de) { @@ -522,32 +517,11 @@ static const struct file_operations snd_info_entry_operations = .release = snd_info_entry_release, }; -/** - * snd_create_proc_entry - create a procfs entry - * @name: the name of the proc file - * @mode: the file permission bits, S_Ixxx - * @parent: the parent proc-directory entry - * - * Creates a new proc file entry with the given name and permission - * on the given directory. - * - * Returns the pointer of new instance or NULL on failure. - */ -static struct proc_dir_entry *snd_create_proc_entry(const char *name, mode_t mode, - struct proc_dir_entry *parent) -{ - struct proc_dir_entry *p; - p = create_proc_entry(name, mode, parent); - if (p) - snd_info_entry_prepare(p); - return p; -} - int __init snd_info_init(void) { struct proc_dir_entry *p; - p = snd_create_proc_entry("asound", S_IFDIR | S_IRUGO | S_IXUGO, NULL); + p = create_proc_entry("asound", S_IFDIR | S_IRUGO | S_IXUGO, NULL); if (p == NULL) return -ENOMEM; snd_proc_root = p; @@ -974,12 +948,11 @@ int snd_info_register(struct snd_info_entry * entry) return -ENXIO; root = entry->parent == NULL ? snd_proc_root : entry->parent->p; mutex_lock(&info_mutex); - p = snd_create_proc_entry(entry->name, entry->mode, root); + p = create_proc_entry(entry->name, entry->mode, root); if (!p) { mutex_unlock(&info_mutex); return -ENOMEM; } - p->owner = entry->module; if (!S_ISDIR(entry->mode)) p->proc_fops = &snd_info_entry_operations; p->size = entry->size; diff --git a/sound/core/init.c b/sound/core/init.c index dc4b80c..fd56afe 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -208,6 +208,7 @@ int snd_card_create(int idx, const char *xid, INIT_LIST_HEAD(&card->controls); INIT_LIST_HEAD(&card->ctl_files); spin_lock_init(&card->files_lock); + INIT_LIST_HEAD(&card->files_list); init_waitqueue_head(&card->shutdown_sleep); #ifdef CONFIG_PM mutex_init(&card->power_lock); @@ -274,6 +275,7 @@ static int snd_disconnect_release(struct inode *inode, struct file *file) list_for_each_entry(_df, &shutdown_files, shutdown_list) { if (_df->file == file) { df = _df; + list_del_init(&df->shutdown_list); break; } } @@ -362,8 +364,7 @@ int snd_card_disconnect(struct snd_card *card) /* phase 2: replace file->f_op with special dummy operations */ spin_lock(&card->files_lock); - mfile = card->files; - while (mfile) { + list_for_each_entry(mfile, &card->files_list, list) { file = mfile->file; /* it's critical part, use endless loop */ @@ -376,8 +377,6 @@ int snd_card_disconnect(struct snd_card *card) mfile->file->f_op = &snd_shutdown_f_ops; fops_get(mfile->file->f_op); - - mfile = mfile->next; } spin_unlock(&card->files_lock); @@ -457,7 +456,7 @@ int snd_card_free_when_closed(struct snd_card *card) return ret; spin_lock(&card->files_lock); - if (card->files == NULL) + if (list_empty(&card->files_list)) free_now = 1; else card->free_on_last_close = 1; @@ -477,7 +476,7 @@ int snd_card_free(struct snd_card *card) return ret; /* wait, until all devices are ready for the free operation */ - wait_event(card->shutdown_sleep, card->files == NULL); + wait_event(card->shutdown_sleep, list_empty(&card->files_list)); snd_card_do_free(card); return 0; } @@ -824,15 +823,13 @@ int snd_card_file_add(struct snd_card *card, struct file *file) return -ENOMEM; mfile->file = file; mfile->disconnected_f_op = NULL; - mfile->next = NULL; spin_lock(&card->files_lock); if (card->shutdown) { spin_unlock(&card->files_lock); kfree(mfile); return -ENODEV; } - mfile->next = card->files; - card->files = mfile; + list_add(&mfile->list, &card->files_list); spin_unlock(&card->files_lock); return 0; } @@ -854,29 +851,20 @@ EXPORT_SYMBOL(snd_card_file_add); */ int snd_card_file_remove(struct snd_card *card, struct file *file) { - struct snd_monitor_file *mfile, *pfile = NULL; + struct snd_monitor_file *mfile, *found = NULL; int last_close = 0; spin_lock(&card->files_lock); - mfile = card->files; - while (mfile) { + list_for_each_entry(mfile, &card->files_list, list) { if (mfile->file == file) { - if (pfile) - pfile->next = mfile->next; - else - card->files = mfile->next; + list_del(&mfile->list); + if (mfile->disconnected_f_op) + fops_put(mfile->disconnected_f_op); + found = mfile; break; } - pfile = mfile; - mfile = mfile->next; - } - if (mfile && mfile->disconnected_f_op) { - fops_put(mfile->disconnected_f_op); - spin_lock(&shutdown_lock); - list_del(&mfile->shutdown_list); - spin_unlock(&shutdown_lock); } - if (card->files == NULL) + if (list_empty(&card->files_list)) last_close = 1; spin_unlock(&card->files_lock); if (last_close) { @@ -884,11 +872,11 @@ int snd_card_file_remove(struct snd_card *card, struct file *file) if (card->free_on_last_close) snd_card_do_free(card); } - if (!mfile) { + if (!found) { snd_printk(KERN_ERR "ALSA card file remove problem (%p)\n", file); return -ENOENT; } - kfree(mfile); + kfree(found); return 0; } diff --git a/sound/core/jack.c b/sound/core/jack.c index dd4a12d..c8254c6 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -23,6 +23,14 @@ #include <sound/jack.h> #include <sound/core.h> +static int jack_types[] = { + SW_HEADPHONE_INSERT, + SW_MICROPHONE_INSERT, + SW_LINEOUT_INSERT, + SW_JACK_PHYSICAL_INSERT, + SW_VIDEOOUT_INSERT, +}; + static int snd_jack_dev_free(struct snd_device *device) { struct snd_jack *jack = device->device_data; @@ -47,7 +55,7 @@ static int snd_jack_dev_register(struct snd_device *device) int err; snprintf(jack->name, sizeof(jack->name), "%s %s", - card->longname, jack->id); + card->shortname, jack->id); jack->input_dev->name = jack->name; /* Default to the sound card device. */ @@ -79,6 +87,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, { struct snd_jack *jack; int err; + int i; static struct snd_device_ops ops = { .dev_free = snd_jack_dev_free, .dev_register = snd_jack_dev_register, @@ -100,18 +109,10 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, jack->type = type; - if (type & SND_JACK_HEADPHONE) - input_set_capability(jack->input_dev, EV_SW, - SW_HEADPHONE_INSERT); - if (type & SND_JACK_LINEOUT) - input_set_capability(jack->input_dev, EV_SW, - SW_LINEOUT_INSERT); - if (type & SND_JACK_MICROPHONE) - input_set_capability(jack->input_dev, EV_SW, - SW_MICROPHONE_INSERT); - if (type & SND_JACK_MECHANICAL) - input_set_capability(jack->input_dev, EV_SW, - SW_JACK_PHYSICAL_INSERT); + for (i = 0; i < ARRAY_SIZE(jack_types); i++) + if (type & (1 << i)) + input_set_capability(jack->input_dev, EV_SW, + jack_types[i]); err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -154,21 +155,17 @@ EXPORT_SYMBOL(snd_jack_set_parent); */ void snd_jack_report(struct snd_jack *jack, int status) { + int i; + if (!jack) return; - if (jack->type & SND_JACK_HEADPHONE) - input_report_switch(jack->input_dev, SW_HEADPHONE_INSERT, - status & SND_JACK_HEADPHONE); - if (jack->type & SND_JACK_LINEOUT) - input_report_switch(jack->input_dev, SW_LINEOUT_INSERT, - status & SND_JACK_LINEOUT); - if (jack->type & SND_JACK_MICROPHONE) - input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, - status & SND_JACK_MICROPHONE); - if (jack->type & SND_JACK_MECHANICAL) - input_report_switch(jack->input_dev, SW_JACK_PHYSICAL_INSERT, - status & SND_JACK_MECHANICAL); + for (i = 0; i < ARRAY_SIZE(jack_types); i++) { + int testbit = 1 << i; + if (jack->type & testbit) + input_report_switch(jack->input_dev, jack_types[i], + status & testbit); + } input_sync(jack->input_dev); } diff --git a/sound/core/misc.c b/sound/core/misc.c index 38524f6..a9710e0 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -95,12 +95,14 @@ snd_pci_quirk_lookup(struct pci_dev *pci, const struct snd_pci_quirk *list) { const struct snd_pci_quirk *q; - for (q = list; q->subvendor; q++) - if (q->subvendor == pci->subsystem_vendor && - (!q->subdevice || q->subdevice == pci->subsystem_device)) + for (q = list; q->subvendor; q++) { + if (q->subvendor != pci->subsystem_vendor) + continue; + if (!q->subdevice || + (pci->subsystem_device & q->subdevice_mask) == q->subdevice) return q; + } return NULL; } - EXPORT_SYMBOL(snd_pci_quirk_lookup); #endif diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c index 4690b8b..5dcd8a5 100644 --- a/sound/core/oss/mixer_oss.c +++ b/sound/core/oss/mixer_oss.c @@ -692,6 +692,9 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PVOLUME], left, right); if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CVOLUME) { + snd_mixer_oss_put_volume1_vol(fmixer, pslot, + slot->numid[SNDRV_MIXER_OSS_ITEM_CVOLUME], left, right); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GVOLUME) { snd_mixer_oss_put_volume1_vol(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GVOLUME], left, right); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GLOBAL) { @@ -700,19 +703,27 @@ static int snd_mixer_oss_put_volume1(struct snd_mixer_oss_file *fmixer, if (left || right) { if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0); + if (slot->present & SNDRV_MIXER_OSS_PRESENT_CSWITCH) + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], left, right, 0); if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0); if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1); + if (slot->present & SNDRV_MIXER_OSS_PRESENT_CROUTE) + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], left, right, 1); if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1); } else { if (slot->present & SNDRV_MIXER_OSS_PRESENT_PSWITCH) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PSWITCH], left, right, 0); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CSWITCH) { + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CSWITCH], left, right, 0); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GSWITCH) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GSWITCH], left, right, 0); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_PROUTE) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_PROUTE], left, right, 1); + } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_CROUTE) { + snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_CROUTE], left, right, 1); } else if (slot->present & SNDRV_MIXER_OSS_PRESENT_GROUTE) { snd_mixer_oss_put_volume1_sw(fmixer, pslot, slot->numid[SNDRV_MIXER_OSS_ITEM_GROUTE], left, right, 1); } diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index e178366..dda000b 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1160,9 +1160,11 @@ snd_pcm_sframes_t snd_pcm_oss_write3(struct snd_pcm_substream *substream, const runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: write: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: write: " + "recovering from XRUN\n"); else - printk("pcm_oss: write: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: write: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_oss_prepare(substream); if (ret < 0) @@ -1196,9 +1198,11 @@ snd_pcm_sframes_t snd_pcm_oss_read3(struct snd_pcm_substream *substream, char *p runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: read: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: read: " + "recovering from XRUN\n"); else - printk("pcm_oss: read: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: read: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); if (ret < 0) @@ -1242,9 +1246,11 @@ snd_pcm_sframes_t snd_pcm_oss_writev3(struct snd_pcm_substream *substream, void runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: writev: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: writev: " + "recovering from XRUN\n"); else - printk("pcm_oss: writev: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: writev: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_oss_prepare(substream); if (ret < 0) @@ -1278,9 +1284,11 @@ snd_pcm_sframes_t snd_pcm_oss_readv3(struct snd_pcm_substream *substream, void * runtime->status->state == SNDRV_PCM_STATE_SUSPENDED) { #ifdef OSS_DEBUG if (runtime->status->state == SNDRV_PCM_STATE_XRUN) - printk("pcm_oss: readv: recovering from XRUN\n"); + printk(KERN_DEBUG "pcm_oss: readv: " + "recovering from XRUN\n"); else - printk("pcm_oss: readv: recovering from SUSPEND\n"); + printk(KERN_DEBUG "pcm_oss: readv: " + "recovering from SUSPEND\n"); #endif ret = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DRAIN, NULL); if (ret < 0) @@ -1533,7 +1541,7 @@ static int snd_pcm_oss_sync1(struct snd_pcm_substream *substream, size_t size) init_waitqueue_entry(&wait, current); add_wait_queue(&runtime->sleep, &wait); #ifdef OSS_DEBUG - printk("sync1: size = %li\n", size); + printk(KERN_DEBUG "sync1: size = %li\n", size); #endif while (1) { result = snd_pcm_oss_write2(substream, runtime->oss.buffer, size, 1); @@ -1590,7 +1598,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) mutex_lock(&runtime->oss.params_lock); if (runtime->oss.buffer_used > 0) { #ifdef OSS_DEBUG - printk("sync: buffer_used\n"); + printk(KERN_DEBUG "sync: buffer_used\n"); #endif size = (8 * (runtime->oss.period_bytes - runtime->oss.buffer_used) + 7) / width; snd_pcm_format_set_silence(format, @@ -1603,7 +1611,7 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) } } else if (runtime->oss.period_ptr > 0) { #ifdef OSS_DEBUG - printk("sync: period_ptr\n"); + printk(KERN_DEBUG "sync: period_ptr\n"); #endif size = runtime->oss.period_bytes - runtime->oss.period_ptr; snd_pcm_format_set_silence(format, @@ -1767,7 +1775,7 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) AFMT_S8 | AFMT_U16_LE | AFMT_U16_BE | AFMT_S32_LE | AFMT_S32_BE | - AFMT_S24_LE | AFMT_S24_LE | + AFMT_S24_LE | AFMT_S24_BE | AFMT_S24_PACKED; params = kmalloc(sizeof(*params), GFP_KERNEL); if (!params) @@ -1895,7 +1903,9 @@ static int snd_pcm_oss_set_fragment(struct snd_pcm_oss_file *pcm_oss_file, unsig static int snd_pcm_oss_nonblock(struct file * file) { + spin_lock(&file->f_lock); file->f_flags |= O_NONBLOCK; + spin_unlock(&file->f_lock); return 0; } @@ -1952,7 +1962,7 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr int err, cmd; #ifdef OSS_DEBUG - printk("pcm_oss: trigger = 0x%x\n", trigger); + printk(KERN_DEBUG "pcm_oss: trigger = 0x%x\n", trigger); #endif psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; @@ -2170,7 +2180,9 @@ static int snd_pcm_oss_get_space(struct snd_pcm_oss_file *pcm_oss_file, int stre } #ifdef OSS_DEBUG - printk("pcm_oss: space: bytes = %i, fragments = %i, fragstotal = %i, fragsize = %i\n", info.bytes, info.fragments, info.fragstotal, info.fragsize); + printk(KERN_DEBUG "pcm_oss: space: bytes = %i, fragments = %i, " + "fragstotal = %i, fragsize = %i\n", + info.bytes, info.fragments, info.fragstotal, info.fragsize); #endif if (copy_to_user(_info, &info, sizeof(info))) return -EFAULT; @@ -2473,7 +2485,7 @@ static long snd_pcm_oss_ioctl(struct file *file, unsigned int cmd, unsigned long if (((cmd >> 8) & 0xff) != 'P') return -EINVAL; #ifdef OSS_DEBUG - printk("pcm_oss: ioctl = 0x%x\n", cmd); + printk(KERN_DEBUG "pcm_oss: ioctl = 0x%x\n", cmd); #endif switch (cmd) { case SNDCTL_DSP_RESET: @@ -2627,7 +2639,8 @@ static ssize_t snd_pcm_oss_read(struct file *file, char __user *buf, size_t coun #else { ssize_t res = snd_pcm_oss_read1(substream, buf, count); - printk("pcm_oss: read %li bytes (returned %li bytes)\n", (long)count, (long)res); + printk(KERN_DEBUG "pcm_oss: read %li bytes " + "(returned %li bytes)\n", (long)count, (long)res); return res; } #endif @@ -2646,7 +2659,8 @@ static ssize_t snd_pcm_oss_write(struct file *file, const char __user *buf, size substream->f_flags = file->f_flags & O_NONBLOCK; result = snd_pcm_oss_write1(substream, buf, count); #ifdef OSS_DEBUG - printk("pcm_oss: write %li bytes (wrote %li bytes)\n", (long)count, (long)result); + printk(KERN_DEBUG "pcm_oss: write %li bytes (wrote %li bytes)\n", + (long)count, (long)result); #endif return result; } @@ -2720,7 +2734,7 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) int err; #ifdef OSS_DEBUG - printk("pcm_oss: mmap begin\n"); + printk(KERN_DEBUG "pcm_oss: mmap begin\n"); #endif pcm_oss_file = file->private_data; switch ((area->vm_flags & (VM_READ | VM_WRITE))) { @@ -2770,7 +2784,8 @@ static int snd_pcm_oss_mmap(struct file *file, struct vm_area_struct *area) runtime->silence_threshold = 0; runtime->silence_size = 0; #ifdef OSS_DEBUG - printk("pcm_oss: mmap ok, bytes = 0x%x\n", runtime->oss.mmap_bytes); + printk(KERN_DEBUG "pcm_oss: mmap ok, bytes = 0x%x\n", + runtime->oss.mmap_bytes); #endif /* In mmap mode we never stop */ runtime->stop_threshold = runtime->boundary; @@ -2872,7 +2887,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, setup = kmalloc(sizeof(*setup), GFP_KERNEL); if (! setup) { buffer->error = -ENOMEM; - mutex_lock(&pstr->oss.setup_mutex); + mutex_unlock(&pstr->oss.setup_mutex); return; } if (pstr->oss.setup_list == NULL) @@ -2886,7 +2901,7 @@ static void snd_pcm_oss_proc_write(struct snd_info_entry *entry, if (! template.task_name) { kfree(setup); buffer->error = -ENOMEM; - mutex_lock(&pstr->oss.setup_mutex); + mutex_unlock(&pstr->oss.setup_mutex); return; } } diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index ca2f4c3..b9afab6 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -176,9 +176,9 @@ static inline int snd_pcm_plug_slave_format(int format, struct snd_mask *format_ #endif #ifdef PLUGIN_DEBUG -#define pdprintf( fmt, args... ) printk( "plugin: " fmt, ##args) +#define pdprintf(fmt, args...) printk(KERN_DEBUG "plugin: " fmt, ##args) #else -#define pdprintf( fmt, args... ) +#define pdprintf(fmt, args...) #endif #endif /* __PCM_PLUGIN_H */ diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index a466443..2fa9299 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -157,7 +157,7 @@ static void resample_shrink(struct snd_pcm_plugin *plugin, while (dst_frames1 > 0) { S1 = S2; if (src_frames1-- > 0) { - S1 = *src; + S2 = *src; src += src_step; } if (pos & ~R_MASK) { diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 192a433..145931a 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -667,7 +667,6 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) spin_lock_init(&substream->self_group.lock); INIT_LIST_HEAD(&substream->self_group.substreams); list_add_tail(&substream->link_list, &substream->self_group.substreams); - spin_lock_init(&substream->timer_lock); atomic_set(&substream->mmap_count, 0); prev = substream; } @@ -692,7 +691,7 @@ EXPORT_SYMBOL(snd_pcm_new_stream); * * Returns zero if successful, or a negative error code on failure. */ -int snd_pcm_new(struct snd_card *card, char *id, int device, +int snd_pcm_new(struct snd_card *card, const char *id, int device, int playback_count, int capture_count, struct snd_pcm ** rpcm) { diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 9216910..fbb2e39 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -125,23 +125,32 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG +#define xrun_debug(substream) ((substream)->pstr->xrun_debug) +#else +#define xrun_debug(substream) 0 +#endif + +#define dump_stack_on_xrun(substream) do { \ + if (xrun_debug(substream) > 1) \ + dump_stack(); \ + } while (0) + static void xrun(struct snd_pcm_substream *substream) { snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (substream->pstr->xrun_debug) { + if (xrun_debug(substream)) { snd_printd(KERN_DEBUG "XRUN: pcmC%dD%d%c\n", substream->pcm->card->number, substream->pcm->device, substream->stream ? 'c' : 'p'); - if (substream->pstr->xrun_debug > 1) - dump_stack(); + dump_stack_on_xrun(substream); } -#endif } -static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) +static snd_pcm_uframes_t +snd_pcm_update_hw_ptr_pos(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t pos; @@ -150,17 +159,21 @@ static inline snd_pcm_uframes_t snd_pcm_update_hw_ptr_pos(struct snd_pcm_substre pos = substream->ops->pointer(substream); if (pos == SNDRV_PCM_POS_XRUN) return pos; /* XRUN */ -#ifdef CONFIG_SND_DEBUG if (pos >= runtime->buffer_size) { - snd_printk(KERN_ERR "BUG: stream = %i, pos = 0x%lx, buffer size = 0x%lx, period size = 0x%lx\n", substream->stream, pos, runtime->buffer_size, runtime->period_size); + if (printk_ratelimit()) { + snd_printd(KERN_ERR "BUG: stream = %i, pos = 0x%lx, " + "buffer size = 0x%lx, period size = 0x%lx\n", + substream->stream, pos, runtime->buffer_size, + runtime->period_size); + } + pos = 0; } -#endif pos -= pos % runtime->min_align; return pos; } -static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, - struct snd_pcm_runtime *runtime) +static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, + struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t avail; @@ -182,11 +195,21 @@ static inline int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream return 0; } -static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) +#define hw_ptr_error(substream, fmt, args...) \ + do { \ + if (xrun_debug(substream)) { \ + if (printk_ratelimit()) { \ + snd_printd("PCM: " fmt, ##args); \ + } \ + dump_stack_on_xrun(substream); \ + } \ + } while (0) + +static int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt; + snd_pcm_uframes_t new_hw_ptr, hw_ptr_interrupt, hw_base; snd_pcm_sframes_t delta; pos = snd_pcm_update_hw_ptr_pos(substream, runtime); @@ -194,36 +217,53 @@ static inline int snd_pcm_update_hw_ptr_interrupt(struct snd_pcm_substream *subs xrun(substream); return -EPIPE; } - if (runtime->period_size == runtime->buffer_size) - goto __next_buf; - new_hw_ptr = runtime->hw_ptr_base + pos; + hw_base = runtime->hw_ptr_base; + new_hw_ptr = hw_base + pos; hw_ptr_interrupt = runtime->hw_ptr_interrupt + runtime->period_size; - - delta = hw_ptr_interrupt - new_hw_ptr; - if (delta > 0) { - if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->periods > 1 && substream->pstr->xrun_debug) { - snd_printd(KERN_ERR "Unexpected hw_pointer value [1] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); - if (substream->pstr->xrun_debug > 1) - dump_stack(); - } -#endif - return 0; + delta = new_hw_ptr - hw_ptr_interrupt; + if (hw_ptr_interrupt >= runtime->boundary) { + hw_ptr_interrupt -= runtime->boundary; + if (hw_base < runtime->boundary / 2) + /* hw_base was already lapped; recalc delta */ + delta = new_hw_ptr - hw_ptr_interrupt; + } + if (delta < 0) { + delta += runtime->buffer_size; + if (delta < 0) { + hw_ptr_error(substream, + "Unexpected hw_pointer value " + "(stream=%i, pos=%ld, intr_ptr=%ld)\n", + substream->stream, (long)pos, + (long)hw_ptr_interrupt); + /* rebase to interrupt position */ + hw_base = new_hw_ptr = hw_ptr_interrupt; + /* align hw_base to buffer_size */ + hw_base -= hw_base % runtime->buffer_size; + delta = 0; + } else { + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; } - __next_buf: - runtime->hw_ptr_base += runtime->buffer_size; - if (runtime->hw_ptr_base == runtime->boundary) - runtime->hw_ptr_base = 0; - new_hw_ptr = runtime->hw_ptr_base + pos; } - + if (delta > runtime->period_size) { + hw_ptr_error(substream, + "Lost interrupts? " + "(stream=%i, delta=%ld, intr_ptr=%ld)\n", + substream->stream, (long)delta, + (long)hw_ptr_interrupt); + /* rebase hw_ptr_interrupt */ + hw_ptr_interrupt = + new_hw_ptr - new_hw_ptr % runtime->period_size; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; - runtime->hw_ptr_interrupt = new_hw_ptr - new_hw_ptr % runtime->period_size; + runtime->hw_ptr_interrupt = hw_ptr_interrupt; return snd_pcm_update_hw_ptr_post(substream, runtime); } @@ -233,7 +273,7 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_uframes_t pos; - snd_pcm_uframes_t old_hw_ptr, new_hw_ptr; + snd_pcm_uframes_t old_hw_ptr, new_hw_ptr, hw_base; snd_pcm_sframes_t delta; old_hw_ptr = runtime->status->hw_ptr; @@ -242,29 +282,38 @@ int snd_pcm_update_hw_ptr(struct snd_pcm_substream *substream) xrun(substream); return -EPIPE; } - new_hw_ptr = runtime->hw_ptr_base + pos; - - delta = old_hw_ptr - new_hw_ptr; - if (delta > 0) { - if ((snd_pcm_uframes_t)delta < runtime->buffer_size / 2) { -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - if (runtime->periods > 2 && substream->pstr->xrun_debug) { - snd_printd(KERN_ERR "Unexpected hw_pointer value [2] (stream = %i, delta: -%ld, max jitter = %ld): wrong interrupt acknowledge?\n", substream->stream, (long) delta, runtime->buffer_size / 2); - if (substream->pstr->xrun_debug > 1) - dump_stack(); - } -#endif + hw_base = runtime->hw_ptr_base; + new_hw_ptr = hw_base + pos; + + delta = new_hw_ptr - old_hw_ptr; + if (delta < 0) { + delta += runtime->buffer_size; + if (delta < 0) { + hw_ptr_error(substream, + "Unexpected hw_pointer value [2] " + "(stream=%i, pos=%ld, old_ptr=%ld)\n", + substream->stream, (long)pos, + (long)old_hw_ptr); return 0; } - runtime->hw_ptr_base += runtime->buffer_size; - if (runtime->hw_ptr_base == runtime->boundary) - runtime->hw_ptr_base = 0; - new_hw_ptr = runtime->hw_ptr_base + pos; + hw_base += runtime->buffer_size; + if (hw_base >= runtime->boundary) + hw_base = 0; + new_hw_ptr = hw_base + pos; + } + if (delta > runtime->period_size && runtime->periods > 1) { + hw_ptr_error(substream, + "hw_ptr skipping! " + "(pos=%ld, delta=%ld, period=%ld)\n", + (long)pos, (long)delta, + (long)runtime->period_size); + return 0; } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && runtime->silence_size > 0) snd_pcm_playback_silence(substream, new_hw_ptr); + runtime->hw_ptr_base = hw_base; runtime->status->hw_ptr = new_hw_ptr; return snd_pcm_update_hw_ptr_post(substream, runtime); diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a789efc..a151fb0 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -186,7 +186,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!(params->rmask & (1 << k))) continue; #ifdef RULES_DEBUG - printk("%s = ", snd_pcm_hw_param_names[k]); + printk(KERN_DEBUG "%s = ", snd_pcm_hw_param_names[k]); printk("%04x%04x%04x%04x -> ", m->bits[3], m->bits[2], m->bits[1], m->bits[0]); #endif changed = snd_mask_refine(m, constrs_mask(constrs, k)); @@ -206,7 +206,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!(params->rmask & (1 << k))) continue; #ifdef RULES_DEBUG - printk("%s = ", snd_pcm_hw_param_names[k]); + printk(KERN_DEBUG "%s = ", snd_pcm_hw_param_names[k]); if (i->empty) printk("empty"); else @@ -251,7 +251,7 @@ int snd_pcm_hw_refine(struct snd_pcm_substream *substream, if (!doit) continue; #ifdef RULES_DEBUG - printk("Rule %d [%p]: ", k, r->func); + printk(KERN_DEBUG "Rule %d [%p]: ", k, r->func); if (r->var >= 0) { printk("%s = ", snd_pcm_hw_param_names[r->var]); if (hw_is_mask(r->var)) { @@ -3246,9 +3246,7 @@ static int snd_pcm_fasync(int fd, struct file * file, int on) err = fasync_helper(fd, file, on, &runtime->fasync); out: unlock_kernel(); - if (err < 0) - return err; - return 0; + return err; } /* diff --git a/sound/core/pcm_timer.c b/sound/core/pcm_timer.c index 2c89c04..ca8068b 100644 --- a/sound/core/pcm_timer.c +++ b/sound/core/pcm_timer.c @@ -85,25 +85,19 @@ static unsigned long snd_pcm_timer_resolution(struct snd_timer * timer) static int snd_pcm_timer_start(struct snd_timer * timer) { - unsigned long flags; struct snd_pcm_substream *substream; substream = snd_timer_chip(timer); - spin_lock_irqsave(&substream->timer_lock, flags); substream->timer_running = 1; - spin_unlock_irqrestore(&substream->timer_lock, flags); return 0; } static int snd_pcm_timer_stop(struct snd_timer * timer) { - unsigned long flags; struct snd_pcm_substream *substream; substream = snd_timer_chip(timer); - spin_lock_irqsave(&substream->timer_lock, flags); substream->timer_running = 0; - spin_unlock_irqrestore(&substream->timer_lock, flags); return 0; } diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 002777b..473247c 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -224,156 +224,143 @@ int snd_rawmidi_drain_input(struct snd_rawmidi_substream *substream) return 0; } -int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, - int mode, struct snd_rawmidi_file * rfile) +/* look for an available substream for the given stream direction; + * if a specific subdevice is given, try to assign it + */ +static int assign_substream(struct snd_rawmidi *rmidi, int subdevice, + int stream, int mode, + struct snd_rawmidi_substream **sub_ret) +{ + struct snd_rawmidi_substream *substream; + struct snd_rawmidi_str *s = &rmidi->streams[stream]; + static unsigned int info_flags[2] = { + [SNDRV_RAWMIDI_STREAM_OUTPUT] = SNDRV_RAWMIDI_INFO_OUTPUT, + [SNDRV_RAWMIDI_STREAM_INPUT] = SNDRV_RAWMIDI_INFO_INPUT, + }; + + if (!(rmidi->info_flags & info_flags[stream])) + return -ENXIO; + if (subdevice >= 0 && subdevice >= s->substream_count) + return -ENODEV; + if (s->substream_opened >= s->substream_count) + return -EAGAIN; + + list_for_each_entry(substream, &s->substreams, list) { + if (substream->opened) { + if (stream == SNDRV_RAWMIDI_STREAM_INPUT || + !(mode & SNDRV_RAWMIDI_LFLG_APPEND)) + continue; + } + if (subdevice < 0 || subdevice == substream->number) { + *sub_ret = substream; + return 0; + } + } + return -EAGAIN; +} + +/* open and do ref-counting for the given substream */ +static int open_substream(struct snd_rawmidi *rmidi, + struct snd_rawmidi_substream *substream, + int mode) +{ + int err; + + err = snd_rawmidi_runtime_create(substream); + if (err < 0) + return err; + err = substream->ops->open(substream); + if (err < 0) + return err; + substream->opened = 1; + if (substream->use_count++ == 0) + substream->active_sensing = 1; + if (mode & SNDRV_RAWMIDI_LFLG_APPEND) + substream->append = 1; + rmidi->streams[substream->stream].substream_opened++; + return 0; +} + +static void close_substream(struct snd_rawmidi *rmidi, + struct snd_rawmidi_substream *substream, + int cleanup); + +static int rawmidi_open_priv(struct snd_rawmidi *rmidi, int subdevice, int mode, + struct snd_rawmidi_file *rfile) { - struct snd_rawmidi *rmidi; - struct list_head *list1, *list2; struct snd_rawmidi_substream *sinput = NULL, *soutput = NULL; - struct snd_rawmidi_runtime *input = NULL, *output = NULL; int err; - if (rfile) - rfile->input = rfile->output = NULL; - mutex_lock(®ister_mutex); - rmidi = snd_rawmidi_search(card, device); - mutex_unlock(®ister_mutex); - if (rmidi == NULL) { - err = -ENODEV; - goto __error1; - } - if (!try_module_get(rmidi->card->module)) { - err = -EFAULT; - goto __error1; - } - if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - mutex_lock(&rmidi->open_mutex); + rfile->input = rfile->output = NULL; if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { - if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_INPUT)) { - err = -ENXIO; - goto __error; - } - if (subdevice >= 0 && (unsigned int)subdevice >= rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_count) { - err = -ENODEV; - goto __error; - } - if (rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened >= - rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_count) { - err = -EAGAIN; + err = assign_substream(rmidi, subdevice, + SNDRV_RAWMIDI_STREAM_INPUT, + mode, &sinput); + if (err < 0) goto __error; - } } if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - if (!(rmidi->info_flags & SNDRV_RAWMIDI_INFO_OUTPUT)) { - err = -ENXIO; - goto __error; - } - if (subdevice >= 0 && (unsigned int)subdevice >= rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_count) { - err = -ENODEV; - goto __error; - } - if (rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened >= - rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_count) { - err = -EAGAIN; + err = assign_substream(rmidi, subdevice, + SNDRV_RAWMIDI_STREAM_OUTPUT, + mode, &soutput); + if (err < 0) goto __error; - } - } - list1 = rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams.next; - while (1) { - if (list1 == &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substreams) { - sinput = NULL; - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { - err = -EAGAIN; - goto __error; - } - break; - } - sinput = list_entry(list1, struct snd_rawmidi_substream, list); - if ((mode & SNDRV_RAWMIDI_LFLG_INPUT) && sinput->opened) - goto __nexti; - if (subdevice < 0 || (subdevice >= 0 && subdevice == sinput->number)) - break; - __nexti: - list1 = list1->next; } - list2 = rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams.next; - while (1) { - if (list2 == &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) { - soutput = NULL; - if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - err = -EAGAIN; - goto __error; - } - break; - } - soutput = list_entry(list2, struct snd_rawmidi_substream, list); - if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - if (mode & SNDRV_RAWMIDI_LFLG_APPEND) { - if (soutput->opened && !soutput->append) - goto __nexto; - } else { - if (soutput->opened) - goto __nexto; - } - } - if (subdevice < 0 || (subdevice >= 0 && subdevice == soutput->number)) - break; - __nexto: - list2 = list2->next; - } - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) { - if ((err = snd_rawmidi_runtime_create(sinput)) < 0) - goto __error; - input = sinput->runtime; - if ((err = sinput->ops->open(sinput)) < 0) + + if (sinput) { + err = open_substream(rmidi, sinput, mode); + if (err < 0) goto __error; - sinput->opened = 1; - rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened++; - } else { - sinput = NULL; } - if (mode & SNDRV_RAWMIDI_LFLG_OUTPUT) { - if (soutput->opened) - goto __skip_output; - if ((err = snd_rawmidi_runtime_create(soutput)) < 0) { - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) - sinput->ops->close(sinput); - goto __error; - } - output = soutput->runtime; - if ((err = soutput->ops->open(soutput)) < 0) { - if (mode & SNDRV_RAWMIDI_LFLG_INPUT) - sinput->ops->close(sinput); + if (soutput) { + err = open_substream(rmidi, soutput, mode); + if (err < 0) { + if (sinput) + close_substream(rmidi, sinput, 0); goto __error; } - __skip_output: - soutput->opened = 1; - if (mode & SNDRV_RAWMIDI_LFLG_APPEND) - soutput->append = 1; - if (soutput->use_count++ == 0) - soutput->active_sensing = 1; - rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened++; - } else { - soutput = NULL; - } - if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - mutex_unlock(&rmidi->open_mutex); - if (rfile) { - rfile->rmidi = rmidi; - rfile->input = sinput; - rfile->output = soutput; } + + rfile->rmidi = rmidi; + rfile->input = sinput; + rfile->output = soutput; return 0; __error: - if (input != NULL) + if (sinput && sinput->runtime) snd_rawmidi_runtime_free(sinput); - if (output != NULL) + if (soutput && soutput->runtime) snd_rawmidi_runtime_free(soutput); - module_put(rmidi->card->module); - if (!(mode & SNDRV_RAWMIDI_LFLG_NOOPENLOCK)) - mutex_unlock(&rmidi->open_mutex); - __error1: + return err; +} + +/* called from sound/core/seq/seq_midi.c */ +int snd_rawmidi_kernel_open(struct snd_card *card, int device, int subdevice, + int mode, struct snd_rawmidi_file * rfile) +{ + struct snd_rawmidi *rmidi; + int err; + + if (snd_BUG_ON(!rfile)) + return -EINVAL; + + mutex_lock(®ister_mutex); + rmidi = snd_rawmidi_search(card, device); + if (rmidi == NULL) { + mutex_unlock(®ister_mutex); + return -ENODEV; + } + if (!try_module_get(rmidi->card->module)) { + mutex_unlock(®ister_mutex); + return -ENXIO; + } + mutex_unlock(®ister_mutex); + + mutex_lock(&rmidi->open_mutex); + err = rawmidi_open_priv(rmidi, subdevice, mode, rfile); + mutex_unlock(&rmidi->open_mutex); + if (err < 0) + module_put(rmidi->card->module); return err; } @@ -385,10 +372,13 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) unsigned short fflags; int err; struct snd_rawmidi *rmidi; - struct snd_rawmidi_file *rawmidi_file; + struct snd_rawmidi_file *rawmidi_file = NULL; wait_queue_t wait; struct snd_ctl_file *kctl; + if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) + return -EINVAL; /* invalid combination */ + if (maj == snd_major) { rmidi = snd_lookup_minor_data(iminor(inode), SNDRV_DEVICE_TYPE_RAWMIDI); @@ -402,24 +392,25 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) if (rmidi == NULL) return -ENODEV; - if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) - return -EINVAL; /* invalid combination */ + + if (!try_module_get(rmidi->card->module)) + return -ENXIO; + + mutex_lock(&rmidi->open_mutex); card = rmidi->card; err = snd_card_file_add(card, file); if (err < 0) - return -ENODEV; + goto __error_card; fflags = snd_rawmidi_file_flags(file); if ((file->f_flags & O_APPEND) || maj == SOUND_MAJOR) /* OSS emul? */ fflags |= SNDRV_RAWMIDI_LFLG_APPEND; - fflags |= SNDRV_RAWMIDI_LFLG_NOOPENLOCK; rawmidi_file = kmalloc(sizeof(*rawmidi_file), GFP_KERNEL); if (rawmidi_file == NULL) { - snd_card_file_remove(card, file); - return -ENOMEM; + err = -ENOMEM; + goto __error; } init_waitqueue_entry(&wait, current); add_wait_queue(&rmidi->open_wait, &wait); - mutex_lock(&rmidi->open_mutex); while (1) { subdevice = -1; read_lock(&card->ctl_files_rwlock); @@ -431,8 +422,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) } } read_unlock(&card->ctl_files_rwlock); - err = snd_rawmidi_kernel_open(rmidi->card, rmidi->device, - subdevice, fflags, rawmidi_file); + err = rawmidi_open_priv(rmidi, subdevice, fflags, rawmidi_file); if (err >= 0) break; if (err == -EAGAIN) { @@ -451,67 +441,89 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) break; } } + remove_wait_queue(&rmidi->open_wait, &wait); + if (err < 0) { + kfree(rawmidi_file); + goto __error; + } #ifdef CONFIG_SND_OSSEMUL if (rawmidi_file->input && rawmidi_file->input->runtime) rawmidi_file->input->runtime->oss = (maj == SOUND_MAJOR); if (rawmidi_file->output && rawmidi_file->output->runtime) rawmidi_file->output->runtime->oss = (maj == SOUND_MAJOR); #endif - remove_wait_queue(&rmidi->open_wait, &wait); - if (err >= 0) { - file->private_data = rawmidi_file; - } else { - snd_card_file_remove(card, file); - kfree(rawmidi_file); - } + file->private_data = rawmidi_file; + mutex_unlock(&rmidi->open_mutex); + return 0; + + __error: + snd_card_file_remove(card, file); + __error_card: mutex_unlock(&rmidi->open_mutex); + module_put(rmidi->card->module); return err; } -int snd_rawmidi_kernel_release(struct snd_rawmidi_file * rfile) +static void close_substream(struct snd_rawmidi *rmidi, + struct snd_rawmidi_substream *substream, + int cleanup) { - struct snd_rawmidi *rmidi; - struct snd_rawmidi_substream *substream; - struct snd_rawmidi_runtime *runtime; + rmidi->streams[substream->stream].substream_opened--; + if (--substream->use_count) + return; - if (snd_BUG_ON(!rfile)) - return -ENXIO; - rmidi = rfile->rmidi; - mutex_lock(&rmidi->open_mutex); - if (rfile->input != NULL) { - substream = rfile->input; - rfile->input = NULL; - runtime = substream->runtime; - snd_rawmidi_input_trigger(substream, 0); - substream->ops->close(substream); - if (runtime->private_free != NULL) - runtime->private_free(substream); - snd_rawmidi_runtime_free(substream); - substream->opened = 0; - rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT].substream_opened--; - } - if (rfile->output != NULL) { - substream = rfile->output; - rfile->output = NULL; - if (--substream->use_count == 0) { - runtime = substream->runtime; + if (cleanup) { + if (substream->stream == SNDRV_RAWMIDI_STREAM_INPUT) + snd_rawmidi_input_trigger(substream, 0); + else { if (substream->active_sensing) { unsigned char buf = 0xfe; - /* sending single active sensing message to shut the device up */ + /* sending single active sensing message + * to shut the device up + */ snd_rawmidi_kernel_write(substream, &buf, 1); } if (snd_rawmidi_drain_output(substream) == -ERESTARTSYS) snd_rawmidi_output_trigger(substream, 0); - substream->ops->close(substream); - if (runtime->private_free != NULL) - runtime->private_free(substream); - snd_rawmidi_runtime_free(substream); - substream->opened = 0; - substream->append = 0; } - rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substream_opened--; } + substream->ops->close(substream); + if (substream->runtime->private_free) + substream->runtime->private_free(substream); + snd_rawmidi_runtime_free(substream); + substream->opened = 0; + substream->append = 0; +} + +static void rawmidi_release_priv(struct snd_rawmidi_file *rfile) +{ + struct snd_rawmidi *rmidi; + + rmidi = rfile->rmidi; + mutex_lock(&rmidi->open_mutex); + if (rfile->input) { + close_substream(rmidi, rfile->input, 1); + rfile->input = NULL; + } + if (rfile->output) { + close_substream(rmidi, rfile->output, 1); + rfile->output = NULL; + } + rfile->rmidi = NULL; mutex_unlock(&rmidi->open_mutex); + wake_up(&rmidi->open_wait); +} + +/* called from sound/core/seq/seq_midi.c */ +int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile) +{ + struct snd_rawmidi *rmidi; + + if (snd_BUG_ON(!rfile)) + return -ENXIO; + + rmidi = rfile->rmidi; + rawmidi_release_priv(rfile); module_put(rmidi->card->module); return 0; } @@ -520,15 +532,14 @@ static int snd_rawmidi_release(struct inode *inode, struct file *file) { struct snd_rawmidi_file *rfile; struct snd_rawmidi *rmidi; - int err; rfile = file->private_data; - err = snd_rawmidi_kernel_release(rfile); rmidi = rfile->rmidi; - wake_up(&rmidi->open_wait); + rawmidi_release_priv(rfile); kfree(rfile); snd_card_file_remove(rmidi->card, file); - return err; + module_put(rmidi->card->module); + return 0; } static int snd_rawmidi_info(struct snd_rawmidi_substream *substream, diff --git a/sound/core/seq/oss/seq_oss_device.h b/sound/core/seq/oss/seq_oss_device.h index bf8d2b4..c0154a9 100644 --- a/sound/core/seq/oss/seq_oss_device.h +++ b/sound/core/seq/oss/seq_oss_device.h @@ -181,7 +181,7 @@ char *enabled_str(int bool); /* for debug */ #ifdef SNDRV_SEQ_OSS_DEBUG extern int seq_oss_debug; -#define debug_printk(x) do { if (seq_oss_debug > 0) snd_printk x; } while (0) +#define debug_printk(x) do { if (seq_oss_debug > 0) snd_printd x; } while (0) #else #define debug_printk(x) /**/ #endif diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c index 0101a8b..29896ab 100644 --- a/sound/core/seq/seq_prioq.c +++ b/sound/core/seq/seq_prioq.c @@ -321,7 +321,8 @@ void snd_seq_prioq_leave(struct snd_seq_prioq * f, int client, int timestamp) freeprev = cell; } else { #if 0 - printk("type = %i, source = %i, dest = %i, client = %i\n", + printk(KERN_DEBUG "type = %i, source = %i, dest = %i, " + "client = %i\n", cell->event.type, cell->event.source.client, cell->event.dest.client, diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index d4564ed..4e7ec2b 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -38,6 +38,10 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) if (! sgbuf) return -EINVAL; + if (dmab->area) + vunmap(dmab->area); + dmab->area = NULL; + tmpb.dev.type = SNDRV_DMA_TYPE_DEV; tmpb.dev.dev = sgbuf->dev; for (i = 0; i < sgbuf->pages; i++) { @@ -48,9 +52,6 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) tmpb.bytes = (sgbuf->table[i].addr & ~PAGE_MASK) << PAGE_SHIFT; snd_dma_free_pages(&tmpb); } - if (dmab->area) - vunmap(dmab->area); - dmab->area = NULL; kfree(sgbuf->table); kfree(sgbuf->page_table); diff --git a/sound/core/timer.c b/sound/core/timer.c index 7965320..3f0050d 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -1825,13 +1825,9 @@ static long snd_timer_user_ioctl(struct file *file, unsigned int cmd, static int snd_timer_user_fasync(int fd, struct file * file, int on) { struct snd_timer_user *tu; - int err; tu = file->private_data; - err = fasync_helper(fd, file, on, &tu->fasync); - if (err < 0) - return err; - return 0; + return fasync_helper(fd, file, on, &tu->fasync); } static ssize_t snd_timer_user_read(struct file *file, char __user *buffer, diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 4cc57f9..257624b 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -50,18 +50,38 @@ struct link_slave { struct link_master *master; struct link_ctl_info info; int vals[2]; /* current values */ + unsigned int flags; struct snd_kcontrol slave; /* the copy of original control entry */ }; +static int slave_update(struct link_slave *slave) +{ + struct snd_ctl_elem_value *uctl; + int err, ch; + + uctl = kmalloc(sizeof(*uctl), GFP_KERNEL); + if (!uctl) + return -ENOMEM; + uctl->id = slave->slave.id; + err = slave->slave.get(&slave->slave, uctl); + for (ch = 0; ch < slave->info.count; ch++) + slave->vals[ch] = uctl->value.integer.value[ch]; + kfree(uctl); + return 0; +} + /* get the slave ctl info and save the initial values */ static int slave_init(struct link_slave *slave) { struct snd_ctl_elem_info *uinfo; - struct snd_ctl_elem_value *uctl; - int err, ch; + int err; - if (slave->info.count) - return 0; /* already initialized */ + if (slave->info.count) { + /* already initialized */ + if (slave->flags & SND_CTL_SLAVE_NEED_UPDATE) + return slave_update(slave); + return 0; + } uinfo = kmalloc(sizeof(*uinfo), GFP_KERNEL); if (!uinfo) @@ -85,15 +105,7 @@ static int slave_init(struct link_slave *slave) slave->info.max_val = uinfo->value.integer.max; kfree(uinfo); - uctl = kmalloc(sizeof(*uctl), GFP_KERNEL); - if (!uctl) - return -ENOMEM; - uctl->id = slave->slave.id; - err = slave->slave.get(&slave->slave, uctl); - for (ch = 0; ch < slave->info.count; ch++) - slave->vals[ch] = uctl->value.integer.value[ch]; - kfree(uctl); - return 0; + return slave_update(slave); } /* initialize master volume */ @@ -229,7 +241,8 @@ static void slave_free(struct snd_kcontrol *kcontrol) * - logarithmic volume control (dB level), no linear volume * - master can only attenuate the volume, no gain */ -int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) +int _snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave, + unsigned int flags) { struct link_master *master_link = snd_kcontrol_chip(master); struct link_slave *srec; @@ -241,6 +254,7 @@ int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) srec->slave = *slave; memcpy(srec->slave.vd, slave->vd, slave->count * sizeof(*slave->vd)); srec->master = master_link; + srec->flags = flags; /* override callbacks */ slave->info = slave_info; @@ -254,8 +268,7 @@ int snd_ctl_add_slave(struct snd_kcontrol *master, struct snd_kcontrol *slave) list_add_tail(&srec->list, &master_link->slaves); return 0; } - -EXPORT_SYMBOL(snd_ctl_add_slave); +EXPORT_SYMBOL(_snd_ctl_add_slave); /* * ctl callbacks for master controls @@ -327,8 +340,20 @@ static void master_free(struct snd_kcontrol *kcontrol) } -/* - * Create a virtual master control with the given name +/** + * snd_ctl_make_virtual_master - Create a virtual master control + * @name: name string of the control element to create + * @tlv: optional TLV int array for dB information + * + * Creates a virtual matster control with the given name string. + * Returns the created control element, or NULL for errors (ENOMEM). + * + * After creating a vmaster element, you can add the slave controls + * via snd_ctl_add_slave() or snd_ctl_add_slave_uncached(). + * + * The optional argument @tlv can be used to specify the TLV information + * for dB scale of the master control. It should be a single element + * with #SNDRV_CTL_TLVT_DB_SCALE type, and should be the max 0dB. */ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, const unsigned int *tlv) @@ -367,5 +392,4 @@ struct snd_kcontrol *snd_ctl_make_virtual_master(char *name, return kctl; } - EXPORT_SYMBOL(snd_ctl_make_virtual_master); diff --git a/sound/drivers/Kconfig b/sound/drivers/Kconfig index 0bcf146..84714a6 100644 --- a/sound/drivers/Kconfig +++ b/sound/drivers/Kconfig @@ -33,7 +33,7 @@ if SND_DRIVERS config SND_PCSP tristate "PC-Speaker support (READ HELP!)" - depends on PCSPKR_PLATFORM && X86_PC && HIGH_RES_TIMERS + depends on PCSPKR_PLATFORM && X86 && HIGH_RES_TIMERS depends on INPUT depends on EXPERIMENTAL select SND_PCM diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index c3e9833..2f8f295 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -303,8 +303,10 @@ static void snd_mtpav_output_port_write(struct mtpav *mtp_card, snd_mtpav_send_byte(mtp_card, 0xf5); snd_mtpav_send_byte(mtp_card, portp->hwport); - //snd_printk("new outport: 0x%x\n", (unsigned int) portp->hwport); - + /* + snd_printk(KERN_DEBUG "new outport: 0x%x\n", + (unsigned int) portp->hwport); + */ if (!(outbyte & 0x80) && portp->running_status) snd_mtpav_send_byte(mtp_card, portp->running_status); } @@ -540,7 +542,7 @@ static void snd_mtpav_read_bytes(struct mtpav *mcrd) u8 sbyt = snd_mtpav_getreg(mcrd, SREG); - //printk("snd_mtpav_read_bytes() sbyt: 0x%x\n", sbyt); + /* printk(KERN_DEBUG "snd_mtpav_read_bytes() sbyt: 0x%x\n", sbyt); */ if (!(sbyt & SIGS_BYTE)) return; @@ -585,12 +587,12 @@ static irqreturn_t snd_mtpav_irqh(int irq, void *dev_id) static int __devinit snd_mtpav_get_ISA(struct mtpav * mcard) { if ((mcard->res_port = request_region(port, 3, "MotuMTPAV MIDI")) == NULL) { - snd_printk("MTVAP port 0x%lx is busy\n", port); + snd_printk(KERN_ERR "MTVAP port 0x%lx is busy\n", port); return -EBUSY; } mcard->port = port; if (request_irq(irq, snd_mtpav_irqh, IRQF_DISABLED, "MOTU MTPAV", mcard)) { - snd_printk("MTVAP IRQ %d busy\n", irq); + snd_printk(KERN_ERR "MTVAP IRQ %d busy\n", irq); return -EBUSY; } mcard->irq = irq; @@ -706,7 +708,6 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev) mtp_card->card = card; mtp_card->irq = -1; mtp_card->share_irq = 0; - mtp_card->inmidiport = 0xffffffff; mtp_card->inmidistate = 0; mtp_card->outmidihwport = 0xffffffff; init_timer(&mtp_card->timer); @@ -719,6 +720,8 @@ static int __devinit snd_mtpav_probe(struct platform_device *dev) if (err < 0) goto __error; + mtp_card->inmidiport = mtp_card->num_ports + MTPAV_PIDX_BROADCAST; + err = snd_mtpav_get_ISA(mtp_card); if (err < 0) goto __error; diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 33d9db7..9284829 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -1015,7 +1015,7 @@ static int __devinit snd_mts64_probe(struct platform_device *pdev) goto __err; } - snd_printk("ESI Miditerminal 4140 on 0x%lx\n", p->base); + snd_printk(KERN_INFO "ESI Miditerminal 4140 on 0x%lx\n", p->base); return 0; __err: diff --git a/sound/drivers/opl3/opl3_lib.c b/sound/drivers/opl3/opl3_lib.c index 7805823..6e31e46 100644 --- a/sound/drivers/opl3/opl3_lib.c +++ b/sound/drivers/opl3/opl3_lib.c @@ -302,7 +302,7 @@ void snd_opl3_interrupt(struct snd_hwdep * hw) opl3 = hw->private_data; status = inb(opl3->l_port); #if 0 - snd_printk("AdLib IRQ status = 0x%x\n", status); + snd_printk(KERN_DEBUG "AdLib IRQ status = 0x%x\n", status); #endif if (!(status & 0x80)) return; diff --git a/sound/drivers/opl3/opl3_midi.c b/sound/drivers/opl3/opl3_midi.c index 16feafa..6e7d09a 100644 --- a/sound/drivers/opl3/opl3_midi.c +++ b/sound/drivers/opl3/opl3_midi.c @@ -125,7 +125,7 @@ static void debug_alloc(struct snd_opl3 *opl3, char *s, int voice) { int i; char *str = "x.24"; - printk("time %.5i: %s [%.2i]: ", opl3->use_time, s, voice); + printk(KERN_DEBUG "time %.5i: %s [%.2i]: ", opl3->use_time, s, voice); for (i = 0; i < opl3->max_voices; i++) printk("%c", *(str + opl3->voices[i].state + 1)); printk("\n"); @@ -218,7 +218,7 @@ static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op, for (i = 0; i < END; i++) { if (best[i].voice >= 0) { #ifdef DEBUG_ALLOC - printk("%s %iop allocation on voice %i\n", + printk(KERN_DEBUG "%s %iop allocation on voice %i\n", alloc_type[i], instr_4op ? 4 : 2, best[i].voice); #endif @@ -317,7 +317,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Note on, ch %i, inst %i, note %i, vel %i\n", + snd_printk(KERN_DEBUG "Note on, ch %i, inst %i, note %i, vel %i\n", chan->number, chan->midi_program, note, vel); #endif @@ -372,7 +372,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) return; } #ifdef DEBUG_MIDI - snd_printk(" --> OPL%i instrument: %s\n", + snd_printk(KERN_DEBUG " --> OPL%i instrument: %s\n", instr_4op ? 3 : 2, patch->name); #endif /* in SYNTH mode, application takes care of voices */ @@ -431,7 +431,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) } #ifdef DEBUG_MIDI - snd_printk(" --> setting OPL3 connection: 0x%x\n", + snd_printk(KERN_DEBUG " --> setting OPL3 connection: 0x%x\n", opl3->connection_reg); #endif /* @@ -466,7 +466,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) /* Program the FM voice characteristics */ for (i = 0; i < (instr_4op ? 4 : 2); i++) { #ifdef DEBUG_MIDI - snd_printk(" --> programming operator %i\n", i); + snd_printk(KERN_DEBUG " --> programming operator %i\n", i); #endif op_offset = snd_opl3_regmap[voice_offset][i]; @@ -546,7 +546,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) blocknum |= OPL3_KEYON_BIT; #ifdef DEBUG_MIDI - snd_printk(" --> trigger voice %i\n", voice); + snd_printk(KERN_DEBUG " --> trigger voice %i\n", voice); #endif /* Set OPL3 KEYON_BLOCK register of requested voice */ opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset); @@ -602,7 +602,7 @@ void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) prg = extra_prg - 1; } #ifdef DEBUG_MIDI - snd_printk(" *** allocating extra program\n"); + snd_printk(KERN_DEBUG " *** allocating extra program\n"); #endif goto __extra_prg; } @@ -633,7 +633,7 @@ static void snd_opl3_kill_voice(struct snd_opl3 *opl3, int voice) /* kill voice */ #ifdef DEBUG_MIDI - snd_printk(" --> kill voice %i\n", voice); + snd_printk(KERN_DEBUG " --> kill voice %i\n", voice); #endif opl3_reg = reg_side | (OPL3_REG_KEYON_BLOCK + voice_offset); /* clear Key ON bit */ @@ -670,7 +670,7 @@ void snd_opl3_note_off(void *p, int note, int vel, struct snd_midi_channel *chan opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Note off, ch %i, inst %i, note %i\n", + snd_printk(KERN_DEBUG "Note off, ch %i, inst %i, note %i\n", chan->number, chan->midi_program, note); #endif @@ -709,7 +709,7 @@ void snd_opl3_key_press(void *p, int note, int vel, struct snd_midi_channel *cha opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Key pressure, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "Key pressure, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); #endif } @@ -723,7 +723,7 @@ void snd_opl3_terminate_note(void *p, int note, struct snd_midi_channel *chan) opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Terminate note, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "Terminate note, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); #endif } @@ -812,7 +812,7 @@ void snd_opl3_control(void *p, int type, struct snd_midi_channel *chan) opl3 = p; #ifdef DEBUG_MIDI - snd_printk("Controller, TYPE = %i, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "Controller, TYPE = %i, ch#: %i, inst#: %i\n", type, chan->number, chan->midi_program); #endif @@ -849,7 +849,7 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, opl3 = p; #ifdef DEBUG_MIDI - snd_printk("NRPN, ch#: %i, inst#: %i\n", + snd_printk(KERN_DEBUG "NRPN, ch#: %i, inst#: %i\n", chan->number, chan->midi_program); #endif } @@ -864,6 +864,6 @@ void snd_opl3_sysex(void *p, unsigned char *buf, int len, opl3 = p; #ifdef DEBUG_MIDI - snd_printk("SYSEX\n"); + snd_printk(KERN_DEBUG "SYSEX\n"); #endif } diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index 9a2271d..a54b1dc 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -220,14 +220,14 @@ static int snd_opl3_load_patch_seq_oss(struct snd_seq_oss_arg *arg, int format, return -EINVAL; if (count < (int)sizeof(sbi)) { - snd_printk("FM Error: Patch record too short\n"); + snd_printk(KERN_ERR "FM Error: Patch record too short\n"); return -EINVAL; } if (copy_from_user(&sbi, buf, sizeof(sbi))) return -EFAULT; if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) { - snd_printk("FM Error: Invalid instrument number %d\n", + snd_printk(KERN_ERR "FM Error: Invalid instrument number %d\n", sbi.channel); return -EINVAL; } @@ -254,7 +254,9 @@ static int snd_opl3_ioctl_seq_oss(struct snd_seq_oss_arg *arg, unsigned int cmd, opl3 = arg->private_data; switch (cmd) { case SNDCTL_FM_LOAD_INSTR: - snd_printk("OPL3: Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. Fix the program.\n"); + snd_printk(KERN_ERR "OPL3: " + "Obsolete ioctl(SNDCTL_FM_LOAD_INSTR) used. " + "Fix the program.\n"); return -EINVAL; case SNDCTL_SYNTH_MEMAVL: diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 962bb9c..6d57b64 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -168,7 +168,7 @@ int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, #ifdef CONFIG_SND_DEBUG default: - snd_printk("unknown IOCTL: 0x%x\n", cmd); + snd_printk(KERN_WARNING "unknown IOCTL: 0x%x\n", cmd); #endif } return -ENOTTY; diff --git a/sound/drivers/pcsp/pcsp.c b/sound/drivers/pcsp/pcsp.c index aa2ae07..b60cef2 100644 --- a/sound/drivers/pcsp/pcsp.c +++ b/sound/drivers/pcsp/pcsp.c @@ -57,7 +57,7 @@ static int __devinit snd_pcsp_create(struct snd_card *card) else min_div = MAX_DIV; #if PCSP_DEBUG - printk("PCSP: lpj=%li, min_div=%i, res=%li\n", + printk(KERN_DEBUG "PCSP: lpj=%li, min_div=%i, res=%li\n", loops_per_jiffy, min_div, tp.tv_nsec); #endif diff --git a/sound/drivers/serial-u16550.c b/sound/drivers/serial-u16550.c index 891d081..b2b6d50 100644 --- a/sound/drivers/serial-u16550.c +++ b/sound/drivers/serial-u16550.c @@ -241,7 +241,8 @@ static void snd_uart16550_io_loop(struct snd_uart16550 * uart) snd_rawmidi_receive(uart->midi_input[substream], &c, 1); if (status & UART_LSR_OE) - snd_printk("%s: Overrun on device at 0x%lx\n", + snd_printk(KERN_WARNING + "%s: Overrun on device at 0x%lx\n", uart->rmidi->name, uart->base); } @@ -636,7 +637,8 @@ static int snd_uart16550_output_byte(struct snd_uart16550 *uart, } } else { if (!snd_uart16550_write_buffer(uart, midi_byte)) { - snd_printk("%s: Buffer overrun on device at 0x%lx\n", + snd_printk(KERN_WARNING + "%s: Buffer overrun on device at 0x%lx\n", uart->rmidi->name, uart->base); return 0; } @@ -815,7 +817,8 @@ static int __devinit snd_uart16550_create(struct snd_card *card, if (irq >= 0 && irq != SNDRV_AUTO_IRQ) { if (request_irq(irq, snd_uart16550_interrupt, IRQF_DISABLED, "Serial MIDI", uart)) { - snd_printk("irq %d busy. Using Polling.\n", irq); + snd_printk(KERN_WARNING + "irq %d busy. Using Polling.\n", irq); } else { uart->irq = irq; } @@ -919,19 +922,22 @@ static int __devinit snd_serial_probe(struct platform_device *devptr) case SNDRV_SERIAL_GENERIC: break; default: - snd_printk("Adaptor type is out of range 0-%d (%d)\n", + snd_printk(KERN_ERR + "Adaptor type is out of range 0-%d (%d)\n", SNDRV_SERIAL_MAX_ADAPTOR, adaptor[dev]); return -ENODEV; } if (outs[dev] < 1 || outs[dev] > SNDRV_SERIAL_MAX_OUTS) { - snd_printk("Count of outputs is out of range 1-%d (%d)\n", + snd_printk(KERN_ERR + "Count of outputs is out of range 1-%d (%d)\n", SNDRV_SERIAL_MAX_OUTS, outs[dev]); return -ENODEV; } if (ins[dev] < 1 || ins[dev] > SNDRV_SERIAL_MAX_INS) { - snd_printk("Count of inputs is out of range 1-%d (%d)\n", + snd_printk(KERN_ERR + "Count of inputs is out of range 1-%d (%d)\n", SNDRV_SERIAL_MAX_INS, ins[dev]); return -ENODEV; } diff --git a/sound/drivers/virmidi.c b/sound/drivers/virmidi.c index 6f48711..0e631c3 100644 --- a/sound/drivers/virmidi.c +++ b/sound/drivers/virmidi.c @@ -98,7 +98,9 @@ static int __devinit snd_virmidi_probe(struct platform_device *devptr) vmidi->card = card; if (midi_devs[dev] > MAX_MIDI_DEVICES) { - snd_printk("too much midi devices for virmidi %d: force to use %d\n", dev, MAX_MIDI_DEVICES); + snd_printk(KERN_WARNING + "too much midi devices for virmidi %d: " + "force to use %d\n", dev, MAX_MIDI_DEVICES); midi_devs[dev] = MAX_MIDI_DEVICES; } for (idx = 0; idx < midi_devs[dev]; idx++) { diff --git a/sound/drivers/vx/vx_core.c b/sound/drivers/vx/vx_core.c index 14e3354..19c6e37 100644 --- a/sound/drivers/vx/vx_core.c +++ b/sound/drivers/vx/vx_core.c @@ -688,7 +688,8 @@ int snd_vx_dsp_load(struct vx_core *chip, const struct firmware *dsp) image = dsp->data + i; /* Wait DSP ready for a new read */ if ((err = vx_wait_isr_bit(chip, ISR_TX_EMPTY)) < 0) { - printk("dsp loading error at position %d\n", i); + printk(KERN_ERR + "dsp loading error at position %d\n", i); return err; } cptr = image; diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index 8d6362e..46df881 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -119,16 +119,6 @@ void snd_vx_free_firmware(struct vx_core *chip) #else /* old style firmware loading */ -static int vx_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int vx_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - static int vx_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { @@ -243,8 +233,6 @@ int snd_vx_setup_firmware(struct vx_core *chip) hw->iface = SNDRV_HWDEP_IFACE_VX; hw->private_data = chip; - hw->ops.open = vx_hwdep_open; - hw->ops.release = vx_hwdep_release; hw->ops.dsp_status = vx_hwdep_dsp_status; hw->ops.dsp_load = vx_hwdep_dsp_load; hw->exclusive = 1; diff --git a/sound/drivers/vx/vx_uer.c b/sound/drivers/vx/vx_uer.c index 0e1ba9b..b0560fec 100644 --- a/sound/drivers/vx/vx_uer.c +++ b/sound/drivers/vx/vx_uer.c @@ -103,7 +103,7 @@ static void vx_write_one_cbit(struct vx_core *chip, int index, int val) * returns the frequency of UER, or 0 if not sync, * or a negative error code. */ -static int vx_read_uer_status(struct vx_core *chip, int *mode) +static int vx_read_uer_status(struct vx_core *chip, unsigned int *mode) { int val, freq; diff --git a/sound/i2c/Makefile b/sound/i2c/Makefile index 37970666..36879bf 100644 --- a/sound/i2c/Makefile +++ b/sound/i2c/Makefile @@ -7,8 +7,6 @@ snd-i2c-objs := i2c.o snd-cs8427-objs := cs8427.o snd-tea6330t-objs := tea6330t.o -obj-$(CONFIG_L3) += l3/ - obj-$(CONFIG_SND) += other/ # Toplevel Module Dependency diff --git a/sound/i2c/l3/Makefile b/sound/i2c/l3/Makefile deleted file mode 100644 index 49455b8..0000000 --- a/sound/i2c/l3/Makefile +++ /dev/null @@ -1,8 +0,0 @@ -# -# Makefile for ALSA -# - -snd-uda1341-objs := uda1341.o - -# Module Dependency -obj-$(CONFIG_SND_SA11XX_UDA1341) += snd-uda1341.o diff --git a/sound/i2c/l3/uda1341.c b/sound/i2c/l3/uda1341.c deleted file mode 100644 index 9840eb4..0000000 --- a/sound/i2c/l3/uda1341.c +++ /dev/null @@ -1,935 +0,0 @@ -/* - * Philips UDA1341 mixer device driver - * Copyright (c) 2002 Tomas Kasparek <tomas.kasparek@seznam.cz> - * - * Portions are Copyright (C) 2000 Lernout & Hauspie Speech Products, N.V. - * - * This program is free software; you can redistribute it and/or - * modify it under the terms of the GNU General Public License. - * - * History: - * - * 2002-03-13 Tomas Kasparek initial release - based on uda1341.c from OSS - * 2002-03-28 Tomas Kasparek basic mixer is working (volume, bass, treble) - * 2002-03-30 Tomas Kasparek proc filesystem support, complete mixer and DSP - * features support - * 2002-04-12 Tomas Kasparek proc interface update, code cleanup - * 2002-05-12 Tomas Kasparek another code cleanup - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/types.h> -#include <linux/slab.h> -#include <linux/errno.h> -#include <linux/ioctl.h> - -#include <asm/uaccess.h> - -#include <sound/core.h> -#include <sound/control.h> -#include <sound/initval.h> -#include <sound/info.h> - -#include <linux/l3/l3.h> - -#include <sound/uda1341.h> - -/* {{{ HW regs definition */ - -#define STAT0 0x00 -#define STAT1 0x80 -#define STAT_MASK 0x80 - -#define DATA0_0 0x00 -#define DATA0_1 0x40 -#define DATA0_2 0x80 -#define DATA_MASK 0xc0 - -#define IS_DATA0(x) ((x) >= data0_0 && (x) <= data0_2) -#define IS_DATA1(x) ((x) == data1) -#define IS_STATUS(x) ((x) == stat0 || (x) == stat1) -#define IS_EXTEND(x) ((x) >= ext0 && (x) <= ext6) - -/* }}} */ - - -static const char *peak_names[] = { - "before", - "after", -}; - -static const char *filter_names[] = { - "flat", - "min", - "min", - "max", -}; - -static const char *mixer_names[] = { - "double differential", - "input channel 1 (line in)", - "input channel 2 (microphone)", - "digital mixer", -}; - -static const char *deemp_names[] = { - "none", - "32 kHz", - "44.1 kHz", - "48 kHz", -}; - -enum uda1341_regs_names { - stat0, - stat1, - data0_0, - data0_1, - data0_2, - data1, - ext0, - ext1, - ext2, - empty, - ext4, - ext5, - ext6, - uda1341_reg_last, -}; - -static const char *uda1341_reg_names[] = { - "stat 0 ", - "stat 1 ", - "data 00", - "data 01", - "data 02", - "data 1 ", - "ext 0", - "ext 1", - "ext 2", - "empty", - "ext 4", - "ext 5", - "ext 6", -}; - -static const int uda1341_enum_items[] = { - 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, - 2, //peak - before/after - 4, //deemp - none/32/44.1/48 - 0, - 4, //filter - flat/min/min/max - 0, 0, 0, - 4, //mixer - differ/line/mic/mixer - 0, 0, 0, 0, 0, -}; - -static const char ** uda1341_enum_names[] = { - NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, - peak_names, //peak - before/after - deemp_names, //deemp - none/32/44.1/48 - NULL, - filter_names, //filter - flat/min/min/max - NULL, NULL, NULL, - mixer_names, //mixer - differ/line/mic/mixer - NULL, NULL, NULL, NULL, NULL, -}; - -typedef int uda1341_cfg[CMD_LAST]; - -struct uda1341 { - int (*write) (struct l3_client *uda1341, unsigned short reg, unsigned short val); - int (*read) (struct l3_client *uda1341, unsigned short reg); - unsigned char regs[uda1341_reg_last]; - int active; - spinlock_t reg_lock; - struct snd_card *card; - uda1341_cfg cfg; -#ifdef CONFIG_PM - unsigned char suspend_regs[uda1341_reg_last]; - uda1341_cfg suspend_cfg; -#endif -}; - -/* transfer 8bit integer into string with binary representation */ -static void int2str_bin8(uint8_t val, char *buf) -{ - const int size = sizeof(val) * 8; - int i; - - for (i= 0; i < size; i++){ - *(buf++) = (val >> (size - 1)) ? '1' : '0'; - val <<= 1; - } - *buf = '\0'; //end the string with zero -} - -/* {{{ HW manipulation routines */ - -static int snd_uda1341_codec_write(struct l3_client *clnt, unsigned short reg, unsigned short val) -{ - struct uda1341 *uda = clnt->driver_data; - unsigned char buf[2] = { 0xc0, 0xe0 }; // for EXT addressing - int err = 0; - - uda->regs[reg] = val; - - if (uda->active) { - if (IS_DATA0(reg)) { - err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)&val, 1); - } else if (IS_DATA1(reg)) { - err = l3_write(clnt, UDA1341_DATA1, (const unsigned char *)&val, 1); - } else if (IS_STATUS(reg)) { - err = l3_write(clnt, UDA1341_STATUS, (const unsigned char *)&val, 1); - } else if (IS_EXTEND(reg)) { - buf[0] |= (reg - ext0) & 0x7; //EXT address - buf[1] |= val; //EXT data - err = l3_write(clnt, UDA1341_DATA0, (const unsigned char *)buf, 2); - } - } else - printk(KERN_ERR "UDA1341 codec not active!\n"); - return err; -} - -static int snd_uda1341_codec_read(struct l3_client *clnt, unsigned short reg) -{ - unsigned char val; - int err; - - err = l3_read(clnt, reg, &val, 1); - if (err == 1) - // use just 6bits - the rest is address of the reg - return val & 63; - return err < 0 ? err : -EIO; -} - -static inline int snd_uda1341_valid_reg(struct l3_client *clnt, unsigned short reg) -{ - return reg < uda1341_reg_last; -} - -static int snd_uda1341_update_bits(struct l3_client *clnt, unsigned short reg, - unsigned short mask, unsigned short shift, - unsigned short value, int flush) -{ - int change; - unsigned short old, new; - struct uda1341 *uda = clnt->driver_data; - -#if 0 - printk(KERN_DEBUG "update_bits: reg: %s mask: %d shift: %d val: %d\n", - uda1341_reg_names[reg], mask, shift, value); -#endif - - if (!snd_uda1341_valid_reg(clnt, reg)) - return -EINVAL; - spin_lock(&uda->reg_lock); - old = uda->regs[reg]; - new = (old & ~(mask << shift)) | (value << shift); - change = old != new; - if (change) { - if (flush) uda->write(clnt, reg, new); - uda->regs[reg] = new; - } - spin_unlock(&uda->reg_lock); - return change; -} - -static int snd_uda1341_cfg_write(struct l3_client *clnt, unsigned short what, - unsigned short value, int flush) -{ - struct uda1341 *uda = clnt->driver_data; - int ret = 0; -#ifdef CONFIG_PM - int reg; -#endif - -#if 0 - printk(KERN_DEBUG "cfg_write what: %d value: %d\n", what, value); -#endif - - uda->cfg[what] = value; - - switch(what) { - case CMD_RESET: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, 1, flush); // MUTE - ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 1, flush); // RESET - ret = snd_uda1341_update_bits(clnt, stat0, 1, 6, 0, flush); // RESTORE - uda->cfg[CMD_RESET]=0; - break; - case CMD_FS: - ret = snd_uda1341_update_bits(clnt, stat0, 3, 4, value, flush); - break; - case CMD_FORMAT: - ret = snd_uda1341_update_bits(clnt, stat0, 7, 1, value, flush); - break; - case CMD_OGAIN: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 6, value, flush); - break; - case CMD_IGAIN: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 5, value, flush); - break; - case CMD_DAC: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 0, value, flush); - break; - case CMD_ADC: - ret = snd_uda1341_update_bits(clnt, stat1, 1, 1, value, flush); - break; - case CMD_VOLUME: - ret = snd_uda1341_update_bits(clnt, data0_0, 63, 0, value, flush); - break; - case CMD_BASS: - ret = snd_uda1341_update_bits(clnt, data0_1, 15, 2, value, flush); - break; - case CMD_TREBBLE: - ret = snd_uda1341_update_bits(clnt, data0_1, 3, 0, value, flush); - break; - case CMD_PEAK: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 5, value, flush); - break; - case CMD_DEEMP: - ret = snd_uda1341_update_bits(clnt, data0_2, 3, 3, value, flush); - break; - case CMD_MUTE: - ret = snd_uda1341_update_bits(clnt, data0_2, 1, 2, value, flush); - break; - case CMD_FILTER: - ret = snd_uda1341_update_bits(clnt, data0_2, 3, 0, value, flush); - break; - case CMD_CH1: - ret = snd_uda1341_update_bits(clnt, ext0, 31, 0, value, flush); - break; - case CMD_CH2: - ret = snd_uda1341_update_bits(clnt, ext1, 31, 0, value, flush); - break; - case CMD_MIC: - ret = snd_uda1341_update_bits(clnt, ext2, 7, 2, value, flush); - break; - case CMD_MIXER: - ret = snd_uda1341_update_bits(clnt, ext2, 3, 0, value, flush); - break; - case CMD_AGC: - ret = snd_uda1341_update_bits(clnt, ext4, 1, 4, value, flush); - break; - case CMD_IG: - ret = snd_uda1341_update_bits(clnt, ext4, 3, 0, value & 0x3, flush); - ret = snd_uda1341_update_bits(clnt, ext5, 31, 0, value >> 2, flush); - break; - case CMD_AGC_TIME: - ret = snd_uda1341_update_bits(clnt, ext6, 7, 2, value, flush); - break; - case CMD_AGC_LEVEL: - ret = snd_uda1341_update_bits(clnt, ext6, 3, 0, value, flush); - break; -#ifdef CONFIG_PM - case CMD_SUSPEND: - for (reg = stat0; reg < uda1341_reg_last; reg++) - uda->suspend_regs[reg] = uda->regs[reg]; - for (reg = 0; reg < CMD_LAST; reg++) - uda->suspend_cfg[reg] = uda->cfg[reg]; - break; - case CMD_RESUME: - for (reg = stat0; reg < uda1341_reg_last; reg++) - snd_uda1341_codec_write(clnt, reg, uda->suspend_regs[reg]); - for (reg = 0; reg < CMD_LAST; reg++) - uda->cfg[reg] = uda->suspend_cfg[reg]; - break; -#endif - default: - ret = -EINVAL; - break; - } - - if (!uda->active) - printk(KERN_ERR "UDA1341 codec not active!\n"); - return ret; -} - -/* }}} */ - -/* {{{ Proc interface */ -#ifdef CONFIG_PROC_FS - -static const char *format_names[] = { - "I2S-bus", - "LSB 16bits", - "LSB 18bits", - "LSB 20bits", - "MSB", - "in LSB 16bits/out MSB", - "in LSB 18bits/out MSB", - "in LSB 20bits/out MSB", -}; - -static const char *fs_names[] = { - "512*fs", - "384*fs", - "256*fs", - "Unused - bad value!", -}; - -static const char* bass_values[][16] = { - {"0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", "0 dB", - "0 dB", "0 dB", "0 dB", "0 dB", "undefined", }, //flat - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB", - "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "18 dB", - "18 dB", "18 dB", "18 dB", "18 dB", "undefined",}, // min - {"0 dB", "2 dB", "4 dB", "6 dB", "8 dB", "10 dB", "12 dB", "14 dB", "16 dB", "18 dB", "20 dB", - "22 dB", "24 dB", "24 dB", "24 dB", "undefined",}, // max -}; - -static const char *mic_sens_value[] = { - "-3 dB", "0 dB", "3 dB", "9 dB", "15 dB", "21 dB", "27 dB", "not used", -}; - -static const unsigned short AGC_atime[] = { - 11, 16, 11, 16, 21, 11, 16, 21, -}; - -static const unsigned short AGC_dtime[] = { - 100, 100, 200, 200, 200, 400, 400, 400, -}; - -static const char *AGC_level[] = { - "-9.0", "-11.5", "-15.0", "-17.5", -}; - -static const char *ig_small_value[] = { - "-3.0", "-2.5", "-2.0", "-1.5", "-1.0", "-0.5", -}; - -/* - * this was computed as peak_value[i] = pow((63-i)*1.42,1.013) - * - * UDA1341 datasheet on page 21: Peak value (dB) = (Peak level - 63.5)*5*log2 - * There is an table with these values [level]=value: [3]=-90.31, [7]=-84.29 - * [61]=-2.78, [62] = -1.48, [63] = 0.0 - * I tried to compute it, but using but even using logarithm with base either 10 or 2 - * i was'n able to get values in the table from the formula. So I constructed another - * formula (see above) to interpolate the values as good as possible. If there is some - * mistake, please contact me on tomas.kasparek@seznam.cz. Thanks. - * UDA1341TS datasheet is available at: - * http://www-us9.semiconductors.com/acrobat/datasheets/UDA1341TS_3.pdf - */ -static const char *peak_value[] = { - "-INF dB", "N.A.", "N.A", "90.31 dB", "N.A.", "N.A.", "N.A.", "-84.29 dB", - "-82.65 dB", "-81.13 dB", "-79.61 dB", "-78.09 dB", "-76.57 dB", "-75.05 dB", "-73.53 dB", - "-72.01 dB", "-70.49 dB", "-68.97 dB", "-67.45 dB", "-65.93 dB", "-64.41 dB", "-62.90 dB", - "-61.38 dB", "-59.86 dB", "-58.35 dB", "-56.83 dB", "-55.32 dB", "-53.80 dB", "-52.29 dB", - "-50.78 dB", "-49.26 dB", "-47.75 dB", "-46.24 dB", "-44.73 dB", "-43.22 dB", "-41.71 dB", - "-40.20 dB", "-38.69 dB", "-37.19 dB", "-35.68 dB", "-34.17 dB", "-32.67 dB", "-31.17 dB", - "-29.66 dB", "-28.16 dB", "-26.66 dB", "-25.16 dB", "-23.66 dB", "-22.16 dB", "-20.67 dB", - "-19.17 dB", "-17.68 dB", "-16.19 dB", "-14.70 dB", "-13.21 dB", "-11.72 dB", "-10.24 dB", - "-8.76 dB", "-7.28 dB", "-5.81 dB", "-4.34 dB", "-2.88 dB", "-1.43 dB", "0.00 dB", -}; - -static void snd_uda1341_proc_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct l3_client *clnt = entry->private_data; - struct uda1341 *uda = clnt->driver_data; - int peak; - - peak = snd_uda1341_codec_read(clnt, UDA1341_DATA1); - if (peak < 0) - peak = 0; - - snd_iprintf(buffer, "%s\n\n", uda->card->longname); - - // for information about computed values see UDA1341TS datasheet pages 15 - 21 - snd_iprintf(buffer, "DAC power : %s\n", uda->cfg[CMD_DAC] ? "on" : "off"); - snd_iprintf(buffer, "ADC power : %s\n", uda->cfg[CMD_ADC] ? "on" : "off"); - snd_iprintf(buffer, "Clock frequency : %s\n", fs_names[uda->cfg[CMD_FS]]); - snd_iprintf(buffer, "Data format : %s\n\n", format_names[uda->cfg[CMD_FORMAT]]); - - snd_iprintf(buffer, "Filter mode : %s\n", filter_names[uda->cfg[CMD_FILTER]]); - snd_iprintf(buffer, "Mixer mode : %s\n", mixer_names[uda->cfg[CMD_MIXER]]); - snd_iprintf(buffer, "De-emphasis : %s\n", deemp_names[uda->cfg[CMD_DEEMP]]); - snd_iprintf(buffer, "Peak detection pos. : %s\n", uda->cfg[CMD_PEAK] ? "after" : "before"); - snd_iprintf(buffer, "Peak value : %s\n\n", peak_value[peak]); - - snd_iprintf(buffer, "Automatic Gain Ctrl : %s\n", uda->cfg[CMD_AGC] ? "on" : "off"); - snd_iprintf(buffer, "AGC attack time : %d ms\n", AGC_atime[uda->cfg[CMD_AGC_TIME]]); - snd_iprintf(buffer, "AGC decay time : %d ms\n", AGC_dtime[uda->cfg[CMD_AGC_TIME]]); - snd_iprintf(buffer, "AGC output level : %s dB\n\n", AGC_level[uda->cfg[CMD_AGC_LEVEL]]); - - snd_iprintf(buffer, "Mute : %s\n", uda->cfg[CMD_MUTE] ? "on" : "off"); - - if (uda->cfg[CMD_VOLUME] == 0) - snd_iprintf(buffer, "Volume : 0 dB\n"); - else if (uda->cfg[CMD_VOLUME] < 62) - snd_iprintf(buffer, "Volume : %d dB\n", -1*uda->cfg[CMD_VOLUME] +1); - else - snd_iprintf(buffer, "Volume : -INF dB\n"); - snd_iprintf(buffer, "Bass : %s\n", bass_values[uda->cfg[CMD_FILTER]][uda->cfg[CMD_BASS]]); - snd_iprintf(buffer, "Trebble : %d dB\n", uda->cfg[CMD_FILTER] ? 2*uda->cfg[CMD_TREBBLE] : 0); - snd_iprintf(buffer, "Input Gain (6dB) : %s\n", uda->cfg[CMD_IGAIN] ? "on" : "off"); - snd_iprintf(buffer, "Output Gain (6dB) : %s\n", uda->cfg[CMD_OGAIN] ? "on" : "off"); - snd_iprintf(buffer, "Mic sensitivity : %s\n", mic_sens_value[uda->cfg[CMD_MIC]]); - - - if(uda->cfg[CMD_CH1] < 31) - snd_iprintf(buffer, "Mixer gain channel 1: -%d.%c dB\n", - ((uda->cfg[CMD_CH1] >> 1) * 3) + (uda->cfg[CMD_CH1] & 1), - uda->cfg[CMD_CH1] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Mixer gain channel 1: -INF dB\n"); - if(uda->cfg[CMD_CH2] < 31) - snd_iprintf(buffer, "Mixer gain channel 2: -%d.%c dB\n", - ((uda->cfg[CMD_CH2] >> 1) * 3) + (uda->cfg[CMD_CH2] & 1), - uda->cfg[CMD_CH2] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Mixer gain channel 2: -INF dB\n"); - - if(uda->cfg[CMD_IG] > 5) - snd_iprintf(buffer, "Input Amp. Gain ch 2: %d.%c dB\n", - (uda->cfg[CMD_IG] >> 1) -3, uda->cfg[CMD_IG] & 1 ? '5' : '0'); - else - snd_iprintf(buffer, "Input Amp. Gain ch 2: %s dB\n", ig_small_value[uda->cfg[CMD_IG]]); -} - -static void snd_uda1341_proc_regs_read(struct snd_info_entry *entry, - struct snd_info_buffer *buffer) -{ - struct l3_client *clnt = entry->private_data; - struct uda1341 *uda = clnt->driver_data; - int reg; - char buf[12]; - - for (reg = 0; reg < uda1341_reg_last; reg ++) { - if (reg == empty) - continue; - int2str_bin8(uda->regs[reg], buf); - snd_iprintf(buffer, "%s = %s\n", uda1341_reg_names[reg], buf); - } - - int2str_bin8(snd_uda1341_codec_read(clnt, UDA1341_DATA1), buf); - snd_iprintf(buffer, "DATA1 = %s\n", buf); -} -#endif /* CONFIG_PROC_FS */ - -static void __devinit snd_uda1341_proc_init(struct snd_card *card, struct l3_client *clnt) -{ - struct snd_info_entry *entry; - - if (! snd_card_proc_new(card, "uda1341", &entry)) - snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_read); - if (! snd_card_proc_new(card, "uda1341-regs", &entry)) - snd_info_set_text_ops(entry, clnt, snd_uda1341_proc_regs_read); -} - -/* }}} */ - -/* {{{ Mixer controls setting */ - -/* {{{ UDA1341 single functions */ - -#define UDA1341_SINGLE(xname, where, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_single, \ - .get = snd_uda1341_get_single, .put = snd_uda1341_put_single, \ - .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \ -} - -static int snd_uda1341_info_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 12) & 63; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_uda1341_get_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int mask = (kcontrol->private_value >> 12) & 63; - int invert = (kcontrol->private_value >> 18) & 1; - - ucontrol->value.integer.value[0] = uda->cfg[where]; - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - - return 0; -} - -static int snd_uda1341_put_single(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg = (kcontrol->private_value >> 5) & 15; - int shift = (kcontrol->private_value >> 9) & 7; - int mask = (kcontrol->private_value >> 12) & 63; - int invert = (kcontrol->private_value >> 18) & 1; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - - uda->cfg[where] = val; - return snd_uda1341_update_bits(clnt, reg, mask, shift, val, FLUSH); -} - -/* }}} */ - -/* {{{ UDA1341 enum functions */ - -#define UDA1341_ENUM(xname, where, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .info = snd_uda1341_info_enum, \ - .get = snd_uda1341_get_enum, .put = snd_uda1341_put_enum, \ - .private_value = where | (reg << 5) | (shift << 9) | (mask << 12) | (invert << 18) \ -} - -static int snd_uda1341_info_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int where = kcontrol->private_value & 31; - const char **texts; - - // this register we don't handle this way - if (!uda1341_enum_items[where]) - return -EINVAL; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 1; - uinfo->value.enumerated.items = uda1341_enum_items[where]; - - if (uinfo->value.enumerated.item >= uda1341_enum_items[where]) - uinfo->value.enumerated.item = uda1341_enum_items[where] - 1; - - texts = uda1341_enum_names[where]; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_uda1341_get_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - - ucontrol->value.enumerated.item[0] = uda->cfg[where]; - return 0; -} - -static int snd_uda1341_put_enum(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg = (kcontrol->private_value >> 5) & 15; - int shift = (kcontrol->private_value >> 9) & 7; - int mask = (kcontrol->private_value >> 12) & 63; - - uda->cfg[where] = (ucontrol->value.enumerated.item[0] & mask); - - return snd_uda1341_update_bits(clnt, reg, mask, shift, uda->cfg[where], FLUSH); -} - -/* }}} */ - -/* {{{ UDA1341 2regs functions */ - -#define UDA1341_2REGS(xname, where, reg_1, reg_2, shift_1, shift_2, mask_1, mask_2, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .info = snd_uda1341_info_2regs, \ - .get = snd_uda1341_get_2regs, .put = snd_uda1341_put_2regs, \ - .private_value = where | (reg_1 << 5) | (reg_2 << 9) | (shift_1 << 13) | (shift_2 << 16) | \ - (mask_1 << 19) | (mask_2 << 25) | (invert << 31) \ -} - - -static int snd_uda1341_info_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int mask; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_uda1341_get_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int invert = (kcontrol->private_value >> 31) & 1; - int mask; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - - ucontrol->value.integer.value[0] = uda->cfg[where]; - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_uda1341_put_2regs(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct l3_client *clnt = snd_kcontrol_chip(kcontrol); - struct uda1341 *uda = clnt->driver_data; - int where = kcontrol->private_value & 31; - int reg_1 = (kcontrol->private_value >> 5) & 15; - int reg_2 = (kcontrol->private_value >> 9) & 15; - int shift_1 = (kcontrol->private_value >> 13) & 7; - int shift_2 = (kcontrol->private_value >> 16) & 7; - int mask_1 = (kcontrol->private_value >> 19) & 63; - int mask_2 = (kcontrol->private_value >> 25) & 63; - int invert = (kcontrol->private_value >> 31) & 1; - int mask; - unsigned short val1, val2, val; - - val = ucontrol->value.integer.value[0]; - - mask = (mask_2 + 1) * (mask_1 + 1) - 1; - - val1 = val & mask_1; - val2 = (val / (mask_1 + 1)) & mask_2; - - if (invert) { - val1 = mask_1 - val1; - val2 = mask_2 - val2; - } - - uda->cfg[where] = invert ? mask - val : val; - - //FIXME - return value - snd_uda1341_update_bits(clnt, reg_1, mask_1, shift_1, val1, FLUSH); - return snd_uda1341_update_bits(clnt, reg_2, mask_2, shift_2, val2, FLUSH); -} - -/* }}} */ - -static struct snd_kcontrol_new snd_uda1341_controls[] = { - UDA1341_SINGLE("Master Playback Switch", CMD_MUTE, data0_2, 2, 1, 1), - UDA1341_SINGLE("Master Playback Volume", CMD_VOLUME, data0_0, 0, 63, 1), - - UDA1341_SINGLE("Bass Playback Volume", CMD_BASS, data0_1, 2, 15, 0), - UDA1341_SINGLE("Treble Playback Volume", CMD_TREBBLE, data0_1, 0, 3, 0), - - UDA1341_SINGLE("Input Gain Switch", CMD_IGAIN, stat1, 5, 1, 0), - UDA1341_SINGLE("Output Gain Switch", CMD_OGAIN, stat1, 6, 1, 0), - - UDA1341_SINGLE("Mixer Gain Channel 1 Volume", CMD_CH1, ext0, 0, 31, 1), - UDA1341_SINGLE("Mixer Gain Channel 2 Volume", CMD_CH2, ext1, 0, 31, 1), - - UDA1341_SINGLE("Mic Sensitivity Volume", CMD_MIC, ext2, 2, 7, 0), - - UDA1341_SINGLE("AGC Output Level", CMD_AGC_LEVEL, ext6, 0, 3, 0), - UDA1341_SINGLE("AGC Time Constant", CMD_AGC_TIME, ext6, 2, 7, 0), - UDA1341_SINGLE("AGC Time Constant Switch", CMD_AGC, ext4, 4, 1, 0), - - UDA1341_SINGLE("DAC Power", CMD_DAC, stat1, 0, 1, 0), - UDA1341_SINGLE("ADC Power", CMD_ADC, stat1, 1, 1, 0), - - UDA1341_ENUM("Peak detection", CMD_PEAK, data0_2, 5, 1, 0), - UDA1341_ENUM("De-emphasis", CMD_DEEMP, data0_2, 3, 3, 0), - UDA1341_ENUM("Mixer mode", CMD_MIXER, ext2, 0, 3, 0), - UDA1341_ENUM("Filter mode", CMD_FILTER, data0_2, 0, 3, 0), - - UDA1341_2REGS("Gain Input Amplifier Gain (channel 2)", CMD_IG, ext4, ext5, 0, 0, 3, 31, 0), -}; - -static void uda1341_free(struct l3_client *clnt) -{ - l3_detach_client(clnt); // calls kfree for driver_data (struct uda1341) - kfree(clnt); -} - -static int uda1341_dev_free(struct snd_device *device) -{ - struct l3_client *clnt = device->device_data; - uda1341_free(clnt); - return 0; -} - -int __init snd_chip_uda1341_mixer_new(struct snd_card *card, struct l3_client **clntp) -{ - static struct snd_device_ops ops = { - .dev_free = uda1341_dev_free, - }; - struct l3_client *clnt; - int idx, err; - - if (snd_BUG_ON(!card)) - return -EINVAL; - - clnt = kzalloc(sizeof(*clnt), GFP_KERNEL); - if (clnt == NULL) - return -ENOMEM; - - if ((err = l3_attach_client(clnt, "l3-bit-sa1100-gpio", UDA1341_ALSA_NAME))) { - kfree(clnt); - return err; - } - - for (idx = 0; idx < ARRAY_SIZE(snd_uda1341_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_uda1341_controls[idx], clnt))) < 0) { - uda1341_free(clnt); - return err; - } - } - - if ((err = snd_device_new(card, SNDRV_DEV_CODEC, clnt, &ops)) < 0) { - uda1341_free(clnt); - return err; - } - - *clntp = clnt; - strcpy(card->mixername, "UDA1341TS Mixer"); - ((struct uda1341 *)clnt->driver_data)->card = card; - - snd_uda1341_proc_init(card, clnt); - - return 0; -} - -/* }}} */ - -/* {{{ L3 operations */ - -static int uda1341_attach(struct l3_client *clnt) -{ - struct uda1341 *uda; - - uda = kzalloc(sizeof(*uda), 0, GFP_KERNEL); - if (!uda) - return -ENOMEM; - - /* init fixed parts of my copy of registers */ - uda->regs[stat0] = STAT0; - uda->regs[stat1] = STAT1; - - uda->regs[data0_0] = DATA0_0; - uda->regs[data0_1] = DATA0_1; - uda->regs[data0_2] = DATA0_2; - - uda->write = snd_uda1341_codec_write; - uda->read = snd_uda1341_codec_read; - - spin_lock_init(&uda->reg_lock); - - clnt->driver_data = uda; - return 0; -} - -static void uda1341_detach(struct l3_client *clnt) -{ - kfree(clnt->driver_data); -} - -static int -uda1341_command(struct l3_client *clnt, int cmd, void *arg) -{ - if (cmd != CMD_READ_REG) - return snd_uda1341_cfg_write(clnt, cmd, (int) arg, FLUSH); - - return snd_uda1341_codec_read(clnt, (int) arg); -} - -static int uda1341_open(struct l3_client *clnt) -{ - struct uda1341 *uda = clnt->driver_data; - - uda->active = 1; - - /* init default configuration */ - snd_uda1341_cfg_write(clnt, CMD_RESET, 0, REGS_ONLY); - snd_uda1341_cfg_write(clnt, CMD_FS, F256, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_FORMAT, LSB16, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_OGAIN, ON, FLUSH); // default off after reset - snd_uda1341_cfg_write(clnt, CMD_IGAIN, ON, FLUSH); // default off after reset - snd_uda1341_cfg_write(clnt, CMD_DAC, ON, FLUSH); // ??? default value after reset - snd_uda1341_cfg_write(clnt, CMD_ADC, ON, FLUSH); // ??? default value after reset - snd_uda1341_cfg_write(clnt, CMD_VOLUME, 20, FLUSH); // default 0dB after reset - snd_uda1341_cfg_write(clnt, CMD_BASS, 0, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_TREBBLE, 0, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_PEAK, AFTER, REGS_ONLY);// default value after reset - snd_uda1341_cfg_write(clnt, CMD_DEEMP, NONE, REGS_ONLY);// default value after reset - //at this moment should be QMUTED by h3600_audio_init - snd_uda1341_cfg_write(clnt, CMD_MUTE, OFF, REGS_ONLY); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_FILTER, MAX, FLUSH); // defaul flat after reset - snd_uda1341_cfg_write(clnt, CMD_CH1, 31, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_CH2, 4, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_MIC, 4, FLUSH); // default 0dB after reset - snd_uda1341_cfg_write(clnt, CMD_MIXER, MIXER, FLUSH); // default doub.dif.mode - snd_uda1341_cfg_write(clnt, CMD_AGC, OFF, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_IG, 0, FLUSH); // unknown state after reset - snd_uda1341_cfg_write(clnt, CMD_AGC_TIME, 0, FLUSH); // default value after reset - snd_uda1341_cfg_write(clnt, CMD_AGC_LEVEL, 0, FLUSH); // default value after reset - - return 0; -} - -static void uda1341_close(struct l3_client *clnt) -{ - struct uda1341 *uda = clnt->driver_data; - - uda->active = 0; -} - -/* }}} */ - -/* {{{ Module and L3 initialization */ - -static struct l3_ops uda1341_ops = { - .open = uda1341_open, - .command = uda1341_command, - .close = uda1341_close, -}; - -static struct l3_driver uda1341_driver = { - .name = UDA1341_ALSA_NAME, - .attach_client = uda1341_attach, - .detach_client = uda1341_detach, - .ops = &uda1341_ops, - .owner = THIS_MODULE, -}; - -static int __init uda1341_init(void) -{ - return l3_add_driver(&uda1341_driver); -} - -static void __exit uda1341_exit(void) -{ - l3_del_driver(&uda1341_driver); -} - -module_init(uda1341_init); -module_exit(uda1341_exit); - -MODULE_AUTHOR("Tomas Kasparek <tomas.kasparek@seznam.cz>"); -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Philips UDA1341 CODEC driver for ALSA"); -MODULE_SUPPORTED_DEVICE("{{UDA1341,UDA1341TS}}"); - -EXPORT_SYMBOL(snd_chip_uda1341_mixer_new); - -/* }}} */ - -/* - * Local variables: - * indent-tabs-mode: t - * End: - */ diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index 9d98a66..d31c373 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -24,6 +24,7 @@ #include <linux/delay.h> #include <linux/interrupt.h> #include <linux/init.h> +#include <linux/version.h> #include <sound/core.h> #include <sound/tea575x-tuner.h> @@ -31,6 +32,13 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_DESCRIPTION("Routines for control of TEA5757/5759 Philips AM/FM radio tuner chips"); MODULE_LICENSE("GPL"); +static int radio_nr = -1; +module_param(radio_nr, int, 0); + +#define RADIO_VERSION KERNEL_VERSION(0, 0, 2) +#define FREQ_LO (87 * 16000) +#define FREQ_HI (108 * 16000) + /* * definitions */ @@ -53,6 +61,17 @@ MODULE_LICENSE("GPL"); #define TEA575X_BIT_DUMMY (1<<15) /* buffer */ #define TEA575X_BIT_FREQ_MASK 0x7fff +static struct v4l2_queryctrl radio_qctrl[] = { + { + .id = V4L2_CID_AUDIO_MUTE, + .name = "Mute", + .minimum = 0, + .maximum = 1, + .default_value = 1, + .type = V4L2_CTRL_TYPE_BOOLEAN, + } +}; + /* * lowlevel part */ @@ -84,94 +103,146 @@ static void snd_tea575x_set_freq(struct snd_tea575x *tea) * Linux Video interface */ -static long snd_tea575x_ioctl(struct file *file, - unsigned int cmd, unsigned long data) +static int vidioc_querycap(struct file *file, void *priv, + struct v4l2_capability *v) { struct snd_tea575x *tea = video_drvdata(file); - void __user *arg = (void __user *)data; - - switch(cmd) { - case VIDIOCGCAP: - { - struct video_capability v; - v.type = VID_TYPE_TUNER; - v.channels = 1; - v.audios = 1; - /* No we don't do pictures */ - v.maxwidth = 0; - v.maxheight = 0; - v.minwidth = 0; - v.minheight = 0; - strcpy(v.name, tea->tea5759 ? "TEA5759" : "TEA5757"); - if (copy_to_user(arg,&v,sizeof(v))) - return -EFAULT; - return 0; - } - case VIDIOCGTUNER: - { - struct video_tuner v; - if (copy_from_user(&v, arg,sizeof(v))!=0) - return -EFAULT; - if (v.tuner) /* Only 1 tuner */ - return -EINVAL; - v.rangelow = (87*16000); - v.rangehigh = (108*16000); - v.flags = VIDEO_TUNER_LOW; - v.mode = VIDEO_MODE_AUTO; - strcpy(v.name, "FM"); - v.signal = 0xFFFF; - if (copy_to_user(arg, &v, sizeof(v))) - return -EFAULT; - return 0; - } - case VIDIOCSTUNER: - { - struct video_tuner v; - if(copy_from_user(&v, arg, sizeof(v))) - return -EFAULT; - if(v.tuner!=0) - return -EINVAL; - /* Only 1 tuner so no setting needed ! */ + + strcpy(v->card, tea->tea5759 ? "TEA5759" : "TEA5757"); + strlcpy(v->driver, "tea575x-tuner", sizeof(v->driver)); + strlcpy(v->card, "Maestro Radio", sizeof(v->card)); + sprintf(v->bus_info, "PCI"); + v->version = RADIO_VERSION; + v->capabilities = V4L2_CAP_TUNER; + return 0; +} + +static int vidioc_g_tuner(struct file *file, void *priv, + struct v4l2_tuner *v) +{ + if (v->index > 0) + return -EINVAL; + + strcpy(v->name, "FM"); + v->type = V4L2_TUNER_RADIO; + v->rangelow = FREQ_LO; + v->rangehigh = FREQ_HI; + v->rxsubchans = V4L2_TUNER_SUB_MONO|V4L2_TUNER_SUB_STEREO; + v->capability = V4L2_TUNER_CAP_LOW; + v->audmode = V4L2_TUNER_MODE_MONO; + v->signal = 0xffff; + return 0; +} + +static int vidioc_s_tuner(struct file *file, void *priv, + struct v4l2_tuner *v) +{ + if (v->index > 0) + return -EINVAL; + return 0; +} + +static int vidioc_g_frequency(struct file *file, void *priv, + struct v4l2_frequency *f) +{ + struct snd_tea575x *tea = video_drvdata(file); + + f->type = V4L2_TUNER_RADIO; + f->frequency = tea->freq; + return 0; +} + +static int vidioc_s_frequency(struct file *file, void *priv, + struct v4l2_frequency *f) +{ + struct snd_tea575x *tea = video_drvdata(file); + + if (f->frequency < FREQ_LO || f->frequency > FREQ_HI) + return -EINVAL; + + tea->freq = f->frequency; + + snd_tea575x_set_freq(tea); + + return 0; +} + +static int vidioc_g_audio(struct file *file, void *priv, + struct v4l2_audio *a) +{ + if (a->index > 1) + return -EINVAL; + + strcpy(a->name, "Radio"); + a->capability = V4L2_AUDCAP_STEREO; + return 0; +} + +static int vidioc_s_audio(struct file *file, void *priv, + struct v4l2_audio *a) +{ + if (a->index != 0) + return -EINVAL; + return 0; +} + +static int vidioc_queryctrl(struct file *file, void *priv, + struct v4l2_queryctrl *qc) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(radio_qctrl); i++) { + if (qc->id && qc->id == radio_qctrl[i].id) { + memcpy(qc, &(radio_qctrl[i]), + sizeof(*qc)); return 0; } - case VIDIOCGFREQ: - if(copy_to_user(arg, &tea->freq, sizeof(tea->freq))) - return -EFAULT; - return 0; - case VIDIOCSFREQ: - if(copy_from_user(&tea->freq, arg, sizeof(tea->freq))) - return -EFAULT; - snd_tea575x_set_freq(tea); - return 0; - case VIDIOCGAUDIO: - { - struct video_audio v; - memset(&v, 0, sizeof(v)); - strcpy(v.name, "Radio"); - if(copy_to_user(arg,&v, sizeof(v))) - return -EFAULT; + } + return -EINVAL; +} + +static int vidioc_g_ctrl(struct file *file, void *priv, + struct v4l2_control *ctrl) +{ + struct snd_tea575x *tea = video_drvdata(file); + + switch (ctrl->id) { + case V4L2_CID_AUDIO_MUTE: + if (tea->ops->mute) { + ctrl->value = tea->mute; return 0; } - case VIDIOCSAUDIO: - { - struct video_audio v; - if(copy_from_user(&v, arg, sizeof(v))) - return -EFAULT; - if (tea->ops->mute) - tea->ops->mute(tea, - (v.flags & - VIDEO_AUDIO_MUTE) ? 1 : 0); - if(v.audio) - return -EINVAL; + } + return -EINVAL; +} + +static int vidioc_s_ctrl(struct file *file, void *priv, + struct v4l2_control *ctrl) +{ + struct snd_tea575x *tea = video_drvdata(file); + + switch (ctrl->id) { + case V4L2_CID_AUDIO_MUTE: + if (tea->ops->mute) { + tea->ops->mute(tea, ctrl->value); + tea->mute = 1; return 0; } - default: - return -ENOIOCTLCMD; } + return -EINVAL; +} + +static int vidioc_g_input(struct file *filp, void *priv, unsigned int *i) +{ + *i = 0; + return 0; } -static void snd_tea575x_release(struct video_device *vfd) +static int vidioc_s_input(struct file *filp, void *priv, unsigned int i) { + if (i != 0) + return -EINVAL; + return 0; } static int snd_tea575x_exclusive_open(struct file *file) @@ -189,50 +260,91 @@ static int snd_tea575x_exclusive_release(struct file *file) return 0; } +static const struct v4l2_file_operations tea575x_fops = { + .owner = THIS_MODULE, + .open = snd_tea575x_exclusive_open, + .release = snd_tea575x_exclusive_release, + .ioctl = video_ioctl2, +}; + +static const struct v4l2_ioctl_ops tea575x_ioctl_ops = { + .vidioc_querycap = vidioc_querycap, + .vidioc_g_tuner = vidioc_g_tuner, + .vidioc_s_tuner = vidioc_s_tuner, + .vidioc_g_audio = vidioc_g_audio, + .vidioc_s_audio = vidioc_s_audio, + .vidioc_g_input = vidioc_g_input, + .vidioc_s_input = vidioc_s_input, + .vidioc_g_frequency = vidioc_g_frequency, + .vidioc_s_frequency = vidioc_s_frequency, + .vidioc_queryctrl = vidioc_queryctrl, + .vidioc_g_ctrl = vidioc_g_ctrl, + .vidioc_s_ctrl = vidioc_s_ctrl, +}; + +static struct video_device tea575x_radio = { + .name = "tea575x-tuner", + .fops = &tea575x_fops, + .ioctl_ops = &tea575x_ioctl_ops, + .release = video_device_release, +}; + /* * initialize all the tea575x chips */ void snd_tea575x_init(struct snd_tea575x *tea) { + int retval; unsigned int val; + struct video_device *tea575x_radio_inst; val = tea->ops->read(tea); if (val == 0x1ffffff || val == 0) { - snd_printk(KERN_ERR "Cannot find TEA575x chip\n"); + snd_printk(KERN_ERR + "tea575x-tuner: Cannot find TEA575x chip\n"); return; } - memset(&tea->vd, 0, sizeof(tea->vd)); - strcpy(tea->vd.name, tea->tea5759 ? "TEA5759 radio" : "TEA5757 radio"); - tea->vd.release = snd_tea575x_release; - video_set_drvdata(&tea->vd, tea); - tea->vd.fops = &tea->fops; tea->in_use = 0; - tea->fops.owner = tea->card->module; - tea->fops.open = snd_tea575x_exclusive_open; - tea->fops.release = snd_tea575x_exclusive_release; - tea->fops.ioctl = snd_tea575x_ioctl; - if (video_register_device(&tea->vd, VFL_TYPE_RADIO, tea->dev_nr - 1) < 0) { - snd_printk(KERN_ERR "unable to register tea575x tuner\n"); + tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; + tea->freq = 90500 * 16; /* 90.5Mhz default */ + + tea575x_radio_inst = video_device_alloc(); + if (tea575x_radio_inst == NULL) { + printk(KERN_ERR "tea575x-tuner: not enough memory\n"); return; } - tea->vd_registered = 1; - tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; - tea->freq = 90500 * 16; /* 90.5Mhz default */ + memcpy(tea575x_radio_inst, &tea575x_radio, sizeof(tea575x_radio)); + + strcpy(tea575x_radio.name, tea->tea5759 ? + "TEA5759 radio" : "TEA5757 radio"); + + video_set_drvdata(tea575x_radio_inst, tea); + + retval = video_register_device(tea575x_radio_inst, + VFL_TYPE_RADIO, radio_nr); + if (retval) { + printk(KERN_ERR "tea575x-tuner: can't register video device!\n"); + kfree(tea575x_radio_inst); + return; + } snd_tea575x_set_freq(tea); /* mute on init */ - if (tea->ops->mute) + if (tea->ops->mute) { tea->ops->mute(tea, 1); + tea->mute = 1; + } + tea->vd = tea575x_radio_inst; } void snd_tea575x_exit(struct snd_tea575x *tea) { - if (tea->vd_registered) { - video_unregister_device(&tea->vd); - tea->vd_registered = 0; + if (tea->vd) { + video_unregister_device(tea->vd); + tea->vd = NULL; } } diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index ce0aa04..c5c9a92 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -56,8 +56,8 @@ config SND_AD1848 Say Y here to include support for AD1848 (Analog Devices) or CS4248 (Cirrus Logic - Crystal Semiconductors) chips. - For newer chips from Cirrus Logic, use the CS4231, CS4232 or - CS4236+ drivers. + For newer chips from Cirrus Logic, use the CS4231 or CS4232+ + drivers. To compile this driver as a module, choose M here: the module will be called snd-ad1848. @@ -94,6 +94,8 @@ config SND_CMI8330 tristate "C-Media CMI8330" select SND_WSS_LIB select SND_SB16_DSP + select SND_OPL3_LIB + select SND_MPU401_UART help Say Y here to include support for soundcards based on the C-Media CMI8330 chip. @@ -112,26 +114,15 @@ config SND_CS4231 To compile this driver as a module, choose M here: the module will be called snd-cs4231. -config SND_CS4232 - tristate "Generic Cirrus Logic CS4232 driver" - select SND_OPL3_LIB - select SND_MPU401_UART - select SND_WSS_LIB - help - Say Y here to include support for CS4232 chips from Cirrus - Logic - Crystal Semiconductors. - - To compile this driver as a module, choose M here: the module - will be called snd-cs4232. - config SND_CS4236 - tristate "Generic Cirrus Logic CS4236+ driver" + tristate "Generic Cirrus Logic CS4232/CS4236+ driver" select SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB help - Say Y to include support for CS4235,CS4236,CS4237B,CS4238B, - CS4239 chips from Cirrus Logic - Crystal Semiconductors. + Say Y to include support for CS4232,CS4235,CS4236,CS4237B, + CS4238B,CS4239 chips from Cirrus Logic - Crystal + Semiconductors. To compile this driver as a module, choose M here: the module will be called snd-cs4236. @@ -377,14 +368,17 @@ config SND_SGALAXY will be called snd-sgalaxy. config SND_SSCAPE - tristate "Ensoniq SoundScape PnP driver" + tristate "Ensoniq SoundScape driver" select SND_HWDEP select SND_MPU401_UART select SND_WSS_LIB help - Say Y here to include support for Ensoniq SoundScape PnP + Say Y here to include support for Ensoniq SoundScape soundcards. + The PCM audio is supported on SoundScape Classic, Elite, PnP + and VIVO cards. The MIDI support is very experimental. + To compile this driver as a module, choose M here: the module will be called snd-sscape. @@ -411,5 +405,36 @@ config SND_WAVEFRONT_FIRMWARE_IN_KERNEL you need to install the firmware files from the alsa-firmware package. +config SND_MSND_PINNACLE + tristate "Turtle Beach MultiSound Pinnacle/Fiji driver" + depends on X86 && EXPERIMENTAL + select FW_LOADER + select SND_MPU401_UART + select SND_PCM + help + Say Y to include support for Turtle Beach MultiSound Pinnacle/ + Fiji soundcards. + + To compile this driver as a module, choose M here: the module + will be called snd-msnd-pinnacle. + +config SND_MSND_CLASSIC + tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey" + depends on X86 && EXPERIMENTAL + select FW_LOADER + select SND_MPU401_UART + select SND_PCM + help + Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or + Monterey (not for the Pinnacle or Fiji). + + See <file:Documentation/sound/oss/MultiSound> for important information + about this driver. Note that it has been discontinued, but the + Voyetra Turtle Beach knowledge base entry for it is still available + at <http://www.turtlebeach.com/site/kb_ftp/790.asp>. + + To compile this driver as a module, choose M here: the module + will be called snd-msnd-classic. + endif # SND_ISA diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 63af13d..b906b9a 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -26,5 +26,5 @@ obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o -obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ opti9xx/ \ +obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ msnd/ opti9xx/ \ sb/ wavefront/ wss/ diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 9660e59..bbcbf92 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -156,6 +156,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard struct snd_card_ad1816a *acard; struct snd_ad1816a *chip; struct snd_opl3 *opl3; + struct snd_timer *timer; error = snd_card_create(index[dev], id[dev], THIS_MODULE, sizeof(struct snd_card_ad1816a), &card); @@ -195,6 +196,12 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard return error; } + error = snd_ad1816a_timer(chip, 0, &timer); + if (error < 0) { + snd_card_free(card); + return error; + } + if (mpu_port[dev] > 0) { if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, mpu_port[dev], 0, mpu_irq[dev], IRQF_DISABLED, @@ -208,11 +215,8 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard OPL3_HW_AUTO, 0, &opl3) < 0) { printk(KERN_ERR PFX "no OPL device at 0x%lx-0x%lx.\n", fm_port[dev], fm_port[dev] + 2); } else { - if ((error = snd_opl3_timer_new(opl3, 1, 2)) < 0) { - snd_card_free(card); - return error; - } - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, NULL)) < 0) { + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) { snd_card_free(card); return error; } diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 3bfca7c..05aef8b 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -37,7 +37,7 @@ static inline int snd_ad1816a_busy_wait(struct snd_ad1816a *chip) if (inb(AD1816A_REG(AD1816A_CHIP_STATUS)) & AD1816A_READY) return 0; - snd_printk("chip busy.\n"); + snd_printk(KERN_WARNING "chip busy.\n"); return -EBUSY; } @@ -196,7 +196,7 @@ static int snd_ad1816a_trigger(struct snd_ad1816a *chip, unsigned char what, spin_unlock(&chip->lock); break; default: - snd_printk("invalid trigger mode 0x%x.\n", what); + snd_printk(KERN_WARNING "invalid trigger mode 0x%x.\n", what); error = -EINVAL; } @@ -377,7 +377,6 @@ static struct snd_pcm_hardware snd_ad1816a_capture = { .fifo_size = 0, }; -#if 0 /* not used now */ static int snd_ad1816a_timer_close(struct snd_timer *timer) { struct snd_ad1816a *chip = snd_timer_chip(timer); @@ -442,8 +441,6 @@ static struct snd_timer_hardware snd_ad1816a_timer_table = { .start = snd_ad1816a_timer_start, .stop = snd_ad1816a_timer_stop, }; -#endif /* not used now */ - static int snd_ad1816a_playback_open(struct snd_pcm_substream *substream) { @@ -568,7 +565,7 @@ static const char __devinit *snd_ad1816a_chip_id(struct snd_ad1816a *chip) case AD1816A_HW_AD1815: return "AD1815"; case AD1816A_HW_AD18MAX10: return "AD18max10"; default: - snd_printk("Unknown chip version %d:%d.\n", + snd_printk(KERN_WARNING "Unknown chip version %d:%d.\n", chip->version, chip->hardware); return "AD1816A - unknown"; } @@ -687,7 +684,6 @@ int __devinit snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_p return 0; } -#if 0 /* not used now */ int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd_timer **rtimer) { struct snd_timer *timer; @@ -709,7 +705,6 @@ int __devinit snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd *rtimer = timer; return 0; } -#endif /* not used now */ /* * diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 24e6090..de83608 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -31,11 +31,11 @@ * To quickly load the module, * * modprobe -a snd-cmi8330 sbport=0x220 sbirq=5 sbdma8=1 - * sbdma16=5 wssport=0x530 wssirq=11 wssdma=0 + * sbdma16=5 wssport=0x530 wssirq=11 wssdma=0 fmport=0x388 * * This card has two mixers and two PCM devices. I've cheesed it such * that recording and playback can be done through the same device. - * The driver "magically" routes the capturing to the AD1848 codec, + * The driver "magically" routes the capturing to the CMI8330 codec, * and playback to the SB16 codec. This allows for full-duplex mode * to some extent. * The utilities in alsa-utils are aware of both devices, so passing @@ -51,6 +51,8 @@ #include <linux/moduleparam.h> #include <sound/core.h> #include <sound/wss.h> +#include <sound/opl3.h> +#include <sound/mpu401.h> #include <sound/sb.h> #include <sound/initval.h> @@ -79,6 +81,9 @@ static int sbdma16[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; static int wssirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; static int wssdma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static long fmport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long mpuport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int mpuirq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for CMI8330 soundcard."); @@ -107,6 +112,12 @@ MODULE_PARM_DESC(wssirq, "IRQ # for CMI8330 WSS driver."); module_param_array(wssdma, int, NULL, 0444); MODULE_PARM_DESC(wssdma, "DMA for CMI8330 WSS driver."); +module_param_array(fmport, long, NULL, 0444); +MODULE_PARM_DESC(fmport, "FM port # for CMI8330 driver."); +module_param_array(mpuport, long, NULL, 0444); +MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8330 driver."); +module_param_array(mpuirq, int, NULL, 0444); +MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8330 MPU-401 port."); #ifdef CONFIG_PNP static int isa_registered; static int pnp_registered; @@ -149,6 +160,7 @@ struct snd_cmi8330 { #ifdef CONFIG_PNP struct pnp_dev *cap; struct pnp_dev *play; + struct pnp_dev *mpu; #endif struct snd_card *card; struct snd_wss *wss; @@ -165,7 +177,7 @@ struct snd_cmi8330 { #ifdef CONFIG_PNP static struct pnp_card_device_id snd_cmi8330_pnpids[] = { - { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" } } }, + { .id = "CMI0001", .devs = { { "@@@0001" }, { "@X@0001" }, { "@H@0001" } } }, { .id = "" } }; @@ -219,8 +231,10 @@ WSS_SINGLE("3D Control - Switch", 0, CMI8330_RMUX3D, 5, 1, 1), WSS_SINGLE("PC Speaker Playback Volume", 0, CMI8330_OUTPUTVOL, 3, 3, 0), -WSS_SINGLE("FM Playback Switch", 0, - CMI8330_RECMUX, 3, 1, 1), +WSS_DOUBLE("FM Playback Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +WSS_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", CAPTURE, SWITCH), 0, CMI8330_RMUX3D, 7, 1, 1), WSS_SINGLE(SNDRV_CTL_NAME_IEC958("Input ", PLAYBACK, SWITCH), 0, @@ -323,16 +337,21 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, if (acard->play == NULL) return -EBUSY; + acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); + if (acard->play == NULL) + return -EBUSY; + pdev = acard->cap; err = pnp_activate_dev(pdev); if (err < 0) { - snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n"); + snd_printk(KERN_ERR "CMI8330/C3D PnP configure failure\n"); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); wssdma[dev] = pnp_dma(pdev, 0); wssirq[dev] = pnp_irq(pdev, 0); + fmport[dev] = pnp_port_start(pdev, 1); /* allocate SB16 resources */ pdev = acard->play; @@ -347,6 +366,17 @@ static int __devinit snd_cmi8330_pnp(int dev, struct snd_cmi8330 *acard, sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); + /* allocate MPU-401 resources */ + pdev = acard->mpu; + + err = pnp_activate_dev(pdev); + if (err < 0) { + snd_printk(KERN_ERR + "CMI8330/C3D (MPU-401) PnP configure failure\n"); + return -EBUSY; + } + mpuport[dev] = pnp_port_start(pdev, 0); + mpuirq[dev] = pnp_irq(pdev, 0); return 0; } #endif @@ -489,6 +519,7 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) { struct snd_cmi8330 *acard; int i, err; + struct snd_opl3 *opl3; acard = card->private_data; err = snd_wss_create(card, wssport[dev] + 4, -1, @@ -496,11 +527,11 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) wssdma[dev], -1, WSS_HW_DETECT, 0, &acard->wss); if (err < 0) { - snd_printk(KERN_ERR PFX "(AD1848) device busy??\n"); + snd_printk(KERN_ERR PFX "(CMI8330) device busy??\n"); return err; } if (acard->wss->hardware != WSS_HW_CMI8330) { - snd_printk(KERN_ERR PFX "(AD1848) not found during probe\n"); + snd_printk(KERN_ERR PFX "(CMI8330) not found during probe\n"); return -ENODEV; } @@ -532,6 +563,27 @@ static int __devinit snd_cmi8330_probe(struct snd_card *card, int dev) snd_printk(KERN_ERR PFX "failed to create pcms\n"); return err; } + if (fmport[dev] != SNDRV_AUTO_PORT) { + if (snd_opl3_create(card, + fmport[dev], fmport[dev] + 2, + OPL3_HW_AUTO, 0, &opl3) < 0) { + snd_printk(KERN_ERR PFX + "no OPL device at 0x%lx-0x%lx ?\n", + fmport[dev], fmport[dev] + 2); + } else { + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + return err; + } + } + + if (mpuport[dev] != SNDRV_AUTO_PORT) { + if (snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpuport[dev], 0, mpuirq[dev], + IRQF_DISABLED, NULL) < 0) + printk(KERN_ERR PFX "no MPU-401 device at 0x%lx.\n", + mpuport[dev]); + } strcpy(card->driver, "CMI8330/C3D"); strcpy(card->shortname, "C-Media CMI8330/C3D"); diff --git a/sound/isa/cs423x/Makefile b/sound/isa/cs423x/Makefile index 5870ca2..6d397e8 100644 --- a/sound/isa/cs423x/Makefile +++ b/sound/isa/cs423x/Makefile @@ -3,13 +3,11 @@ # Copyright (c) 2001 by Jaroslav Kysela <perex@perex.cz> # -snd-cs4236-lib-objs := cs4236_lib.o snd-cs4231-objs := cs4231.o -snd-cs4232-objs := cs4232.o -snd-cs4236-objs := cs4236.o +snd-cs4236-objs := cs4236.o cs4236_lib.o # Toplevel Module Dependency obj-$(CONFIG_SND_CS4231) += snd-cs4231.o -obj-$(CONFIG_SND_CS4232) += snd-cs4232.o -obj-$(CONFIG_SND_CS4236) += snd-cs4236.o snd-cs4236-lib.o +obj-$(CONFIG_SND_CS4236) += snd-cs4236.o + diff --git a/sound/isa/cs423x/cs4232.c b/sound/isa/cs423x/cs4232.c deleted file mode 100644 index 9fad2e6..0000000 --- a/sound/isa/cs423x/cs4232.c +++ /dev/null @@ -1,2 +0,0 @@ -#define CS4232 -#include "cs4236.c" diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index f784598..a076a6c 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -33,17 +33,14 @@ MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); MODULE_LICENSE("GPL"); -#ifdef CS4232 -MODULE_DESCRIPTION("Cirrus Logic CS4232"); +MODULE_DESCRIPTION("Cirrus Logic CS4232-9"); MODULE_SUPPORTED_DEVICE("{{Turtle Beach,TBS-2000}," "{Turtle Beach,Tropez Plus}," "{SIC CrystalWave 32}," "{Hewlett Packard,Omnibook 5500}," "{TerraTec,Maestro 32/96}," - "{Philips,PCA70PS}}"); -#else -MODULE_DESCRIPTION("Cirrus Logic CS4235-9"); -MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235}," + "{Philips,PCA70PS}}," + "{{Crystal Semiconductors,CS4235}," "{Crystal Semiconductors,CS4236}," "{Crystal Semiconductors,CS4237}," "{Crystal Semiconductors,CS4238}," @@ -70,15 +67,11 @@ MODULE_SUPPORTED_DEVICE("{{Crystal Semiconductors,CS4235}," "{Typhoon Soundsystem,CS4236B}," "{Turtle Beach,Malibu}," "{Unknown,Digital PC 5000 Onboard}}"); -#endif -#ifdef CS4232 -#define IDENT "CS4232" -#define DEV_NAME "cs4232" -#else -#define IDENT "CS4236+" -#define DEV_NAME "cs4236" -#endif +MODULE_ALIAS("snd_cs4232"); + +#define IDENT "CS4232+" +#define DEV_NAME "cs4232+" static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ @@ -128,9 +121,7 @@ MODULE_PARM_DESC(dma2, "DMA2 # for " IDENT " driver."); #ifdef CONFIG_PNP static int isa_registered; static int pnpc_registered; -#ifdef CS4232 static int pnp_registered; -#endif #endif /* CONFIG_PNP */ struct snd_card_cs4236 { @@ -145,11 +136,10 @@ struct snd_card_cs4236 { #ifdef CONFIG_PNP -#ifdef CS4232 /* * PNP BIOS */ -static const struct pnp_device_id snd_cs4232_pnpbiosids[] = { +static const struct pnp_device_id snd_cs423x_pnpbiosids[] = { { .id = "CSC0100" }, { .id = "CSC0000" }, /* Guillemot Turtlebeach something appears to be cs4232 compatible @@ -157,10 +147,8 @@ static const struct pnp_device_id snd_cs4232_pnpbiosids[] = { { .id = "GIM0100" }, { .id = "" } }; -MODULE_DEVICE_TABLE(pnp, snd_cs4232_pnpbiosids); -#endif /* CS4232 */ +MODULE_DEVICE_TABLE(pnp, snd_cs423x_pnpbiosids); -#ifdef CS4232 #define CS423X_ISAPNP_DRIVER "cs4232_isapnp" static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* Philips PCA70PS */ @@ -179,12 +167,6 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { { .id = "CSCf032", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Netfinity 3000 on-board soundcard */ { .id = "CSCe825", .devs = { { "CSC0100" }, { "CSC0110" }, { "CSC010f" } } }, - /* --- */ - { .id = "" } /* end */ -}; -#else /* CS4236 */ -#define CS423X_ISAPNP_DRIVER "cs4236_isapnp" -static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* Intel Marlin Spike Motherboard - CS4235 */ { .id = "CSC0225", .devs = { { "CSC0000" }, { "CSC0010" }, { "CSC0003" } } }, /* Intel Marlin Spike Motherboard (#2) - CS4235 */ @@ -266,7 +248,6 @@ static struct pnp_card_device_id snd_cs423x_pnpids[] = { /* --- */ { .id = "" } /* end */ }; -#endif MODULE_DEVICE_TABLE(pnp_card, snd_cs423x_pnpids); @@ -323,17 +304,19 @@ static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) return 0; } -#ifdef CS4232 -static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard, - struct pnp_dev *pdev) +static int __devinit snd_card_cs423x_pnp(int dev, struct snd_card_cs4236 *acard, + struct pnp_dev *pdev, + struct pnp_dev *cdev) { acard->wss = pdev; if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) return -EBUSY; - cport[dev] = -1; + if (cdev) + cport[dev] = pnp_port_start(cdev, 0); + else + cport[dev] = -1; return 0; } -#endif static int __devinit snd_card_cs423x_pnpc(int dev, struct snd_card_cs4236 *acard, struct pnp_card_link *card, @@ -411,40 +394,39 @@ static int __devinit snd_cs423x_probe(struct snd_card *card, int dev) return -EBUSY; } -#ifdef CS4232 err = snd_wss_create(card, port[dev], cport[dev], irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; - - err = snd_wss_pcm(chip, 0, &pcm); - if (err < 0) - return err; - - err = snd_wss_mixer(chip); + WSS_HW_DETECT3, 0, &chip); if (err < 0) return err; - -#else /* CS4236 */ - err = snd_cs4236_create(card, - port[dev], cport[dev], - irq[dev], dma1[dev], dma2[dev], - WSS_HW_DETECT, 0, &chip); - if (err < 0) - return err; - acard->chip = chip; - - err = snd_cs4236_pcm(chip, 0, &pcm); - if (err < 0) - return err; - - err = snd_cs4236_mixer(chip); - if (err < 0) - return err; -#endif + if (chip->hardware & WSS_HW_CS4236B_MASK) { + snd_wss_free(chip); + err = snd_cs4236_create(card, + port[dev], cport[dev], + irq[dev], dma1[dev], dma2[dev], + WSS_HW_DETECT, 0, &chip); + if (err < 0) + return err; + acard->chip = chip; + + err = snd_cs4236_pcm(chip, 0, &pcm); + if (err < 0) + return err; + + err = snd_cs4236_mixer(chip); + if (err < 0) + return err; + } else { + acard->chip = chip; + err = snd_wss_pcm(chip, 0, &pcm); + if (err < 0) + return err; + + err = snd_wss_mixer(chip); + if (err < 0) + return err; + } strcpy(card->driver, pcm->name); strcpy(card->shortname, pcm->name); sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", @@ -579,13 +561,14 @@ static struct isa_driver cs423x_isa_driver = { #ifdef CONFIG_PNP -#ifdef CS4232 -static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, +static int __devinit snd_cs423x_pnpbios_detect(struct pnp_dev *pdev, const struct pnp_device_id *id) { static int dev; int err; struct snd_card *card; + struct pnp_dev *cdev; + char cid[PNP_ID_LEN]; if (pnp_device_is_isapnp(pdev)) return -ENOENT; /* we have another procedure - card */ @@ -596,10 +579,19 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, if (dev >= SNDRV_CARDS) return -ENODEV; + /* prepare second id */ + strcpy(cid, pdev->id[0].id); + cid[5] = '1'; + cdev = NULL; + list_for_each_entry(cdev, &(pdev->protocol->devices), protocol_list) { + if (!strcmp(cdev->id[0].id, cid)) + break; + } err = snd_cs423x_card_new(dev, &card); if (err < 0) return err; - if ((err = snd_card_cs4232_pnp(dev, card->private_data, pdev)) < 0) { + err = snd_card_cs423x_pnp(dev, card->private_data, pdev, cdev); + if (err < 0) { printk(KERN_ERR "PnP BIOS detection failed for " IDENT "\n"); snd_card_free(card); return err; @@ -614,35 +606,34 @@ static int __devinit snd_cs4232_pnpbios_detect(struct pnp_dev *pdev, return 0; } -static void __devexit snd_cs4232_pnp_remove(struct pnp_dev * pdev) +static void __devexit snd_cs423x_pnp_remove(struct pnp_dev *pdev) { snd_card_free(pnp_get_drvdata(pdev)); pnp_set_drvdata(pdev, NULL); } #ifdef CONFIG_PM -static int snd_cs4232_pnp_suspend(struct pnp_dev *pdev, pm_message_t state) +static int snd_cs423x_pnp_suspend(struct pnp_dev *pdev, pm_message_t state) { return snd_cs423x_suspend(pnp_get_drvdata(pdev)); } -static int snd_cs4232_pnp_resume(struct pnp_dev *pdev) +static int snd_cs423x_pnp_resume(struct pnp_dev *pdev) { return snd_cs423x_resume(pnp_get_drvdata(pdev)); } #endif -static struct pnp_driver cs4232_pnp_driver = { - .name = "cs4232-pnpbios", - .id_table = snd_cs4232_pnpbiosids, - .probe = snd_cs4232_pnpbios_detect, - .remove = __devexit_p(snd_cs4232_pnp_remove), +static struct pnp_driver cs423x_pnp_driver = { + .name = "cs423x-pnpbios", + .id_table = snd_cs423x_pnpbiosids, + .probe = snd_cs423x_pnpbios_detect, + .remove = __devexit_p(snd_cs423x_pnp_remove), #ifdef CONFIG_PM - .suspend = snd_cs4232_pnp_suspend, - .resume = snd_cs4232_pnp_resume, + .suspend = snd_cs423x_pnp_suspend, + .resume = snd_cs423x_pnp_resume, #endif }; -#endif /* CS4232 */ static int __devinit snd_cs423x_pnpc_detect(struct pnp_card_link *pcard, const struct pnp_card_device_id *pid) @@ -716,18 +707,14 @@ static int __init alsa_card_cs423x_init(void) #ifdef CONFIG_PNP if (!err) isa_registered = 1; -#ifdef CS4232 - err = pnp_register_driver(&cs4232_pnp_driver); + err = pnp_register_driver(&cs423x_pnp_driver); if (!err) pnp_registered = 1; -#endif err = pnp_register_card_driver(&cs423x_pnpc_driver); if (!err) pnpc_registered = 1; -#ifdef CS4232 if (pnp_registered) err = 0; -#endif if (isa_registered) err = 0; #endif @@ -739,10 +726,8 @@ static void __exit alsa_card_cs423x_exit(void) #ifdef CONFIG_PNP if (pnpc_registered) pnp_unregister_card_driver(&cs423x_pnpc_driver); -#ifdef CS4232 if (pnp_registered) - pnp_unregister_driver(&cs4232_pnp_driver); -#endif + pnp_unregister_driver(&cs423x_pnp_driver); if (isa_registered) #endif isa_unregister_driver(&cs423x_isa_driver); diff --git a/sound/isa/cs423x/cs4236_lib.c b/sound/isa/cs423x/cs4236_lib.c index 6a85fdc..38835f3 100644 --- a/sound/isa/cs423x/cs4236_lib.c +++ b/sound/isa/cs423x/cs4236_lib.c @@ -88,10 +88,6 @@ #include <sound/wss.h> #include <sound/asoundef.h> -MODULE_AUTHOR("Jaroslav Kysela <perex@perex.cz>"); -MODULE_DESCRIPTION("Routines for control of CS4235/4236B/4237B/4238B/4239 chips"); -MODULE_LICENSE("GPL"); - /* * */ @@ -286,7 +282,8 @@ int snd_cs4236_create(struct snd_card *card, if (hardware == WSS_HW_DETECT) hardware = WSS_HW_DETECT3; if (cport < 0x100) { - snd_printk("please, specify control port for CS4236+ chips\n"); + snd_printk(KERN_ERR "please, specify control port " + "for CS4236+ chips\n"); return -ENODEV; } err = snd_wss_create(card, port, cport, @@ -295,7 +292,8 @@ int snd_cs4236_create(struct snd_card *card, return err; if (!(chip->hardware & WSS_HW_CS4236B_MASK)) { - snd_printk("CS4236+: MODE3 and extended registers not available, hardware=0x%x\n",chip->hardware); + snd_printk(KERN_ERR "CS4236+: MODE3 and extended registers " + "not available, hardware=0x%x\n", chip->hardware); snd_device_free(card, chip); return -ENODEV; } @@ -303,16 +301,19 @@ int snd_cs4236_create(struct snd_card *card, { int idx; for (idx = 0; idx < 8; idx++) - snd_printk("CD%i = 0x%x\n", idx, inb(chip->cport + idx)); + snd_printk(KERN_DEBUG "CD%i = 0x%x\n", + idx, inb(chip->cport + idx)); for (idx = 0; idx < 9; idx++) - snd_printk("C%i = 0x%x\n", idx, snd_cs4236_ctrl_in(chip, idx)); + snd_printk(KERN_DEBUG "C%i = 0x%x\n", + idx, snd_cs4236_ctrl_in(chip, idx)); } #endif ver1 = snd_cs4236_ctrl_in(chip, 1); ver2 = snd_cs4236_ext_in(chip, CS4236_VERSION); snd_printdd("CS4236: [0x%lx] C1 (version) = 0x%x, ext = 0x%x\n", cport, ver1, ver2); if (ver1 != ver2) { - snd_printk("CS4236+ chip detected, but control port 0x%lx is not valid\n", cport); + snd_printk(KERN_ERR "CS4236+ chip detected, but " + "control port 0x%lx is not valid\n", cport); snd_device_free(card, chip); return -ENODEV; } @@ -883,7 +884,8 @@ static int snd_cs4236_get_iec958_switch(struct snd_kcontrol *kcontrol, struct sn spin_lock_irqsave(&chip->reg_lock, flags); ucontrol->value.integer.value[0] = chip->image[CS4231_ALT_FEATURE_1] & 0x02 ? 1 : 0; #if 0 - printk("get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", + printk(KERN_DEBUG "get valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, " + "C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", snd_wss_in(chip, CS4231_ALT_FEATURE_1), snd_cs4236_ctrl_in(chip, 3), snd_cs4236_ctrl_in(chip, 4), @@ -920,7 +922,8 @@ static int snd_cs4236_put_iec958_switch(struct snd_kcontrol *kcontrol, struct sn mutex_unlock(&chip->mce_mutex); #if 0 - printk("set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", + printk(KERN_DEBUG "set valid: ALT = 0x%x, C3 = 0x%x, C4 = 0x%x, " + "C5 = 0x%x, C6 = 0x%x, C8 = 0x%x\n", snd_wss_in(chip, CS4231_ALT_FEATURE_1), snd_cs4236_ctrl_in(chip, 3), snd_cs4236_ctrl_in(chip, 4), @@ -1015,23 +1018,3 @@ int snd_cs4236_mixer(struct snd_wss *chip) } return 0; } - -EXPORT_SYMBOL(snd_cs4236_create); -EXPORT_SYMBOL(snd_cs4236_pcm); -EXPORT_SYMBOL(snd_cs4236_mixer); - -/* - * INIT part - */ - -static int __init alsa_cs4236_init(void) -{ - return 0; -} - -static void __exit alsa_cs4236_exit(void) -{ -} - -module_init(alsa_cs4236_init) -module_exit(alsa_cs4236_exit) diff --git a/sound/isa/es1688/es1688.c b/sound/isa/es1688/es1688.c index d746750..442b081 100644 --- a/sound/isa/es1688/es1688.c +++ b/sound/isa/es1688/es1688.c @@ -49,6 +49,7 @@ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240,0x260 */ +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* Usually 0x388 */ static long mpu_port[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS - 1)] = -1}; static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5,7,9,10 */ @@ -65,6 +66,8 @@ MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); module_param_array(mpu_port, long, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); module_param_array(irq, int, NULL, 0444); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for ES1688 driver."); MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); module_param_array(mpu_irq, int, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); @@ -143,13 +146,19 @@ static int __devinit snd_es1688_probe(struct device *dev, unsigned int n) sprintf(card->longname, "%s at 0x%lx, irq %i, dma %i", pcm->name, chip->port, chip->irq, chip->dma8); - if (snd_opl3_create(card, chip->port, chip->port + 2, - OPL3_HW_OPL3, 0, &opl3) < 0) - dev_warn(dev, "opl3 not detected at 0x%lx\n", chip->port); - else { - error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); - if (error < 0) - goto out; + if (fm_port[n] == SNDRV_AUTO_PORT) + fm_port[n] = port[n]; /* share the same port */ + + if (fm_port[n] > 0) { + if (snd_opl3_create(card, fm_port[n], fm_port[n] + 2, + OPL3_HW_OPL3, 0, &opl3) < 0) + dev_warn(dev, + "opl3 not detected at 0x%lx\n", fm_port[n]); + else { + error = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (error < 0) + goto out; + } } if (mpu_irq[n] >= 0 && mpu_irq[n] != SNDRV_AUTO_IRQ && diff --git a/sound/isa/es1688/es1688_lib.c b/sound/isa/es1688/es1688_lib.c index 4fbb508..4c6e14f 100644 --- a/sound/isa/es1688/es1688_lib.c +++ b/sound/isa/es1688/es1688_lib.c @@ -45,7 +45,7 @@ static int snd_es1688_dsp_command(struct snd_es1688 *chip, unsigned char val) return 1; } #ifdef CONFIG_SND_DEBUG - printk("snd_es1688_dsp_command: timeout (0x%x)\n", val); + printk(KERN_DEBUG "snd_es1688_dsp_command: timeout (0x%x)\n", val); #endif return 0; } @@ -167,13 +167,16 @@ static int snd_es1688_probe(struct snd_es1688 *chip) hw = ES1688_HW_AUTO; switch (chip->version & 0xfff0) { case 0x4880: - snd_printk("[0x%lx] ESS: AudioDrive ES488 detected, but driver is in another place\n", chip->port); + snd_printk(KERN_ERR "[0x%lx] ESS: AudioDrive ES488 detected, " + "but driver is in another place\n", chip->port); return -ENODEV; case 0x6880: hw = (chip->version & 0x0f) >= 8 ? ES1688_HW_1688 : ES1688_HW_688; break; default: - snd_printk("[0x%lx] ESS: unknown AudioDrive chip with version 0x%x (Jazz16 soundcard?)\n", chip->port, chip->version); + snd_printk(KERN_ERR "[0x%lx] ESS: unknown AudioDrive chip " + "with version 0x%x (Jazz16 soundcard?)\n", + chip->port, chip->version); return -ENODEV; } @@ -223,7 +226,7 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) } } #if 0 - snd_printk("mpu cfg = 0x%x\n", cfg); + snd_printk(KERN_DEBUG "mpu cfg = 0x%x\n", cfg); #endif spin_lock_irqsave(&chip->reg_lock, flags); snd_es1688_mixer_write(chip, 0x40, cfg); @@ -237,7 +240,9 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) cfg = 0xf0; /* enable only DMA counter interrupt */ irq_bits = irqs[chip->irq & 0x0f]; if (irq_bits < 0) { - snd_printk("[0x%lx] ESS: bad IRQ %d for ES1688 chip!!\n", chip->port, chip->irq); + snd_printk(KERN_ERR "[0x%lx] ESS: bad IRQ %d " + "for ES1688 chip!!\n", + chip->port, chip->irq); #if 0 irq_bits = 0; cfg = 0x10; @@ -250,7 +255,8 @@ static int snd_es1688_init(struct snd_es1688 * chip, int enable) cfg = 0xf0; /* extended mode DMA enable */ dma = chip->dma8; if (dma > 3 || dma == 2) { - snd_printk("[0x%lx] ESS: bad DMA channel %d for ES1688 chip!!\n", chip->port, dma); + snd_printk(KERN_ERR "[0x%lx] ESS: bad DMA channel %d " + "for ES1688 chip!!\n", chip->port, dma); #if 0 dma_bits = 0; cfg = 0x00; /* disable all DMA */ @@ -341,8 +347,9 @@ static int snd_es1688_trigger(struct snd_es1688 *chip, int cmd, unsigned char va return -EINVAL; /* something is wrong */ } #if 0 - printk("trigger: val = 0x%x, value = 0x%x\n", val, value); - printk("trigger: pointer = 0x%x\n", snd_dma_pointer(chip->dma8, chip->dma_size)); + printk(KERN_DEBUG "trigger: val = 0x%x, value = 0x%x\n", val, value); + printk(KERN_DEBUG "trigger: pointer = 0x%x\n", + snd_dma_pointer(chip->dma8, chip->dma_size)); #endif snd_es1688_write(chip, 0xb8, (val & 0xf0) | value); spin_unlock(&chip->reg_lock); diff --git a/sound/isa/gus/gus_dma.c b/sound/isa/gus/gus_dma.c index f45f611..36c27c8 100644 --- a/sound/isa/gus/gus_dma.c +++ b/sound/isa/gus/gus_dma.c @@ -45,7 +45,8 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus, unsigned char dma_cmd; unsigned int address_high; - // snd_printk("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", addr, (long) buf, count); + snd_printdd("dma_transfer: addr=0x%x, buf=0x%lx, count=0x%x\n", + addr, buf_addr, count); if (gus->gf1.dma1 > 3) { if (gus->gf1.enh_mode) { @@ -77,7 +78,8 @@ static void snd_gf1_dma_program(struct snd_gus_card * gus, snd_gf1_dma_ack(gus); snd_dma_program(gus->gf1.dma1, buf_addr, count, dma_cmd & SNDRV_GF1_DMA_READ ? DMA_MODE_READ : DMA_MODE_WRITE); #if 0 - snd_printk("address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n", address << 1, count, dma_cmd); + snd_printk(KERN_DEBUG "address = 0x%x, count = 0x%x, dma_cmd = 0x%x\n", + address << 1, count, dma_cmd); #endif spin_lock_irqsave(&gus->reg_lock, flags); if (gus->gf1.enh_mode) { @@ -142,7 +144,9 @@ static void snd_gf1_dma_interrupt(struct snd_gus_card * gus) snd_gf1_dma_program(gus, block->addr, block->buf_addr, block->count, (unsigned short) block->cmd); kfree(block); #if 0 - printk("program dma (IRQ) - addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", addr, (long) buffer, count, cmd); + snd_printd(KERN_DEBUG "program dma (IRQ) - " + "addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", + block->addr, block->buf_addr, block->count, block->cmd); #endif } @@ -203,13 +207,16 @@ int snd_gf1_dma_transfer_block(struct snd_gus_card * gus, } *block = *__block; block->next = NULL; -#if 0 - printk("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", block->addr, (long) block->buffer, block->count, block->cmd); -#endif -#if 0 - printk("gus->gf1.dma_data_pcm_last = 0x%lx\n", (long)gus->gf1.dma_data_pcm_last); - printk("gus->gf1.dma_data_pcm = 0x%lx\n", (long)gus->gf1.dma_data_pcm); -#endif + + snd_printdd("addr = 0x%x, buffer = 0x%lx, count = 0x%x, cmd = 0x%x\n", + block->addr, (long) block->buffer, block->count, + block->cmd); + + snd_printdd("gus->gf1.dma_data_pcm_last = 0x%lx\n", + (long)gus->gf1.dma_data_pcm_last); + snd_printdd("gus->gf1.dma_data_pcm = 0x%lx\n", + (long)gus->gf1.dma_data_pcm); + spin_lock_irqsave(&gus->dma_lock, flags); if (synth) { if (gus->gf1.dma_data_synth_last) { diff --git a/sound/isa/gus/gus_irq.c b/sound/isa/gus/gus_irq.c index 041894d..2055aff 100644 --- a/sound/isa/gus/gus_irq.c +++ b/sound/isa/gus/gus_irq.c @@ -41,7 +41,7 @@ __again: if (status == 0) return IRQ_RETVAL(handled); handled = 1; - // snd_printk("IRQ: status = 0x%x\n", status); + /* snd_printk(KERN_DEBUG "IRQ: status = 0x%x\n", status); */ if (status & 0x02) { STAT_ADD(gus->gf1.interrupt_stat_midi_in); if (gus->gf1.interrupt_handler_midi_in) @@ -65,7 +65,9 @@ __again: continue; /* multi request */ already |= _current_; /* mark request */ #if 0 - printk("voice = %i, voice_status = 0x%x, voice_verify = %i\n", voice, voice_status, inb(GUSP(gus, GF1PAGE))); + printk(KERN_DEBUG "voice = %i, voice_status = 0x%x, " + "voice_verify = %i\n", + voice, voice_status, inb(GUSP(gus, GF1PAGE))); #endif pvoice = &gus->gf1.voices[voice]; if (pvoice->use) { diff --git a/sound/isa/gus/gus_pcm.c b/sound/isa/gus/gus_pcm.c index 38510ae..edb11ee 100644 --- a/sound/isa/gus/gus_pcm.c +++ b/sound/isa/gus/gus_pcm.c @@ -82,7 +82,10 @@ static int snd_gf1_pcm_block_change(struct snd_pcm_substream *substream, count += offset & 31; offset &= ~31; - // snd_printk("block change - offset = 0x%x, count = 0x%x\n", offset, count); + /* + snd_printk(KERN_DEBUG "block change - offset = 0x%x, count = 0x%x\n", + offset, count); + */ memset(&block, 0, sizeof(block)); block.cmd = SNDRV_GF1_DMA_IRQ; if (snd_pcm_format_unsigned(runtime->format)) @@ -135,7 +138,11 @@ static void snd_gf1_pcm_trigger_up(struct snd_pcm_substream *substream) curr = begin + (pcmp->bpos * pcmp->block_size) / runtime->channels; end = curr + (pcmp->block_size / runtime->channels); end -= snd_pcm_format_width(runtime->format) == 16 ? 2 : 1; - // snd_printk("init: curr=0x%x, begin=0x%x, end=0x%x, ctrl=0x%x, ramp=0x%x, rate=0x%x\n", curr, begin, end, voice_ctrl, ramp_ctrl, rate); + /* + snd_printk(KERN_DEBUG "init: curr=0x%x, begin=0x%x, end=0x%x, " + "ctrl=0x%x, ramp=0x%x, rate=0x%x\n", + curr, begin, end, voice_ctrl, ramp_ctrl, rate); + */ pan = runtime->channels == 2 ? (!voice ? 1 : 14) : 8; vol = !voice ? gus->gf1.pcm_volume_level_left : gus->gf1.pcm_volume_level_right; spin_lock_irqsave(&gus->reg_lock, flags); @@ -205,9 +212,11 @@ static void snd_gf1_pcm_interrupt_wave(struct snd_gus_card * gus, ramp_ctrl = (snd_gf1_read8(gus, SNDRV_GF1_VB_VOLUME_CONTROL) & ~0xa4) | 0x03; #if 0 snd_gf1_select_voice(gus, pvoice->number); - printk("position = 0x%x\n", (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); + printk(KERN_DEBUG "position = 0x%x\n", + (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); snd_gf1_select_voice(gus, pcmp->pvoices[1]->number); - printk("position = 0x%x\n", (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); + printk(KERN_DEBUG "position = 0x%x\n", + (snd_gf1_read_addr(gus, SNDRV_GF1_VA_CURRENT, voice_ctrl & 4) >> 4)); snd_gf1_select_voice(gus, pvoice->number); #endif pcmp->bpos++; @@ -299,7 +308,11 @@ static int snd_gf1_pcm_poke_block(struct snd_gus_card *gus, unsigned char *buf, unsigned int len; unsigned long flags; - // printk("poke block; buf = 0x%x, pos = %i, count = %i, port = 0x%x\n", (int)buf, pos, count, gus->gf1.port); + /* + printk(KERN_DEBUG + "poke block; buf = 0x%x, pos = %i, count = %i, port = 0x%x\n", + (int)buf, pos, count, gus->gf1.port); + */ while (count > 0) { len = count; if (len > 512) /* limit, to allow IRQ */ @@ -680,7 +693,8 @@ static int snd_gf1_pcm_playback_open(struct snd_pcm_substream *substream) runtime->private_free = snd_gf1_pcm_playback_free; #if 0 - printk("playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n", (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer); + printk(KERN_DEBUG "playback.buffer = 0x%lx, gf1.pcm_buffer = 0x%lx\n", + (long) pcm->playback.buffer, (long) gus->gf1.pcm_buffer); #endif if ((err = snd_gf1_dma_init(gus)) < 0) return err; diff --git a/sound/isa/gus/gus_uart.c b/sound/isa/gus/gus_uart.c index f0af3f7..21cc42e 100644 --- a/sound/isa/gus/gus_uart.c +++ b/sound/isa/gus/gus_uart.c @@ -129,8 +129,14 @@ static int snd_gf1_uart_input_open(struct snd_rawmidi_substream *substream) } spin_unlock_irqrestore(&gus->uart_cmd_lock, flags); #if 0 - snd_printk("read init - enable = %i, cmd = 0x%x, stat = 0x%x\n", gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus)); - snd_printk("[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x (page = 0x%x)\n", gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100), inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102)); + snd_printk(KERN_DEBUG + "read init - enable = %i, cmd = 0x%x, stat = 0x%x\n", + gus->uart_enable, gus->gf1.uart_cmd, snd_gf1_uart_stat(gus)); + snd_printk(KERN_DEBUG + "[0x%x] reg (ctrl/status) = 0x%x, reg (data) = 0x%x " + "(page = 0x%x)\n", + gus->gf1.port + 0x100, inb(gus->gf1.port + 0x100), + inb(gus->gf1.port + 0x101), inb(gus->gf1.port + 0x102)); #endif return 0; } diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 50e429a..534a6ec 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -170,7 +170,7 @@ static void snd_interwave_i2c_setlines(struct snd_i2c_bus *bus, int ctrl, int da unsigned long port = bus->private_value; #if 0 - printk("i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data); + printk(KERN_DEBUG "i2c_setlines - 0x%lx <- %i,%i\n", port, ctrl, data); #endif outb((data << 1) | ctrl, port); udelay(10); @@ -183,7 +183,7 @@ static int snd_interwave_i2c_getclockline(struct snd_i2c_bus *bus) res = inb(port) & 1; #if 0 - printk("i2c_getclockline - 0x%lx -> %i\n", port, res); + printk(KERN_DEBUG "i2c_getclockline - 0x%lx -> %i\n", port, res); #endif return res; } @@ -197,7 +197,7 @@ static int snd_interwave_i2c_getdataline(struct snd_i2c_bus *bus, int ack) udelay(10); res = (inb(port) & 2) >> 1; #if 0 - printk("i2c_getdataline - 0x%lx -> %i\n", port, res); + printk(KERN_DEBUG "i2c_getdataline - 0x%lx -> %i\n", port, res); #endif return res; } @@ -342,7 +342,8 @@ static void __devinit snd_interwave_bank_sizes(struct snd_gus_card * gus, int *s snd_gf1_poke(gus, local, d); snd_gf1_poke(gus, local + 1, d + 1); #if 0 - printk("d = 0x%x, local = 0x%x, local + 1 = 0x%x, idx << 22 = 0x%x\n", + printk(KERN_DEBUG "d = 0x%x, local = 0x%x, " + "local + 1 = 0x%x, idx << 22 = 0x%x\n", d, snd_gf1_peek(gus, local), snd_gf1_peek(gus, local + 1), @@ -356,7 +357,8 @@ static void __devinit snd_interwave_bank_sizes(struct snd_gus_card * gus, int *s } } #if 0 - printk("sizes: %i %i %i %i\n", sizes[0], sizes[1], sizes[2], sizes[3]); + printk(KERN_DEBUG "sizes: %i %i %i %i\n", + sizes[0], sizes[1], sizes[2], sizes[3]); #endif } @@ -410,12 +412,12 @@ static void __devinit snd_interwave_detect_memory(struct snd_gus_card * gus) lmct = (psizes[3] << 24) | (psizes[2] << 16) | (psizes[1] << 8) | psizes[0]; #if 0 - printk("lmct = 0x%08x\n", lmct); + printk(KERN_DEBUG "lmct = 0x%08x\n", lmct); #endif for (i = 0; i < ARRAY_SIZE(lmc); i++) if (lmct == lmc[i]) { #if 0 - printk("found !!! %i\n", i); + printk(KERN_DEBUG "found !!! %i\n", i); #endif snd_gf1_write16(gus, SNDRV_GF1_GW_MEMORY_CONFIG, (snd_gf1_look16(gus, SNDRV_GF1_GW_MEMORY_CONFIG) & 0xfff0) | i); snd_interwave_bank_sizes(gus, psizes); diff --git a/sound/isa/msnd/Makefile b/sound/isa/msnd/Makefile new file mode 100644 index 0000000..2171c0a --- /dev/null +++ b/sound/isa/msnd/Makefile @@ -0,0 +1,9 @@ + +snd-msnd-lib-objs := msnd.o msnd_midi.o msnd_pinnacle_mixer.o +snd-msnd-pinnacle-objs := msnd_pinnacle.o +snd-msnd-classic-objs := msnd_classic.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_MSND_PINNACLE) += snd-msnd-pinnacle.o snd-msnd-lib.o +obj-$(CONFIG_SND_MSND_CLASSIC) += snd-msnd-classic.o snd-msnd-lib.o + diff --git a/sound/isa/msnd/msnd.c b/sound/isa/msnd/msnd.c new file mode 100644 index 0000000..9064544 --- /dev/null +++ b/sound/isa/msnd/msnd.c @@ -0,0 +1,705 @@ +/********************************************************************* + * + * 2002/06/30 Karsten Wiese: + * removed kernel-version dependencies. + * ripped from linux kernel 2.4.18 (OSS Implementation) by me. + * In the OSS Version, this file is compiled to a separate MODULE, + * that is used by the pinnacle and the classic driver. + * since there is no classic driver for alsa yet (i dont have a classic + * & writing one blindfold is difficult) this file's object is statically + * linked into the pinnacle-driver-module for now. look for the string + * "uncomment this to make this a module again" + * to do guess what. + * + * the following is a copy of the 2.4.18 OSS FREE file-heading comment: + * + * msnd.c - Driver Base + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Copyright (C) 1998 Andrew Veliath + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include <linux/kernel.h> +#include <linux/types.h> +#include <linux/interrupt.h> +#include <linux/io.h> +#include <linux/fs.h> +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "msnd.h" + +#define LOGNAME "msnd" + + +void snd_msnd_init_queue(void *base, int start, int size) +{ + writew(PCTODSP_BASED(start), base + JQS_wStart); + writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize); + writew(0, base + JQS_wHead); + writew(0, base + JQS_wTail); +} +EXPORT_SYMBOL(snd_msnd_init_queue); + +static int snd_msnd_wait_TXDE(struct snd_msnd *dev) +{ + unsigned int io = dev->io; + int timeout = 1000; + + while (timeout-- > 0) + if (inb(io + HP_ISR) & HPISR_TXDE) + return 0; + + return -EIO; +} + +static int snd_msnd_wait_HC0(struct snd_msnd *dev) +{ + unsigned int io = dev->io; + int timeout = 1000; + + while (timeout-- > 0) + if (!(inb(io + HP_CVR) & HPCVR_HC)) + return 0; + + return -EIO; +} + +int snd_msnd_send_dsp_cmd(struct snd_msnd *dev, u8 cmd) +{ + unsigned long flags; + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_HC0(dev) == 0) { + outb(cmd, dev->io + HP_CVR); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(KERN_ERR LOGNAME ": Send DSP command timeout\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_send_dsp_cmd); + +int snd_msnd_send_word(struct snd_msnd *dev, unsigned char high, + unsigned char mid, unsigned char low) +{ + unsigned int io = dev->io; + + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(high, io + HP_TXH); + outb(mid, io + HP_TXM); + outb(low, io + HP_TXL); + return 0; + } + + snd_printd(KERN_ERR LOGNAME ": Send host word timeout\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_send_word); + +int snd_msnd_upload_host(struct snd_msnd *dev, const u8 *bin, int len) +{ + int i; + + if (len % 3 != 0) { + snd_printk(KERN_ERR LOGNAME + ": Upload host data not multiple of 3!\n"); + return -EINVAL; + } + + for (i = 0; i < len; i += 3) + if (snd_msnd_send_word(dev, bin[i], bin[i + 1], bin[i + 2])) + return -EIO; + + inb(dev->io + HP_RXL); + inb(dev->io + HP_CVR); + + return 0; +} +EXPORT_SYMBOL(snd_msnd_upload_host); + +int snd_msnd_enable_irq(struct snd_msnd *dev) +{ + unsigned long flags; + + if (dev->irq_ref++) + return 0; + + snd_printdd(LOGNAME ": Enabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(inb(dev->io + HP_ICR) | HPICR_TREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + outb(dev->irqid, dev->io + HP_IRQM); + + outb(inb(dev->io + HP_ICR) & ~HPICR_TREQ, dev->io + HP_ICR); + outb(inb(dev->io + HP_ICR) | HPICR_RREQ, dev->io + HP_ICR); + enable_irq(dev->irq); + snd_msnd_init_queue(dev->DSPQ, dev->dspq_data_buff, + dev->dspq_buff_size); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(KERN_ERR LOGNAME ": Enable IRQ failed\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_enable_irq); + +int snd_msnd_disable_irq(struct snd_msnd *dev) +{ + unsigned long flags; + + if (--dev->irq_ref > 0) + return 0; + + if (dev->irq_ref < 0) + snd_printd(KERN_WARNING LOGNAME ": IRQ ref count is %d\n", + dev->irq_ref); + + snd_printdd(LOGNAME ": Disabling IRQ\n"); + + spin_lock_irqsave(&dev->lock, flags); + if (snd_msnd_wait_TXDE(dev) == 0) { + outb(inb(dev->io + HP_ICR) & ~HPICR_RREQ, dev->io + HP_ICR); + if (dev->type == msndClassic) + outb(HPIRQ_NONE, dev->io + HP_IRQM); + disable_irq(dev->irq); + spin_unlock_irqrestore(&dev->lock, flags); + return 0; + } + spin_unlock_irqrestore(&dev->lock, flags); + + snd_printd(KERN_ERR LOGNAME ": Disable IRQ failed\n"); + + return -EIO; +} +EXPORT_SYMBOL(snd_msnd_disable_irq); + +static inline long get_play_delay_jiffies(struct snd_msnd *chip, long size) +{ + long tmp = (size * HZ * chip->play_sample_size) / 8; + return tmp / (chip->play_sample_rate * chip->play_channels); +} + +static void snd_msnd_dsp_write_flush(struct snd_msnd *chip) +{ + if (!(chip->mode & FMODE_WRITE) || !test_bit(F_WRITING, &chip->flags)) + return; + set_bit(F_WRITEFLUSH, &chip->flags); +/* interruptible_sleep_on_timeout( + &chip->writeflush, + get_play_delay_jiffies(&chip, chip->DAPF.len));*/ + clear_bit(F_WRITEFLUSH, &chip->flags); + if (!signal_pending(current)) + schedule_timeout_interruptible( + get_play_delay_jiffies(chip, chip->play_period_bytes)); + clear_bit(F_WRITING, &chip->flags); +} + +void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file) +{ + if ((file ? file->f_mode : chip->mode) & FMODE_READ) { + clear_bit(F_READING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP); + snd_msnd_disable_irq(chip); + if (file) { + snd_printd(KERN_INFO LOGNAME + ": Stopping read for %p\n", file); + chip->mode &= ~FMODE_READ; + } + clear_bit(F_AUDIO_READ_INUSE, &chip->flags); + } + if ((file ? file->f_mode : chip->mode) & FMODE_WRITE) { + if (test_bit(F_WRITING, &chip->flags)) { + snd_msnd_dsp_write_flush(chip); + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP); + } + snd_msnd_disable_irq(chip); + if (file) { + snd_printd(KERN_INFO + LOGNAME ": Stopping write for %p\n", file); + chip->mode &= ~FMODE_WRITE; + } + clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + } +} +EXPORT_SYMBOL(snd_msnd_dsp_halt); + + +int snd_msnd_DARQ(struct snd_msnd *chip, int bank) +{ + int /*size, n,*/ timeout = 3; + u16 wTmp; + /* void *DAQD; */ + + /* Increment the tail and check for queue wrap */ + wTmp = readw(chip->DARQ + JQS_wTail) + PCTODSP_OFFSET(DAQDS__size); + if (wTmp > readw(chip->DARQ + JQS_wSize)) + wTmp = 0; + while (wTmp == readw(chip->DARQ + JQS_wHead) && timeout--) + udelay(1); + + if (chip->capturePeriods == 2) { + void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF + + bank * DAQDS__size + DAQDS_wStart; + unsigned short offset = 0x3000 + chip->capturePeriodBytes; + + if (readw(pDAQ) != PCTODSP_BASED(0x3000)) + offset = 0x3000; + writew(PCTODSP_BASED(offset), pDAQ); + } + + writew(wTmp, chip->DARQ + JQS_wTail); + +#if 0 + /* Get our digital audio queue struct */ + DAQD = bank * DAQDS__size + chip->mappedbase + DARQ_DATA_BUFF; + + /* Get length of data */ + size = readw(DAQD + DAQDS_wSize); + + /* Read data from the head (unprotected bank 1 access okay + since this is only called inside an interrupt) */ + outb(HPBLKSEL_1, chip->io + HP_BLKS); + n = msnd_fifo_write(&chip->DARF, + (char *)(chip->base + bank * DAR_BUFF_SIZE), + size, 0); + if (n <= 0) { + outb(HPBLKSEL_0, chip->io + HP_BLKS); + return n; + } + outb(HPBLKSEL_0, chip->io + HP_BLKS); +#endif + + return 1; +} +EXPORT_SYMBOL(snd_msnd_DARQ); + +int snd_msnd_DAPQ(struct snd_msnd *chip, int start) +{ + u16 DAPQ_tail; + int protect = start, nbanks = 0; + void *DAQD; + static int play_banks_submitted; + /* unsigned long flags; + spin_lock_irqsave(&chip->lock, flags); not necessary */ + + DAPQ_tail = readw(chip->DAPQ + JQS_wTail); + while (DAPQ_tail != readw(chip->DAPQ + JQS_wHead) || start) { + int bank_num = DAPQ_tail / PCTODSP_OFFSET(DAQDS__size); + + if (start) { + start = 0; + play_banks_submitted = 0; + } + + /* Get our digital audio queue struct */ + DAQD = bank_num * DAQDS__size + chip->mappedbase + + DAPQ_DATA_BUFF; + + /* Write size of this bank */ + writew(chip->play_period_bytes, DAQD + DAQDS_wSize); + if (play_banks_submitted < 3) + ++play_banks_submitted; + else if (chip->playPeriods == 2) { + unsigned short offset = chip->play_period_bytes; + + if (readw(DAQD + DAQDS_wStart) != PCTODSP_BASED(0x0)) + offset = 0; + + writew(PCTODSP_BASED(offset), DAQD + DAQDS_wStart); + } + ++nbanks; + + /* Then advance the tail */ + /* + if (protect) + snd_printd(KERN_INFO "B %X %lX\n", + bank_num, xtime.tv_usec); + */ + + DAPQ_tail = (++bank_num % 3) * PCTODSP_OFFSET(DAQDS__size); + writew(DAPQ_tail, chip->DAPQ + JQS_wTail); + /* Tell the DSP to play the bank */ + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_START); + if (protect) + if (2 == bank_num) + break; + } + /* + if (protect) + snd_printd(KERN_INFO "%lX\n", xtime.tv_usec); + */ + /* spin_unlock_irqrestore(&chip->lock, flags); not necessary */ + return nbanks; +} +EXPORT_SYMBOL(snd_msnd_DAPQ); + +static void snd_msnd_play_reset_queue(struct snd_msnd *chip, + unsigned int pcm_periods, + unsigned int pcm_count) +{ + int n; + void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + + chip->last_playbank = -1; + chip->playLimit = pcm_count * (pcm_periods - 1); + chip->playPeriods = pcm_periods; + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wHead); + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DAPQ + JQS_wTail); + + chip->play_period_bytes = pcm_count; + + for (n = 0; n < pcm_periods; ++n, pDAQ += DAQDS__size) { + writew(PCTODSP_BASED((u32)(pcm_count * n)), + pDAQ + DAQDS_wStart); + writew(0, pDAQ + DAQDS_wSize); + writew(1, pDAQ + DAQDS_wFormat); + writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->play_channels, pDAQ + DAQDS_wChannels); + writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate); + writew(HIMT_PLAY_DONE * 0x100 + n, pDAQ + DAQDS_wIntMsg); + writew(n, pDAQ + DAQDS_wFlags); + } +} + +static void snd_msnd_capture_reset_queue(struct snd_msnd *chip, + unsigned int pcm_periods, + unsigned int pcm_count) +{ + int n; + void *pDAQ; + /* unsigned long flags; */ + + /* snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); */ + + chip->last_recbank = 2; + chip->captureLimit = pcm_count * (pcm_periods - 1); + chip->capturePeriods = pcm_periods; + writew(PCTODSP_OFFSET(0 * DAQDS__size), chip->DARQ + JQS_wHead); + writew(PCTODSP_OFFSET(chip->last_recbank * DAQDS__size), + chip->DARQ + JQS_wTail); + +#if 0 /* Critical section: bank 1 access. this is how the OSS driver does it:*/ + spin_lock_irqsave(&chip->lock, flags); + outb(HPBLKSEL_1, chip->io + HP_BLKS); + memset_io(chip->mappedbase, 0, DAR_BUFF_SIZE * 3); + outb(HPBLKSEL_0, chip->io + HP_BLKS); + spin_unlock_irqrestore(&chip->lock, flags); +#endif + + chip->capturePeriodBytes = pcm_count; + snd_printdd("snd_msnd_capture_reset_queue() %i\n", pcm_count); + + pDAQ = chip->mappedbase + DARQ_DATA_BUFF; + + for (n = 0; n < pcm_periods; ++n, pDAQ += DAQDS__size) { + u32 tmp = pcm_count * n; + + writew(PCTODSP_BASED(tmp + 0x3000), pDAQ + DAQDS_wStart); + writew(pcm_count, pDAQ + DAQDS_wSize); + writew(1, pDAQ + DAQDS_wFormat); + writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->capture_channels, pDAQ + DAQDS_wChannels); + writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate); + writew(HIMT_RECORD_DONE * 0x100 + n, pDAQ + DAQDS_wIntMsg); + writew(n, pDAQ + DAQDS_wFlags); + } +} + +static struct snd_pcm_hardware snd_msnd_playback = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3000, + .period_bytes_min = 0x40, + .period_bytes_max = 0x1800, + .periods_min = 2, + .periods_max = 3, + .fifo_size = 0, +}; + +static struct snd_pcm_hardware snd_msnd_capture = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID, + .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3000, + .period_bytes_min = 0x40, + .period_bytes_max = 0x1800, + .periods_min = 2, + .periods_max = 3, + .fifo_size = 0, +}; + + +static int snd_msnd_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + set_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + clear_bit(F_WRITING, &chip->flags); + snd_msnd_enable_irq(chip); + + runtime->dma_area = chip->mappedbase; + runtime->dma_bytes = 0x3000; + + chip->playback_substream = substream; + runtime->hw = snd_msnd_playback; + return 0; +} + +static int snd_msnd_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + snd_msnd_disable_irq(chip); + clear_bit(F_AUDIO_WRITE_INUSE, &chip->flags); + return 0; +} + + +static int snd_msnd_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int i; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + void *pDAQ = chip->mappedbase + DAPQ_DATA_BUFF; + + chip->play_sample_size = snd_pcm_format_width(params_format(params)); + chip->play_channels = params_channels(params); + chip->play_sample_rate = params_rate(params); + + for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) { + writew(chip->play_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->play_channels, pDAQ + DAQDS_wChannels); + writew(chip->play_sample_rate, pDAQ + DAQDS_wSampleRate); + } + /* dont do this here: + * snd_msnd_calibrate_adc(chip->play_sample_rate); + */ + + return 0; +} + +static int snd_msnd_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream); + unsigned int pcm_count = snd_pcm_lib_period_bytes(substream); + unsigned int pcm_periods = pcm_size / pcm_count; + + snd_msnd_play_reset_queue(chip, pcm_periods, pcm_count); + chip->playDMAPos = 0; + return 0; +} + +static int snd_msnd_playback_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + int result = 0; + + if (cmd == SNDRV_PCM_TRIGGER_START) { + snd_printdd("snd_msnd_playback_trigger(START)\n"); + chip->banksPlayed = 0; + set_bit(F_WRITING, &chip->flags); + snd_msnd_DAPQ(chip, 1); + } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + snd_printdd("snd_msnd_playback_trigger(STop)\n"); + /* interrupt diagnostic, comment this out later */ + clear_bit(F_WRITING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_PLAY_STOP); + } else { + snd_printd(KERN_ERR "snd_msnd_playback_trigger(?????)\n"); + result = -EINVAL; + } + + snd_printdd("snd_msnd_playback_trigger() ENDE\n"); + return result; +} + +static snd_pcm_uframes_t +snd_msnd_playback_pointer(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + return bytes_to_frames(substream->runtime, chip->playDMAPos); +} + + +static struct snd_pcm_ops snd_msnd_playback_ops = { + .open = snd_msnd_playback_open, + .close = snd_msnd_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_msnd_playback_hw_params, + .prepare = snd_msnd_playback_prepare, + .trigger = snd_msnd_playback_trigger, + .pointer = snd_msnd_playback_pointer, +}; + +static int snd_msnd_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + set_bit(F_AUDIO_READ_INUSE, &chip->flags); + snd_msnd_enable_irq(chip); + runtime->dma_area = chip->mappedbase + 0x3000; + runtime->dma_bytes = 0x3000; + memset(runtime->dma_area, 0, runtime->dma_bytes); + chip->capture_substream = substream; + runtime->hw = snd_msnd_capture; + return 0; +} + +static int snd_msnd_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + snd_msnd_disable_irq(chip); + clear_bit(F_AUDIO_READ_INUSE, &chip->flags); + return 0; +} + +static int snd_msnd_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + unsigned int pcm_size = snd_pcm_lib_buffer_bytes(substream); + unsigned int pcm_count = snd_pcm_lib_period_bytes(substream); + unsigned int pcm_periods = pcm_size / pcm_count; + + snd_msnd_capture_reset_queue(chip, pcm_periods, pcm_count); + chip->captureDMAPos = 0; + return 0; +} + +static int snd_msnd_capture_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + if (cmd == SNDRV_PCM_TRIGGER_START) { + chip->last_recbank = -1; + set_bit(F_READING, &chip->flags); + if (snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_START) == 0) + return 0; + + clear_bit(F_READING, &chip->flags); + } else if (cmd == SNDRV_PCM_TRIGGER_STOP) { + clear_bit(F_READING, &chip->flags); + snd_msnd_send_dsp_cmd(chip, HDEX_RECORD_STOP); + return 0; + } + return -EINVAL; +} + + +static snd_pcm_uframes_t +snd_msnd_capture_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + + return bytes_to_frames(runtime, chip->captureDMAPos); +} + + +static int snd_msnd_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int i; + struct snd_msnd *chip = snd_pcm_substream_chip(substream); + void *pDAQ = chip->mappedbase + DARQ_DATA_BUFF; + + chip->capture_sample_size = snd_pcm_format_width(params_format(params)); + chip->capture_channels = params_channels(params); + chip->capture_sample_rate = params_rate(params); + + for (i = 0; i < 3; ++i, pDAQ += DAQDS__size) { + writew(chip->capture_sample_size, pDAQ + DAQDS_wSampleSize); + writew(chip->capture_channels, pDAQ + DAQDS_wChannels); + writew(chip->capture_sample_rate, pDAQ + DAQDS_wSampleRate); + } + return 0; +} + + +static struct snd_pcm_ops snd_msnd_capture_ops = { + .open = snd_msnd_capture_open, + .close = snd_msnd_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_msnd_capture_hw_params, + .prepare = snd_msnd_capture_prepare, + .trigger = snd_msnd_capture_trigger, + .pointer = snd_msnd_capture_pointer, +}; + + +int snd_msnd_pcm(struct snd_card *card, int device, + struct snd_pcm **rpcm) +{ + struct snd_msnd *chip = card->private_data; + struct snd_pcm *pcm; + int err; + + err = snd_pcm_new(card, "MSNDPINNACLE", device, 1, 1, &pcm); + if (err < 0) + return err; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_msnd_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_msnd_capture_ops); + + pcm->private_data = chip; + strcpy(pcm->name, "Hurricane"); + + + if (rpcm) + *rpcm = pcm; + return 0; +} +EXPORT_SYMBOL(snd_msnd_pcm); + +MODULE_DESCRIPTION("Common routines for Turtle Beach Multisound drivers"); +MODULE_LICENSE("GPL"); + diff --git a/sound/isa/msnd/msnd.h b/sound/isa/msnd/msnd.h new file mode 100644 index 0000000..3773e24 --- /dev/null +++ b/sound/isa/msnd/msnd.h @@ -0,0 +1,308 @@ +/********************************************************************* + * + * msnd.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_H +#define __MSND_H + +#define DEFSAMPLERATE 44100 +#define DEFSAMPLESIZE SNDRV_PCM_FORMAT_S16 +#define DEFCHANNELS 1 + +#define SRAM_BANK_SIZE 0x8000 +#define SRAM_CNTL_START 0x7F00 +#define SMA_STRUCT_START 0x7F40 + +#define DSP_BASE_ADDR 0x4000 +#define DSP_BANK_BASE 0x4000 + +#define AGND 0x01 +#define SIGNAL 0x02 + +#define EXT_DSP_BIT_DCAL 0x0001 +#define EXT_DSP_BIT_MIDI_CON 0x0002 + +#define BUFFSIZE 0x8000 +#define HOSTQ_SIZE 0x40 + +#define DAP_BUFF_SIZE 0x2400 + +#define DAPQ_STRUCT_SIZE 0x10 +#define DARQ_STRUCT_SIZE 0x10 +#define DAPQ_BUFF_SIZE (3 * 0x10) +#define DARQ_BUFF_SIZE (3 * 0x10) +#define MODQ_BUFF_SIZE 0x400 + +#define DAPQ_DATA_BUFF 0x6C00 +#define DARQ_DATA_BUFF 0x6C30 +#define MODQ_DATA_BUFF 0x6C60 +#define MIDQ_DATA_BUFF 0x7060 + +#define DAPQ_OFFSET SRAM_CNTL_START +#define DARQ_OFFSET (SRAM_CNTL_START + 0x08) +#define MODQ_OFFSET (SRAM_CNTL_START + 0x10) +#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18) +#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20) + +#define HP_ICR 0x00 +#define HP_CVR 0x01 +#define HP_ISR 0x02 +#define HP_IVR 0x03 +#define HP_NU 0x04 +#define HP_INFO 0x04 +#define HP_TXH 0x05 +#define HP_RXH 0x05 +#define HP_TXM 0x06 +#define HP_RXM 0x06 +#define HP_TXL 0x07 +#define HP_RXL 0x07 + +#define HP_ICR_DEF 0x00 +#define HP_CVR_DEF 0x12 +#define HP_ISR_DEF 0x06 +#define HP_IVR_DEF 0x0f +#define HP_NU_DEF 0x00 + +#define HP_IRQM 0x09 + +#define HPR_BLRC 0x08 +#define HPR_SPR1 0x09 +#define HPR_SPR2 0x0A +#define HPR_TCL0 0x0B +#define HPR_TCL1 0x0C +#define HPR_TCL2 0x0D +#define HPR_TCL3 0x0E +#define HPR_TCL4 0x0F + +#define HPICR_INIT 0x80 +#define HPICR_HM1 0x40 +#define HPICR_HM0 0x20 +#define HPICR_HF1 0x10 +#define HPICR_HF0 0x08 +#define HPICR_TREQ 0x02 +#define HPICR_RREQ 0x01 + +#define HPCVR_HC 0x80 + +#define HPISR_HREQ 0x80 +#define HPISR_DMA 0x40 +#define HPISR_HF3 0x10 +#define HPISR_HF2 0x08 +#define HPISR_TRDY 0x04 +#define HPISR_TXDE 0x02 +#define HPISR_RXDF 0x01 + +#define HPIO_290 0 +#define HPIO_260 1 +#define HPIO_250 2 +#define HPIO_240 3 +#define HPIO_230 4 +#define HPIO_220 5 +#define HPIO_210 6 +#define HPIO_3E0 7 + +#define HPMEM_NONE 0 +#define HPMEM_B000 1 +#define HPMEM_C800 2 +#define HPMEM_D000 3 +#define HPMEM_D400 4 +#define HPMEM_D800 5 +#define HPMEM_E000 6 +#define HPMEM_E800 7 + +#define HPIRQ_NONE 0 +#define HPIRQ_5 1 +#define HPIRQ_7 2 +#define HPIRQ_9 3 +#define HPIRQ_10 4 +#define HPIRQ_11 5 +#define HPIRQ_12 6 +#define HPIRQ_15 7 + +#define HIMT_PLAY_DONE 0x00 +#define HIMT_RECORD_DONE 0x01 +#define HIMT_MIDI_EOS 0x02 +#define HIMT_MIDI_OUT 0x03 + +#define HIMT_MIDI_IN_UCHAR 0x0E +#define HIMT_DSP 0x0F + +#define HDEX_BASE 0x92 +#define HDEX_PLAY_START (0 + HDEX_BASE) +#define HDEX_PLAY_STOP (1 + HDEX_BASE) +#define HDEX_PLAY_PAUSE (2 + HDEX_BASE) +#define HDEX_PLAY_RESUME (3 + HDEX_BASE) +#define HDEX_RECORD_START (4 + HDEX_BASE) +#define HDEX_RECORD_STOP (5 + HDEX_BASE) +#define HDEX_MIDI_IN_START (6 + HDEX_BASE) +#define HDEX_MIDI_IN_STOP (7 + HDEX_BASE) +#define HDEX_MIDI_OUT_START (8 + HDEX_BASE) +#define HDEX_MIDI_OUT_STOP (9 + HDEX_BASE) +#define HDEX_AUX_REQ (10 + HDEX_BASE) + +#define HDEXAR_CLEAR_PEAKS 1 +#define HDEXAR_IN_SET_POTS 2 +#define HDEXAR_AUX_SET_POTS 3 +#define HDEXAR_CAL_A_TO_D 4 +#define HDEXAR_RD_EXT_DSP_BITS 5 + +/* Pinnacle only HDEXAR defs */ +#define HDEXAR_SET_ANA_IN 0 +#define HDEXAR_SET_SYNTH_IN 4 +#define HDEXAR_READ_DAT_IN 5 +#define HDEXAR_MIC_SET_POTS 6 +#define HDEXAR_SET_DAT_IN 7 + +#define HDEXAR_SET_SYNTH_48 8 +#define HDEXAR_SET_SYNTH_44 9 + +#define HIWORD(l) ((u16)((((u32)(l)) >> 16) & 0xFFFF)) +#define LOWORD(l) ((u16)(u32)(l)) +#define HIBYTE(w) ((u8)(((u16)(w) >> 8) & 0xFF)) +#define LOBYTE(w) ((u8)(w)) +#define MAKELONG(low, hi) ((long)(((u16)(low))|(((u32)((u16)(hi)))<<16))) +#define MAKEWORD(low, hi) ((u16)(((u8)(low))|(((u16)((u8)(hi)))<<8))) + +#define PCTODSP_OFFSET(w) (u16)((w)/2) +#define PCTODSP_BASED(w) (u16)(((w)/2) + DSP_BASE_ADDR) +#define DSPTOPC_BASED(w) (((w) - DSP_BASE_ADDR) * 2) + +#ifdef SLOWIO +# undef outb +# undef inb +# define outb outb_p +# define inb inb_p +#endif + +/* JobQueueStruct */ +#define JQS_wStart 0x00 +#define JQS_wSize 0x02 +#define JQS_wHead 0x04 +#define JQS_wTail 0x06 +#define JQS__size 0x08 + +/* DAQueueDataStruct */ +#define DAQDS_wStart 0x00 +#define DAQDS_wSize 0x02 +#define DAQDS_wFormat 0x04 +#define DAQDS_wSampleSize 0x06 +#define DAQDS_wChannels 0x08 +#define DAQDS_wSampleRate 0x0A +#define DAQDS_wIntMsg 0x0C +#define DAQDS_wFlags 0x0E +#define DAQDS__size 0x10 + +#include <sound/pcm.h> + +struct snd_msnd { + void __iomem *mappedbase; + int play_period_bytes; + int playLimit; + int playPeriods; + int playDMAPos; + int banksPlayed; + int captureDMAPos; + int capturePeriodBytes; + int captureLimit; + int capturePeriods; + struct snd_card *card; + void *msndmidi_mpu; + struct snd_rawmidi *rmidi; + + /* Hardware resources */ + long io; + int memid, irqid; + int irq, irq_ref; + unsigned long base; + + /* Motorola 56k DSP SMA */ + void __iomem *SMA; + void __iomem *DAPQ; + void __iomem *DARQ; + void __iomem *MODQ; + void __iomem *MIDQ; + void __iomem *DSPQ; + int dspq_data_buff, dspq_buff_size; + + /* State variables */ + enum { msndClassic, msndPinnacle } type; + mode_t mode; + unsigned long flags; +#define F_RESETTING 0 +#define F_HAVEDIGITAL 1 +#define F_AUDIO_WRITE_INUSE 2 +#define F_WRITING 3 +#define F_WRITEBLOCK 4 +#define F_WRITEFLUSH 5 +#define F_AUDIO_READ_INUSE 6 +#define F_READING 7 +#define F_READBLOCK 8 +#define F_EXT_MIDI_INUSE 9 +#define F_HDR_MIDI_INUSE 10 +#define F_DISABLE_WRITE_NDELAY 11 + spinlock_t lock; + spinlock_t mixer_lock; + int nresets; + unsigned recsrc; +#define LEVEL_ENTRIES 32 + int left_levels[LEVEL_ENTRIES]; + int right_levels[LEVEL_ENTRIES]; + int calibrate_signal; + int play_sample_size, play_sample_rate, play_channels; + int play_ndelay; + int capture_sample_size, capture_sample_rate, capture_channels; + int capture_ndelay; + u8 bCurrentMidiPatch; + + int last_playbank, last_recbank; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + +}; + +void snd_msnd_init_queue(void *base, int start, int size); + +int snd_msnd_send_dsp_cmd(struct snd_msnd *chip, u8 cmd); +int snd_msnd_send_word(struct snd_msnd *chip, + unsigned char high, + unsigned char mid, + unsigned char low); +int snd_msnd_upload_host(struct snd_msnd *chip, + const u8 *bin, int len); +int snd_msnd_enable_irq(struct snd_msnd *chip); +int snd_msnd_disable_irq(struct snd_msnd *chip); +void snd_msnd_dsp_halt(struct snd_msnd *chip, struct file *file); +int snd_msnd_DAPQ(struct snd_msnd *chip, int start); +int snd_msnd_DARQ(struct snd_msnd *chip, int start); +int snd_msnd_pcm(struct snd_card *card, int device, struct snd_pcm **rpcm); + +int snd_msndmidi_new(struct snd_card *card, int device); +void snd_msndmidi_input_read(void *mpu); + +void snd_msndmix_setup(struct snd_msnd *chip); +int __devinit snd_msndmix_new(struct snd_card *card); +int snd_msndmix_force_recsrc(struct snd_msnd *chip, int recsrc); +#endif /* __MSND_H */ diff --git a/sound/isa/msnd/msnd_classic.c b/sound/isa/msnd/msnd_classic.c new file mode 100644 index 0000000..3b23a09 --- /dev/null +++ b/sound/isa/msnd/msnd_classic.c @@ -0,0 +1,3 @@ +/* The work is in msnd_pinnacle.c, just define MSND_CLASSIC before it. */ +#define MSND_CLASSIC +#include "msnd_pinnacle.c" diff --git a/sound/isa/msnd/msnd_classic.h b/sound/isa/msnd/msnd_classic.h new file mode 100644 index 0000000..f18d5fa --- /dev/null +++ b/sound/isa/msnd/msnd_classic.h @@ -0,0 +1,129 @@ +/********************************************************************* + * + * msnd_classic.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_CLASSIC_H +#define __MSND_CLASSIC_H + +#define DSP_NUMIO 0x10 + +#define HP_MEMM 0x08 + +#define HP_BITM 0x0E +#define HP_WAIT 0x0D +#define HP_DSPR 0x0A +#define HP_PROR 0x0B +#define HP_BLKS 0x0C + +#define HPPRORESET_OFF 0 +#define HPPRORESET_ON 1 + +#define HPDSPRESET_OFF 0 +#define HPDSPRESET_ON 1 + +#define HPBLKSEL_0 0 +#define HPBLKSEL_1 1 + +#define HPWAITSTATE_0 0 +#define HPWAITSTATE_1 1 + +#define HPBITMODE_16 0 +#define HPBITMODE_8 1 + +#define HIDSP_INT_PLAY_UNDER 0x00 +#define HIDSP_INT_RECORD_OVER 0x01 +#define HIDSP_INPUT_CLIPPING 0x02 +#define HIDSP_MIDI_IN_OVER 0x10 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x0040 +#define TIME_PRO_RESET 0x0032 + +#define DAR_BUFF_SIZE 0x2000 + +#define MIDQ_BUFF_SIZE 0x200 +#define DSPQ_BUFF_SIZE 0x40 + +#define DSPQ_DATA_BUFF 0x7260 + +#define MOP_SYNTH 0x10 +#define MOP_EXTOUT 0x32 +#define MOP_EXTTHRU 0x02 +#define MOP_OUTMASK 0x01 + +#define MIP_EXTIN 0x01 +#define MIP_SYNTH 0x00 +#define MIP_INMASK 0x32 + +/* Classic SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wUser_3 0x000c +#define SMA_wUser_4 0x000e +#define SMA_dwUser_5 0x0010 +#define SMA_dwUser_6 0x0014 +#define SMA_wUser_7 0x0018 +#define SMA_wReserved_A 0x001a +#define SMA_wReserved_B 0x001c +#define SMA_wReserved_C 0x001e +#define SMA_wReserved_D 0x0020 +#define SMA_wReserved_E 0x0022 +#define SMA_wReserved_F 0x0024 +#define SMA_wReserved_G 0x0026 +#define SMA_wReserved_H 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_wExtDSPbits 0x0034 +#define SMA_bExtHostbits 0x0036 +#define SMA_bBoardLevel 0x0037 +#define SMA_bInPotPosRight 0x0038 +#define SMA_bInPotPosLeft 0x0039 +#define SMA_bAuxPotPosRight 0x003a +#define SMA_bAuxPotPosLeft 0x003b +#define SMA_wCurrMastVolLeft 0x003c +#define SMA_wCurrMastVolRight 0x003e +#define SMA_bUser_12 0x0040 +#define SMA_bUser_13 0x0041 +#define SMA_wUser_14 0x0042 +#define SMA_wUser_15 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wUser_16 0x0048 +#define SMA_wUser_17 0x004a +#define SMA__size 0x004c + +#define INITCODEFILE "turtlebeach/msndinit.bin" +#define PERMCODEFILE "turtlebeach/msndperm.bin" +#define LONGNAME "MultiSound (Classic/Monterey/Tahiti)" + +#endif /* __MSND_CLASSIC_H */ diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c new file mode 100644 index 0000000..cb9aa4c --- /dev/null +++ b/sound/isa/msnd/msnd_midi.c @@ -0,0 +1,180 @@ +/* + * Copyright (c) by Jaroslav Kysela <perex@perex.cz> + * Copyright (c) 2009 by Krzysztof Helt + * Routines for control of MPU-401 in UART mode + * + * MPU-401 supports UART mode which is not capable generate transmit + * interrupts thus output is done via polling. Also, if irq < 0, then + * input is done also via polling. Do not expect good performance. + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/ioport.h> +#include <linux/errno.h> +#include <sound/core.h> +#include <sound/rawmidi.h> + +#include "msnd.h" + +#define MSNDMIDI_MODE_BIT_INPUT 0 +#define MSNDMIDI_MODE_BIT_OUTPUT 1 +#define MSNDMIDI_MODE_BIT_INPUT_TRIGGER 2 +#define MSNDMIDI_MODE_BIT_OUTPUT_TRIGGER 3 + +struct snd_msndmidi { + struct snd_msnd *dev; + + unsigned long mode; /* MSNDMIDI_MODE_XXXX */ + + struct snd_rawmidi_substream *substream_input; + + spinlock_t input_lock; +}; + +/* + * input/output open/close - protected by open_mutex in rawmidi.c + */ +static int snd_msndmidi_input_open(struct snd_rawmidi_substream *substream) +{ + struct snd_msndmidi *mpu; + + snd_printdd("snd_msndmidi_input_open()\n"); + + mpu = substream->rmidi->private_data; + + mpu->substream_input = substream; + + snd_msnd_enable_irq(mpu->dev); + + snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_START); + set_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode); + return 0; +} + +static int snd_msndmidi_input_close(struct snd_rawmidi_substream *substream) +{ + struct snd_msndmidi *mpu; + + mpu = substream->rmidi->private_data; + snd_msnd_send_dsp_cmd(mpu->dev, HDEX_MIDI_IN_STOP); + clear_bit(MSNDMIDI_MODE_BIT_INPUT, &mpu->mode); + mpu->substream_input = NULL; + snd_msnd_disable_irq(mpu->dev); + return 0; +} + +static void snd_msndmidi_input_drop(struct snd_msndmidi *mpu) +{ + u16 tail; + + tail = readw(mpu->dev->MIDQ + JQS_wTail); + writew(tail, mpu->dev->MIDQ + JQS_wHead); +} + +/* + * trigger input + */ +static void snd_msndmidi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + unsigned long flags; + struct snd_msndmidi *mpu; + + snd_printdd("snd_msndmidi_input_trigger(, %i)\n", up); + + mpu = substream->rmidi->private_data; + spin_lock_irqsave(&mpu->input_lock, flags); + if (up) { + if (!test_and_set_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, + &mpu->mode)) + snd_msndmidi_input_drop(mpu); + } else { + clear_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, &mpu->mode); + } + spin_unlock_irqrestore(&mpu->input_lock, flags); + if (up) + snd_msndmidi_input_read(mpu); +} + +void snd_msndmidi_input_read(void *mpuv) +{ + unsigned long flags; + struct snd_msndmidi *mpu = mpuv; + void *pwMIDQData = mpu->dev->mappedbase + MIDQ_DATA_BUFF; + + spin_lock_irqsave(&mpu->input_lock, flags); + while (readw(mpu->dev->MIDQ + JQS_wTail) != + readw(mpu->dev->MIDQ + JQS_wHead)) { + u16 wTmp, val; + val = readw(pwMIDQData + 2 * readw(mpu->dev->MIDQ + JQS_wHead)); + + if (test_bit(MSNDMIDI_MODE_BIT_INPUT_TRIGGER, + &mpu->mode)) + snd_rawmidi_receive(mpu->substream_input, + (unsigned char *)&val, 1); + + wTmp = readw(mpu->dev->MIDQ + JQS_wHead) + 1; + if (wTmp > readw(mpu->dev->MIDQ + JQS_wSize)) + writew(0, mpu->dev->MIDQ + JQS_wHead); + else + writew(wTmp, mpu->dev->MIDQ + JQS_wHead); + } + spin_unlock_irqrestore(&mpu->input_lock, flags); +} +EXPORT_SYMBOL(snd_msndmidi_input_read); + +static struct snd_rawmidi_ops snd_msndmidi_input = { + .open = snd_msndmidi_input_open, + .close = snd_msndmidi_input_close, + .trigger = snd_msndmidi_input_trigger, +}; + +static void snd_msndmidi_free(struct snd_rawmidi *rmidi) +{ + struct snd_msndmidi *mpu = rmidi->private_data; + kfree(mpu); +} + +int snd_msndmidi_new(struct snd_card *card, int device) +{ + struct snd_msnd *chip = card->private_data; + struct snd_msndmidi *mpu; + struct snd_rawmidi *rmidi; + int err; + + err = snd_rawmidi_new(card, "MSND-MIDI", device, 1, 1, &rmidi); + if (err < 0) + return err; + mpu = kcalloc(1, sizeof(*mpu), GFP_KERNEL); + if (mpu == NULL) { + snd_device_free(card, rmidi); + return -ENOMEM; + } + mpu->dev = chip; + chip->msndmidi_mpu = mpu; + rmidi->private_data = mpu; + rmidi->private_free = snd_msndmidi_free; + spin_lock_init(&mpu->input_lock); + strcpy(rmidi->name, "MSND MIDI"); + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &snd_msndmidi_input); + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + return 0; +} diff --git a/sound/isa/msnd/msnd_pinnacle.c b/sound/isa/msnd/msnd_pinnacle.c new file mode 100644 index 0000000..60b6abd --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle.c @@ -0,0 +1,1238 @@ +/********************************************************************* + * + * Linux multisound pinnacle/fiji driver for ALSA. + * + * 2002/06/30 Karsten Wiese: + * for now this is only used to build a pinnacle / fiji driver. + * the OSS parent of this code is designed to also support + * the multisound classic via the file msnd_classic.c. + * to make it easier for some brave heart to implemt classic + * support in alsa, i left all the MSND_CLASSIC tokens in this file. + * but for now this untested & undone. + * + * + * ripped from linux kernel 2.4.18 by Karsten Wiese. + * + * the following is a copy of the 2.4.18 OSS FREE file-heading comment: + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * msnd_pinnacle.c / msnd_classic.c + * + * -- If MSND_CLASSIC is defined: + * + * -> driver for Turtle Beach Classic/Monterey/Tahiti + * + * -- Else + * + * -> driver for Turtle Beach Pinnacle/Fiji + * + * 12-3-2000 Modified IO port validation Steve Sycamore + * + * Copyright (C) 1998 Andrew Veliath + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ + +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/types.h> +#include <linux/delay.h> +#include <linux/ioport.h> +#include <linux/firmware.h> +#include <linux/isa.h> +#include <linux/isapnp.h> +#include <linux/irq.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/asound.h> +#include <sound/pcm.h> +#include <sound/mpu401.h> + +#ifdef MSND_CLASSIC +# ifndef __alpha__ +# define SLOWIO +# endif +#endif +#include "msnd.h" +#ifdef MSND_CLASSIC +# include "msnd_classic.h" +# define LOGNAME "msnd_classic" +#else +# include "msnd_pinnacle.h" +# define LOGNAME "snd_msnd_pinnacle" +#endif + +static void __devinit set_default_audio_parameters(struct snd_msnd *chip) +{ + chip->play_sample_size = DEFSAMPLESIZE; + chip->play_sample_rate = DEFSAMPLERATE; + chip->play_channels = DEFCHANNELS; + chip->capture_sample_size = DEFSAMPLESIZE; + chip->capture_sample_rate = DEFSAMPLERATE; + chip->capture_channels = DEFCHANNELS; +} + +static void snd_msnd_eval_dsp_msg(struct snd_msnd *chip, u16 wMessage) +{ + switch (HIBYTE(wMessage)) { + case HIMT_PLAY_DONE: { + if (chip->banksPlayed < 3) + snd_printdd("%08X: HIMT_PLAY_DONE: %i\n", + (unsigned)jiffies, LOBYTE(wMessage)); + + if (chip->last_playbank == LOBYTE(wMessage)) { + snd_printdd("chip.last_playbank == LOBYTE(wMessage)\n"); + break; + } + chip->banksPlayed++; + + if (test_bit(F_WRITING, &chip->flags)) + snd_msnd_DAPQ(chip, 0); + + chip->last_playbank = LOBYTE(wMessage); + chip->playDMAPos += chip->play_period_bytes; + if (chip->playDMAPos > chip->playLimit) + chip->playDMAPos = 0; + snd_pcm_period_elapsed(chip->playback_substream); + + break; + } + case HIMT_RECORD_DONE: + if (chip->last_recbank == LOBYTE(wMessage)) + break; + chip->last_recbank = LOBYTE(wMessage); + chip->captureDMAPos += chip->capturePeriodBytes; + if (chip->captureDMAPos > (chip->captureLimit)) + chip->captureDMAPos = 0; + + if (test_bit(F_READING, &chip->flags)) + snd_msnd_DARQ(chip, chip->last_recbank); + + snd_pcm_period_elapsed(chip->capture_substream); + break; + + case HIMT_DSP: + switch (LOBYTE(wMessage)) { +#ifndef MSND_CLASSIC + case HIDSP_PLAY_UNDER: +#endif + case HIDSP_INT_PLAY_UNDER: + snd_printd(KERN_WARNING LOGNAME ": Play underflow %i\n", + chip->banksPlayed); + if (chip->banksPlayed > 2) + clear_bit(F_WRITING, &chip->flags); + break; + + case HIDSP_INT_RECORD_OVER: + snd_printd(KERN_WARNING LOGNAME ": Record overflow\n"); + clear_bit(F_READING, &chip->flags); + break; + + default: + snd_printd(KERN_WARNING LOGNAME + ": DSP message %d 0x%02x\n", + LOBYTE(wMessage), LOBYTE(wMessage)); + break; + } + break; + + case HIMT_MIDI_IN_UCHAR: + if (chip->msndmidi_mpu) + snd_msndmidi_input_read(chip->msndmidi_mpu); + break; + + default: + snd_printd(KERN_WARNING LOGNAME ": HIMT message %d 0x%02x\n", + HIBYTE(wMessage), HIBYTE(wMessage)); + break; + } +} + +static irqreturn_t snd_msnd_interrupt(int irq, void *dev_id) +{ + struct snd_msnd *chip = dev_id; + void *pwDSPQData = chip->mappedbase + DSPQ_DATA_BUFF; + + /* Send ack to DSP */ + /* inb(chip->io + HP_RXL); */ + + /* Evaluate queued DSP messages */ + while (readw(chip->DSPQ + JQS_wTail) != readw(chip->DSPQ + JQS_wHead)) { + u16 wTmp; + + snd_msnd_eval_dsp_msg(chip, + readw(pwDSPQData + 2 * readw(chip->DSPQ + JQS_wHead))); + + wTmp = readw(chip->DSPQ + JQS_wHead) + 1; + if (wTmp > readw(chip->DSPQ + JQS_wSize)) + writew(0, chip->DSPQ + JQS_wHead); + else + writew(wTmp, chip->DSPQ + JQS_wHead); + } + /* Send ack to DSP */ + inb(chip->io + HP_RXL); + return IRQ_HANDLED; +} + + +static int snd_msnd_reset_dsp(long io, unsigned char *info) +{ + int timeout = 100; + + outb(HPDSPRESET_ON, io + HP_DSPR); + msleep(1); +#ifndef MSND_CLASSIC + if (info) + *info = inb(io + HP_INFO); +#endif + outb(HPDSPRESET_OFF, io + HP_DSPR); + msleep(1); + while (timeout-- > 0) { + if (inb(io + HP_CVR) == HP_CVR_DEF) + return 0; + msleep(1); + } + snd_printk(KERN_ERR LOGNAME ": Cannot reset DSP\n"); + + return -EIO; +} + +static int __devinit snd_msnd_probe(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + unsigned char info; +#ifndef MSND_CLASSIC + char *xv, *rev = NULL; + char *pin = "TB Pinnacle", *fiji = "TB Fiji"; + char *pinfiji = "TB Pinnacle/Fiji"; +#endif + + if (!request_region(chip->io, DSP_NUMIO, "probing")) { + snd_printk(KERN_ERR LOGNAME ": I/O port conflict\n"); + return -ENODEV; + } + + if (snd_msnd_reset_dsp(chip->io, &info) < 0) { + release_region(chip->io, DSP_NUMIO); + return -ENODEV; + } + +#ifdef MSND_CLASSIC + strcpy(card->shortname, "Classic/Tahiti/Monterey"); + strcpy(card->longname, "Turtle Beach Multisound"); + printk(KERN_INFO LOGNAME ": %s, " + "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n", + card->shortname, + chip->io, chip->io + DSP_NUMIO - 1, + chip->irq, + chip->base, chip->base + 0x7fff); +#else + switch (info >> 4) { + case 0xf: + xv = "<= 1.15"; + break; + case 0x1: + xv = "1.18/1.2"; + break; + case 0x2: + xv = "1.3"; + break; + case 0x3: + xv = "1.4"; + break; + default: + xv = "unknown"; + break; + } + + switch (info & 0x7) { + case 0x0: + rev = "I"; + strcpy(card->shortname, pin); + break; + case 0x1: + rev = "F"; + strcpy(card->shortname, pin); + break; + case 0x2: + rev = "G"; + strcpy(card->shortname, pin); + break; + case 0x3: + rev = "H"; + strcpy(card->shortname, pin); + break; + case 0x4: + rev = "E"; + strcpy(card->shortname, fiji); + break; + case 0x5: + rev = "C"; + strcpy(card->shortname, fiji); + break; + case 0x6: + rev = "D"; + strcpy(card->shortname, fiji); + break; + case 0x7: + rev = "A-B (Fiji) or A-E (Pinnacle)"; + strcpy(card->shortname, pinfiji); + break; + } + strcpy(card->longname, "Turtle Beach Multisound Pinnacle"); + printk(KERN_INFO LOGNAME ": %s revision %s, Xilinx version %s, " + "I/O 0x%lx-0x%lx, IRQ %d, memory mapped to 0x%lX-0x%lX\n", + card->shortname, + rev, xv, + chip->io, chip->io + DSP_NUMIO - 1, + chip->irq, + chip->base, chip->base + 0x7fff); +#endif + + release_region(chip->io, DSP_NUMIO); + return 0; +} + +static int snd_msnd_init_sma(struct snd_msnd *chip) +{ + static int initted; + u16 mastVolLeft, mastVolRight; + unsigned long flags; + +#ifdef MSND_CLASSIC + outb(chip->memid, chip->io + HP_MEMM); +#endif + outb(HPBLKSEL_0, chip->io + HP_BLKS); + /* Motorola 56k shared memory base */ + chip->SMA = chip->mappedbase + SMA_STRUCT_START; + + if (initted) { + mastVolLeft = readw(chip->SMA + SMA_wCurrMastVolLeft); + mastVolRight = readw(chip->SMA + SMA_wCurrMastVolRight); + } else + mastVolLeft = mastVolRight = 0; + memset_io(chip->mappedbase, 0, 0x8000); + + /* Critical section: bank 1 access */ + spin_lock_irqsave(&chip->lock, flags); + outb(HPBLKSEL_1, chip->io + HP_BLKS); + memset_io(chip->mappedbase, 0, 0x8000); + outb(HPBLKSEL_0, chip->io + HP_BLKS); + spin_unlock_irqrestore(&chip->lock, flags); + + /* Digital audio play queue */ + chip->DAPQ = chip->mappedbase + DAPQ_OFFSET; + snd_msnd_init_queue(chip->DAPQ, DAPQ_DATA_BUFF, DAPQ_BUFF_SIZE); + + /* Digital audio record queue */ + chip->DARQ = chip->mappedbase + DARQ_OFFSET; + snd_msnd_init_queue(chip->DARQ, DARQ_DATA_BUFF, DARQ_BUFF_SIZE); + + /* MIDI out queue */ + chip->MODQ = chip->mappedbase + MODQ_OFFSET; + snd_msnd_init_queue(chip->MODQ, MODQ_DATA_BUFF, MODQ_BUFF_SIZE); + + /* MIDI in queue */ + chip->MIDQ = chip->mappedbase + MIDQ_OFFSET; + snd_msnd_init_queue(chip->MIDQ, MIDQ_DATA_BUFF, MIDQ_BUFF_SIZE); + + /* DSP -> host message queue */ + chip->DSPQ = chip->mappedbase + DSPQ_OFFSET; + snd_msnd_init_queue(chip->DSPQ, DSPQ_DATA_BUFF, DSPQ_BUFF_SIZE); + + /* Setup some DSP values */ +#ifndef MSND_CLASSIC + writew(1, chip->SMA + SMA_wCurrPlayFormat); + writew(chip->play_sample_size, chip->SMA + SMA_wCurrPlaySampleSize); + writew(chip->play_channels, chip->SMA + SMA_wCurrPlayChannels); + writew(chip->play_sample_rate, chip->SMA + SMA_wCurrPlaySampleRate); +#endif + writew(chip->play_sample_rate, chip->SMA + SMA_wCalFreqAtoD); + writew(mastVolLeft, chip->SMA + SMA_wCurrMastVolLeft); + writew(mastVolRight, chip->SMA + SMA_wCurrMastVolRight); +#ifndef MSND_CLASSIC + writel(0x00010000, chip->SMA + SMA_dwCurrPlayPitch); + writel(0x00000001, chip->SMA + SMA_dwCurrPlayRate); +#endif + writew(0x303, chip->SMA + SMA_wCurrInputTagBits); + + initted = 1; + + return 0; +} + + +static int upload_dsp_code(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + const struct firmware *init_fw = NULL, *perm_fw = NULL; + int err; + + outb(HPBLKSEL_0, chip->io + HP_BLKS); + + err = request_firmware(&init_fw, INITCODEFILE, card->dev); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Error loading " INITCODEFILE); + goto cleanup1; + } + err = request_firmware(&perm_fw, PERMCODEFILE, card->dev); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Error loading " PERMCODEFILE); + goto cleanup; + } + + memcpy_toio(chip->mappedbase, perm_fw->data, perm_fw->size); + if (snd_msnd_upload_host(chip, init_fw->data, init_fw->size) < 0) { + printk(KERN_WARNING LOGNAME ": Error uploading to DSP\n"); + err = -ENODEV; + goto cleanup; + } + printk(KERN_INFO LOGNAME ": DSP firmware uploaded\n"); + err = 0; + +cleanup: + release_firmware(perm_fw); +cleanup1: + release_firmware(init_fw); + return err; +} + +#ifdef MSND_CLASSIC +static void reset_proteus(struct snd_msnd *chip) +{ + outb(HPPRORESET_ON, chip->io + HP_PROR); + msleep(TIME_PRO_RESET); + outb(HPPRORESET_OFF, chip->io + HP_PROR); + msleep(TIME_PRO_RESET_DONE); +} +#endif + +static int snd_msnd_initialize(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err, timeout; + +#ifdef MSND_CLASSIC + outb(HPWAITSTATE_0, chip->io + HP_WAIT); + outb(HPBITMODE_16, chip->io + HP_BITM); + + reset_proteus(chip); +#endif + err = snd_msnd_init_sma(chip); + if (err < 0) { + printk(KERN_WARNING LOGNAME ": Cannot initialize SMA\n"); + return err; + } + + err = snd_msnd_reset_dsp(chip->io, NULL); + if (err < 0) + return err; + + err = upload_dsp_code(card); + if (err < 0) { + printk(KERN_WARNING LOGNAME ": Cannot upload DSP code\n"); + return err; + } + + timeout = 200; + + while (readw(chip->mappedbase)) { + msleep(1); + if (!timeout--) { + snd_printd(KERN_ERR LOGNAME ": DSP reset timeout\n"); + return -EIO; + } + } + + snd_msndmix_setup(chip); + return 0; +} + +static int snd_msnd_dsp_full_reset(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int rv; + + if (test_bit(F_RESETTING, &chip->flags) || ++chip->nresets > 10) + return 0; + + set_bit(F_RESETTING, &chip->flags); + snd_msnd_dsp_halt(chip, NULL); /* Unconditionally halt */ + + rv = snd_msnd_initialize(card); + if (rv) + printk(KERN_WARNING LOGNAME ": DSP reset failed\n"); + snd_msndmix_force_recsrc(chip, 0); + clear_bit(F_RESETTING, &chip->flags); + return rv; +} + +static int snd_msnd_dev_free(struct snd_device *device) +{ + snd_printdd("snd_msnd_chip_free()\n"); + return 0; +} + +static int snd_msnd_send_dsp_cmd_chk(struct snd_msnd *chip, u8 cmd) +{ + if (snd_msnd_send_dsp_cmd(chip, cmd) == 0) + return 0; + snd_msnd_dsp_full_reset(chip->card); + return snd_msnd_send_dsp_cmd(chip, cmd); +} + +static int __devinit snd_msnd_calibrate_adc(struct snd_msnd *chip, u16 srate) +{ + snd_printdd("snd_msnd_calibrate_adc(%i)\n", srate); + writew(srate, chip->SMA + SMA_wCalFreqAtoD); + if (chip->calibrate_signal == 0) + writew(readw(chip->SMA + SMA_wCurrHostStatusFlags) + | 0x0001, chip->SMA + SMA_wCurrHostStatusFlags); + else + writew(readw(chip->SMA + SMA_wCurrHostStatusFlags) + & ~0x0001, chip->SMA + SMA_wCurrHostStatusFlags); + if (snd_msnd_send_word(chip, 0, 0, HDEXAR_CAL_A_TO_D) == 0 && + snd_msnd_send_dsp_cmd_chk(chip, HDEX_AUX_REQ) == 0) { + schedule_timeout_interruptible(msecs_to_jiffies(333)); + return 0; + } + printk(KERN_WARNING LOGNAME ": ADC calibration failed\n"); + return -EIO; +} + +/* + * ALSA callback function, called when attempting to open the MIDI device. + */ +static int snd_msnd_mpu401_open(struct snd_mpu401 *mpu) +{ + snd_msnd_enable_irq(mpu->private_data); + snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_START); + return 0; +} + +static void snd_msnd_mpu401_close(struct snd_mpu401 *mpu) +{ + snd_msnd_send_dsp_cmd(mpu->private_data, HDEX_MIDI_IN_STOP); + snd_msnd_disable_irq(mpu->private_data); +} + +static long mpu_io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; + +static int __devinit snd_msnd_attach(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_msnd_dev_free, + }; + + err = request_irq(chip->irq, snd_msnd_interrupt, 0, card->shortname, + chip); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Couldn't grab IRQ %d\n", chip->irq); + return err; + } + request_region(chip->io, DSP_NUMIO, card->shortname); + + if (!request_mem_region(chip->base, BUFFSIZE, card->shortname)) { + printk(KERN_ERR LOGNAME + ": unable to grab memory region 0x%lx-0x%lx\n", + chip->base, chip->base + BUFFSIZE - 1); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + return -EBUSY; + } + chip->mappedbase = ioremap_nocache(chip->base, 0x8000); + if (!chip->mappedbase) { + printk(KERN_ERR LOGNAME + ": unable to map memory region 0x%lx-0x%lx\n", + chip->base, chip->base + BUFFSIZE - 1); + err = -EIO; + goto err_release_region; + } + + err = snd_msnd_dsp_full_reset(card); + if (err < 0) + goto err_release_region; + + /* Register device */ + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) + goto err_release_region; + + err = snd_msnd_pcm(card, 0, NULL); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new PCM device\n"); + goto err_release_region; + } + + err = snd_msndmix_new(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": error creating new Mixer device\n"); + goto err_release_region; + } + + + if (mpu_io[0] != SNDRV_AUTO_PORT) { + struct snd_mpu401 *mpu; + + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_io[0], + MPU401_MODE_INPUT | + MPU401_MODE_OUTPUT, + mpu_irq[0], IRQF_DISABLED, + &chip->rmidi); + if (err < 0) { + printk(KERN_ERR LOGNAME + ": error creating new Midi device\n"); + goto err_release_region; + } + mpu = chip->rmidi->private_data; + + mpu->open_input = snd_msnd_mpu401_open; + mpu->close_input = snd_msnd_mpu401_close; + mpu->private_data = chip; + } + + disable_irq(chip->irq); + snd_msnd_calibrate_adc(chip, chip->play_sample_rate); + snd_msndmix_force_recsrc(chip, 0); + + err = snd_card_register(card); + if (err < 0) + goto err_release_region; + + return 0; + +err_release_region: + if (chip->mappedbase) + iounmap(chip->mappedbase); + release_mem_region(chip->base, BUFFSIZE); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + return err; +} + + +static void __devexit snd_msnd_unload(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + + iounmap(chip->mappedbase); + release_mem_region(chip->base, BUFFSIZE); + release_region(chip->io, DSP_NUMIO); + free_irq(chip->irq, chip); + snd_card_free(card); +} + +#ifndef MSND_CLASSIC + +/* Pinnacle/Fiji Logical Device Configuration */ + +static int __devinit snd_msnd_write_cfg(int cfg, int reg, int value) +{ + outb(reg, cfg); + outb(value, cfg + 1); + if (value != inb(cfg + 1)) { + printk(KERN_ERR LOGNAME ": snd_msnd_write_cfg: I/O error\n"); + return -EIO; + } + return 0; +} + +static int __devinit snd_msnd_write_cfg_io0(int cfg, int num, u16 io) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO0_BASEHI, HIBYTE(io))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO0_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_io1(int cfg, int num, u16 io) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO1_BASEHI, HIBYTE(io))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IO1_BASELO, LOBYTE(io))) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_irq(int cfg, int num, u16 irq) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IRQ_NUMBER, LOBYTE(irq))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_IRQ_TYPE, IRQTYPE_EDGE)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_mem(int cfg, int num, int mem) +{ + u16 wmem; + + mem >>= 8; + wmem = (u16)(mem & 0xfff); + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_MEMBASEHI, HIBYTE(wmem))) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_MEMBASELO, LOBYTE(wmem))) + return -EIO; + if (wmem && snd_msnd_write_cfg(cfg, IREG_MEMCONTROL, + MEMTYPE_HIADDR | MEMTYPE_16BIT)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_activate_logical(int cfg, int num) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg(cfg, IREG_ACTIVATE, LD_ACTIVATE)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_write_cfg_logical(int cfg, int num, u16 io0, + u16 io1, u16 irq, int mem) +{ + if (snd_msnd_write_cfg(cfg, IREG_LOGDEVICE, num)) + return -EIO; + if (snd_msnd_write_cfg_io0(cfg, num, io0)) + return -EIO; + if (snd_msnd_write_cfg_io1(cfg, num, io1)) + return -EIO; + if (snd_msnd_write_cfg_irq(cfg, num, irq)) + return -EIO; + if (snd_msnd_write_cfg_mem(cfg, num, mem)) + return -EIO; + if (snd_msnd_activate_logical(cfg, num)) + return -EIO; + return 0; +} + +static int __devinit snd_msnd_pinnacle_cfg_reset(int cfg) +{ + int i; + + /* Reset devices if told to */ + printk(KERN_INFO LOGNAME ": Resetting all devices\n"); + for (i = 0; i < 4; ++i) + if (snd_msnd_write_cfg_logical(cfg, i, 0, 0, 0, 0)) + return -EIO; + + return 0; +} +#endif + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ + +module_param_array(index, int, NULL, S_IRUGO); +MODULE_PARM_DESC(index, "Index value for msnd_pinnacle soundcard."); +module_param_array(id, charp, NULL, S_IRUGO); +MODULE_PARM_DESC(id, "ID string for msnd_pinnacle soundcard."); + +static long io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static long mem[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; + +static long cfg[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; + +#ifndef MSND_CLASSIC +/* Extra Peripheral Configuration (Default: Disable) */ +static long ide_io0[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long ide_io1[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int ide_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; + +static long joystick_io[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +/* If we have the digital daugherboard... */ +static int digital[SNDRV_CARDS]; + +/* Extra Peripheral Configuration */ +static int reset[SNDRV_CARDS]; +#endif + +static int write_ndelay[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)] = 1 }; + +static int calibrate_signal; + +#ifdef CONFIG_PNP +static int isapnp[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; +module_param_array(isapnp, bool, NULL, 0444); +MODULE_PARM_DESC(isapnp, "ISA PnP detection for specified soundcard."); +#define has_isapnp(x) isapnp[x] +#else +#define has_isapnp(x) 0 +#endif + +MODULE_AUTHOR("Karsten Wiese <annabellesgarden@yahoo.de>"); +MODULE_DESCRIPTION("Turtle Beach " LONGNAME " Linux Driver"); +MODULE_LICENSE("GPL"); +MODULE_FIRMWARE(INITCODEFILE); +MODULE_FIRMWARE(PERMCODEFILE); + +module_param_array(io, long, NULL, S_IRUGO); +MODULE_PARM_DESC(io, "IO port #"); +module_param_array(irq, int, NULL, S_IRUGO); +module_param_array(mem, long, NULL, S_IRUGO); +module_param_array(write_ndelay, int, NULL, S_IRUGO); +module_param(calibrate_signal, int, S_IRUGO); +#ifndef MSND_CLASSIC +module_param_array(digital, int, NULL, S_IRUGO); +module_param_array(cfg, long, NULL, S_IRUGO); +module_param_array(reset, int, 0, S_IRUGO); +module_param_array(mpu_io, long, NULL, S_IRUGO); +module_param_array(mpu_irq, int, NULL, S_IRUGO); +module_param_array(ide_io0, long, NULL, S_IRUGO); +module_param_array(ide_io1, long, NULL, S_IRUGO); +module_param_array(ide_irq, int, NULL, S_IRUGO); +module_param_array(joystick_io, long, NULL, S_IRUGO); +#endif + + +static int __devinit snd_msnd_isa_match(struct device *pdev, unsigned int i) +{ + if (io[i] == SNDRV_AUTO_PORT) + return 0; + + if (irq[i] == SNDRV_AUTO_PORT || mem[i] == SNDRV_AUTO_PORT) { + printk(KERN_WARNING LOGNAME ": io, irq and mem must be set\n"); + return 0; + } + +#ifdef MSND_CLASSIC + if (!(io[i] == 0x290 || + io[i] == 0x260 || + io[i] == 0x250 || + io[i] == 0x240 || + io[i] == 0x230 || + io[i] == 0x220 || + io[i] == 0x210 || + io[i] == 0x3e0)) { + printk(KERN_ERR LOGNAME ": \"io\" - DSP I/O base must be set " + " to 0x210, 0x220, 0x230, 0x240, 0x250, 0x260, 0x290, " + "or 0x3E0\n"); + return 0; + } +#else + if (io[i] < 0x100 || io[i] > 0x3e0 || (io[i] % 0x10) != 0) { + printk(KERN_ERR LOGNAME + ": \"io\" - DSP I/O base must within the range 0x100 " + "to 0x3E0 and must be evenly divisible by 0x10\n"); + return 0; + } +#endif /* MSND_CLASSIC */ + + if (!(irq[i] == 5 || + irq[i] == 7 || + irq[i] == 9 || + irq[i] == 10 || + irq[i] == 11 || + irq[i] == 12)) { + printk(KERN_ERR LOGNAME + ": \"irq\" - must be set to 5, 7, 9, 10, 11 or 12\n"); + return 0; + } + + if (!(mem[i] == 0xb0000 || + mem[i] == 0xc8000 || + mem[i] == 0xd0000 || + mem[i] == 0xd8000 || + mem[i] == 0xe0000 || + mem[i] == 0xe8000)) { + printk(KERN_ERR LOGNAME ": \"mem\" - must be set to " + "0xb0000, 0xc8000, 0xd0000, 0xd8000, 0xe0000 or " + "0xe8000\n"); + return 0; + } + +#ifndef MSND_CLASSIC + if (cfg[i] == SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + } else if (cfg[i] != 0x250 && cfg[i] != 0x260 && cfg[i] != 0x270) { + printk(KERN_INFO LOGNAME + ": Config port must be 0x250, 0x260 or 0x270 " + "(or unspecified for PnP mode)\n"); + return 0; + } +#endif /* MSND_CLASSIC */ + + return 1; +} + +static int __devinit snd_msnd_isa_probe(struct device *pdev, unsigned int idx) +{ + int err; + struct snd_card *card; + struct snd_msnd *chip; + + if (has_isapnp(idx) || cfg[idx] == SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME ": Assuming PnP mode\n"); + return -ENODEV; + } + + err = snd_card_create(index[idx], id[idx], THIS_MODULE, + sizeof(struct snd_msnd), &card); + if (err < 0) + return err; + + snd_card_set_dev(card, pdev); + chip = card->private_data; + chip->card = card; + +#ifdef MSND_CLASSIC + switch (irq[idx]) { + case 5: + chip->irqid = HPIRQ_5; break; + case 7: + chip->irqid = HPIRQ_7; break; + case 9: + chip->irqid = HPIRQ_9; break; + case 10: + chip->irqid = HPIRQ_10; break; + case 11: + chip->irqid = HPIRQ_11; break; + case 12: + chip->irqid = HPIRQ_12; break; + } + + switch (mem[idx]) { + case 0xb0000: + chip->memid = HPMEM_B000; break; + case 0xc8000: + chip->memid = HPMEM_C800; break; + case 0xd0000: + chip->memid = HPMEM_D000; break; + case 0xd8000: + chip->memid = HPMEM_D800; break; + case 0xe0000: + chip->memid = HPMEM_E000; break; + case 0xe8000: + chip->memid = HPMEM_E800; break; + } +#else + printk(KERN_INFO LOGNAME ": Non-PnP mode: configuring at port 0x%lx\n", + cfg[idx]); + + if (!request_region(cfg[idx], 2, "Pinnacle/Fiji Config")) { + printk(KERN_ERR LOGNAME ": Config port 0x%lx conflict\n", + cfg[idx]); + snd_card_free(card); + return -EIO; + } + if (reset[idx]) + if (snd_msnd_pinnacle_cfg_reset(cfg[idx])) { + err = -EIO; + goto cfg_error; + } + + /* DSP */ + err = snd_msnd_write_cfg_logical(cfg[idx], 0, + io[idx], 0, + irq[idx], mem[idx]); + + if (err) + goto cfg_error; + + /* The following are Pinnacle specific */ + + /* MPU */ + if (mpu_io[idx] != SNDRV_AUTO_PORT + && mpu_irq[idx] != SNDRV_AUTO_IRQ) { + printk(KERN_INFO LOGNAME + ": Configuring MPU to I/O 0x%lx IRQ %d\n", + mpu_io[idx], mpu_irq[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 1, + mpu_io[idx], 0, + mpu_irq[idx], 0); + + if (err) + goto cfg_error; + } + + /* IDE */ + if (ide_io0[idx] != SNDRV_AUTO_PORT + && ide_io1[idx] != SNDRV_AUTO_PORT + && ide_irq[idx] != SNDRV_AUTO_IRQ) { + printk(KERN_INFO LOGNAME + ": Configuring IDE to I/O 0x%lx, 0x%lx IRQ %d\n", + ide_io0[idx], ide_io1[idx], ide_irq[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 2, + ide_io0[idx], ide_io1[idx], + ide_irq[idx], 0); + + if (err) + goto cfg_error; + } + + /* Joystick */ + if (joystick_io[idx] != SNDRV_AUTO_PORT) { + printk(KERN_INFO LOGNAME + ": Configuring joystick to I/O 0x%lx\n", + joystick_io[idx]); + err = snd_msnd_write_cfg_logical(cfg[idx], 3, + joystick_io[idx], 0, + 0, 0); + + if (err) + goto cfg_error; + } + release_region(cfg[idx], 2); + +#endif /* MSND_CLASSIC */ + + set_default_audio_parameters(chip); +#ifdef MSND_CLASSIC + chip->type = msndClassic; +#else + chip->type = msndPinnacle; +#endif + chip->io = io[idx]; + chip->irq = irq[idx]; + chip->base = mem[idx]; + + chip->calibrate_signal = calibrate_signal ? 1 : 0; + chip->recsrc = 0; + chip->dspq_data_buff = DSPQ_DATA_BUFF; + chip->dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay[idx]) + clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); +#ifndef MSND_CLASSIC + if (digital[idx]) + set_bit(F_HAVEDIGITAL, &chip->flags); +#endif + spin_lock_init(&chip->lock); + err = snd_msnd_probe(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + snd_card_free(card); + return err; + } + + err = snd_msnd_attach(card); + if (err < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + snd_card_free(card); + return err; + } + dev_set_drvdata(pdev, card); + + return 0; + +#ifndef MSND_CLASSIC +cfg_error: + release_region(cfg[idx], 2); + snd_card_free(card); + return err; +#endif +} + +static int __devexit snd_msnd_isa_remove(struct device *pdev, unsigned int dev) +{ + snd_msnd_unload(dev_get_drvdata(pdev)); + dev_set_drvdata(pdev, NULL); + return 0; +} + +#define DEV_NAME "msnd-pinnacle" + +static struct isa_driver snd_msnd_driver = { + .match = snd_msnd_isa_match, + .probe = snd_msnd_isa_probe, + .remove = __devexit_p(snd_msnd_isa_remove), + /* FIXME: suspend, resume */ + .driver = { + .name = DEV_NAME + }, +}; + +#ifdef CONFIG_PNP +static int __devinit snd_msnd_pnp_detect(struct pnp_card_link *pcard, + const struct pnp_card_device_id *pid) +{ + static int idx; + struct pnp_dev *pnp_dev; + struct pnp_dev *mpu_dev; + struct snd_card *card; + struct snd_msnd *chip; + int ret; + + for ( ; idx < SNDRV_CARDS; idx++) { + if (has_isapnp(idx)) + break; + } + if (idx >= SNDRV_CARDS) + return -ENODEV; + + /* + * Check that we still have room for another sound card ... + */ + pnp_dev = pnp_request_card_device(pcard, pid->devs[0].id, NULL); + if (!pnp_dev) + return -ENODEV; + + mpu_dev = pnp_request_card_device(pcard, pid->devs[1].id, NULL); + if (!mpu_dev) + return -ENODEV; + + if (!pnp_is_active(pnp_dev) && pnp_activate_dev(pnp_dev) < 0) { + printk(KERN_INFO "msnd_pinnacle: device is inactive\n"); + return -EBUSY; + } + + if (!pnp_is_active(mpu_dev) && pnp_activate_dev(mpu_dev) < 0) { + printk(KERN_INFO "msnd_pinnacle: MPU device is inactive\n"); + return -EBUSY; + } + + /* + * Create a new ALSA sound card entry, in anticipation + * of detecting our hardware ... + */ + ret = snd_card_create(index[idx], id[idx], THIS_MODULE, + sizeof(struct snd_msnd), &card); + if (ret < 0) + return ret; + + chip = card->private_data; + chip->card = card; + snd_card_set_dev(card, &pcard->card->dev); + + /* + * Read the correct parameters off the ISA PnP bus ... + */ + io[idx] = pnp_port_start(pnp_dev, 0); + irq[idx] = pnp_irq(pnp_dev, 0); + mem[idx] = pnp_mem_start(pnp_dev, 0); + mpu_io[idx] = pnp_port_start(mpu_dev, 0); + mpu_irq[idx] = pnp_irq(mpu_dev, 0); + + set_default_audio_parameters(chip); +#ifdef MSND_CLASSIC + chip->type = msndClassic; +#else + chip->type = msndPinnacle; +#endif + chip->io = io[idx]; + chip->irq = irq[idx]; + chip->base = mem[idx]; + + chip->calibrate_signal = calibrate_signal ? 1 : 0; + chip->recsrc = 0; + chip->dspq_data_buff = DSPQ_DATA_BUFF; + chip->dspq_buff_size = DSPQ_BUFF_SIZE; + if (write_ndelay[idx]) + clear_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); + else + set_bit(F_DISABLE_WRITE_NDELAY, &chip->flags); +#ifndef MSND_CLASSIC + if (digital[idx]) + set_bit(F_HAVEDIGITAL, &chip->flags); +#endif + spin_lock_init(&chip->lock); + ret = snd_msnd_probe(card); + if (ret < 0) { + printk(KERN_ERR LOGNAME ": Probe failed\n"); + goto _release_card; + } + + ret = snd_msnd_attach(card); + if (ret < 0) { + printk(KERN_ERR LOGNAME ": Attach failed\n"); + goto _release_card; + } + + pnp_set_card_drvdata(pcard, card); + ++idx; + return 0; + +_release_card: + snd_card_free(card); + return ret; +} + +static void __devexit snd_msnd_pnp_remove(struct pnp_card_link *pcard) +{ + snd_msnd_unload(pnp_get_card_drvdata(pcard)); + pnp_set_card_drvdata(pcard, NULL); +} + +static int isa_registered; +static int pnp_registered; + +static struct pnp_card_device_id msnd_pnpids[] = { + /* Pinnacle PnP */ + { .id = "BVJ0440", .devs = { { "TBS0000" }, { "TBS0001" } } }, + { .id = "" } /* end */ +}; + +MODULE_DEVICE_TABLE(pnp_card, msnd_pnpids); + +static struct pnp_card_driver msnd_pnpc_driver = { + .flags = PNP_DRIVER_RES_DO_NOT_CHANGE, + .name = "msnd_pinnacle", + .id_table = msnd_pnpids, + .probe = snd_msnd_pnp_detect, + .remove = __devexit_p(snd_msnd_pnp_remove), +}; +#endif /* CONFIG_PNP */ + +static int __init snd_msnd_init(void) +{ + int err; + + err = isa_register_driver(&snd_msnd_driver, SNDRV_CARDS); +#ifdef CONFIG_PNP + if (!err) + isa_registered = 1; + + err = pnp_register_card_driver(&msnd_pnpc_driver); + if (!err) + pnp_registered = 1; + + if (isa_registered) + err = 0; +#endif + return err; +} + +static void __exit snd_msnd_exit(void) +{ +#ifdef CONFIG_PNP + if (pnp_registered) + pnp_unregister_card_driver(&msnd_pnpc_driver); + if (isa_registered) +#endif + isa_unregister_driver(&snd_msnd_driver); +} + +module_init(snd_msnd_init); +module_exit(snd_msnd_exit); + diff --git a/sound/isa/msnd/msnd_pinnacle.h b/sound/isa/msnd/msnd_pinnacle.h new file mode 100644 index 0000000..48318d1 --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle.h @@ -0,0 +1,181 @@ +/********************************************************************* + * + * msnd_pinnacle.h + * + * Turtle Beach MultiSound Sound Card Driver for Linux + * + * Some parts of this header file were derived from the Turtle Beach + * MultiSound Driver Development Kit. + * + * Copyright (C) 1998 Andrew Veliath + * Copyright (C) 1993 Turtle Beach Systems, Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + * + ********************************************************************/ +#ifndef __MSND_PINNACLE_H +#define __MSND_PINNACLE_H + +#define DSP_NUMIO 0x08 + +#define IREG_LOGDEVICE 0x07 +#define IREG_ACTIVATE 0x30 +#define LD_ACTIVATE 0x01 +#define LD_DISACTIVATE 0x00 +#define IREG_EECONTROL 0x3F +#define IREG_MEMBASEHI 0x40 +#define IREG_MEMBASELO 0x41 +#define IREG_MEMCONTROL 0x42 +#define IREG_MEMRANGEHI 0x43 +#define IREG_MEMRANGELO 0x44 +#define MEMTYPE_8BIT 0x00 +#define MEMTYPE_16BIT 0x02 +#define MEMTYPE_RANGE 0x00 +#define MEMTYPE_HIADDR 0x01 +#define IREG_IO0_BASEHI 0x60 +#define IREG_IO0_BASELO 0x61 +#define IREG_IO1_BASEHI 0x62 +#define IREG_IO1_BASELO 0x63 +#define IREG_IRQ_NUMBER 0x70 +#define IREG_IRQ_TYPE 0x71 +#define IRQTYPE_HIGH 0x02 +#define IRQTYPE_LOW 0x00 +#define IRQTYPE_LEVEL 0x01 +#define IRQTYPE_EDGE 0x00 + +#define HP_DSPR 0x04 +#define HP_BLKS 0x04 + +#define HPDSPRESET_OFF 2 +#define HPDSPRESET_ON 0 + +#define HPBLKSEL_0 2 +#define HPBLKSEL_1 3 + +#define HIMT_DAT_OFF 0x03 + +#define HIDSP_PLAY_UNDER 0x00 +#define HIDSP_INT_PLAY_UNDER 0x01 +#define HIDSP_SSI_TX_UNDER 0x02 +#define HIDSP_RECQ_OVERFLOW 0x08 +#define HIDSP_INT_RECORD_OVER 0x09 +#define HIDSP_SSI_RX_OVERFLOW 0x0a + +#define HIDSP_MIDI_IN_OVER 0x10 + +#define HIDSP_MIDI_FRAME_ERR 0x11 +#define HIDSP_MIDI_PARITY_ERR 0x12 +#define HIDSP_MIDI_OVERRUN_ERR 0x13 + +#define HIDSP_INPUT_CLIPPING 0x20 +#define HIDSP_MIX_CLIPPING 0x30 +#define HIDSP_DAT_IN_OFF 0x21 + +#define TIME_PRO_RESET_DONE 0x028A +#define TIME_PRO_SYSEX 0x001E +#define TIME_PRO_RESET 0x0032 + +#define DAR_BUFF_SIZE 0x1000 + +#define MIDQ_BUFF_SIZE 0x800 +#define DSPQ_BUFF_SIZE 0x5A0 + +#define DSPQ_DATA_BUFF 0x7860 + +#define MOP_WAVEHDR 0 +#define MOP_EXTOUT 1 +#define MOP_HWINIT 0xfe +#define MOP_NONE 0xff +#define MOP_MAX 1 + +#define MIP_EXTIN 0 +#define MIP_WAVEHDR 1 +#define MIP_HWINIT 0xfe +#define MIP_MAX 1 + +/* Pinnacle/Fiji SMA Common Data */ +#define SMA_wCurrPlayBytes 0x0000 +#define SMA_wCurrRecordBytes 0x0002 +#define SMA_wCurrPlayVolLeft 0x0004 +#define SMA_wCurrPlayVolRight 0x0006 +#define SMA_wCurrInVolLeft 0x0008 +#define SMA_wCurrInVolRight 0x000a +#define SMA_wCurrMHdrVolLeft 0x000c +#define SMA_wCurrMHdrVolRight 0x000e +#define SMA_dwCurrPlayPitch 0x0010 +#define SMA_dwCurrPlayRate 0x0014 +#define SMA_wCurrMIDIIOPatch 0x0018 +#define SMA_wCurrPlayFormat 0x001a +#define SMA_wCurrPlaySampleSize 0x001c +#define SMA_wCurrPlayChannels 0x001e +#define SMA_wCurrPlaySampleRate 0x0020 +#define SMA_wCurrRecordFormat 0x0022 +#define SMA_wCurrRecordSampleSize 0x0024 +#define SMA_wCurrRecordChannels 0x0026 +#define SMA_wCurrRecordSampleRate 0x0028 +#define SMA_wCurrDSPStatusFlags 0x002a +#define SMA_wCurrHostStatusFlags 0x002c +#define SMA_wCurrInputTagBits 0x002e +#define SMA_wCurrLeftPeak 0x0030 +#define SMA_wCurrRightPeak 0x0032 +#define SMA_bMicPotPosLeft 0x0034 +#define SMA_bMicPotPosRight 0x0035 +#define SMA_bMicPotMaxLeft 0x0036 +#define SMA_bMicPotMaxRight 0x0037 +#define SMA_bInPotPosLeft 0x0038 +#define SMA_bInPotPosRight 0x0039 +#define SMA_bAuxPotPosLeft 0x003a +#define SMA_bAuxPotPosRight 0x003b +#define SMA_bInPotMaxLeft 0x003c +#define SMA_bInPotMaxRight 0x003d +#define SMA_bAuxPotMaxLeft 0x003e +#define SMA_bAuxPotMaxRight 0x003f +#define SMA_bInPotMaxMethod 0x0040 +#define SMA_bAuxPotMaxMethod 0x0041 +#define SMA_wCurrMastVolLeft 0x0042 +#define SMA_wCurrMastVolRight 0x0044 +#define SMA_wCalFreqAtoD 0x0046 +#define SMA_wCurrAuxVolLeft 0x0048 +#define SMA_wCurrAuxVolRight 0x004a +#define SMA_wCurrPlay1VolLeft 0x004c +#define SMA_wCurrPlay1VolRight 0x004e +#define SMA_wCurrPlay2VolLeft 0x0050 +#define SMA_wCurrPlay2VolRight 0x0052 +#define SMA_wCurrPlay3VolLeft 0x0054 +#define SMA_wCurrPlay3VolRight 0x0056 +#define SMA_wCurrPlay4VolLeft 0x0058 +#define SMA_wCurrPlay4VolRight 0x005a +#define SMA_wCurrPlay1PeakLeft 0x005c +#define SMA_wCurrPlay1PeakRight 0x005e +#define SMA_wCurrPlay2PeakLeft 0x0060 +#define SMA_wCurrPlay2PeakRight 0x0062 +#define SMA_wCurrPlay3PeakLeft 0x0064 +#define SMA_wCurrPlay3PeakRight 0x0066 +#define SMA_wCurrPlay4PeakLeft 0x0068 +#define SMA_wCurrPlay4PeakRight 0x006a +#define SMA_wCurrPlayPeakLeft 0x006c +#define SMA_wCurrPlayPeakRight 0x006e +#define SMA_wCurrDATSR 0x0070 +#define SMA_wCurrDATRXCHNL 0x0072 +#define SMA_wCurrDATTXCHNL 0x0074 +#define SMA_wCurrDATRXRate 0x0076 +#define SMA_dwDSPPlayCount 0x0078 +#define SMA__size 0x007c + +#define INITCODEFILE "turtlebeach/pndspini.bin" +#define PERMCODEFILE "turtlebeach/pndsperm.bin" +#define LONGNAME "MultiSound (Pinnacle/Fiji)" + +#endif /* __MSND_PINNACLE_H */ diff --git a/sound/isa/msnd/msnd_pinnacle_mixer.c b/sound/isa/msnd/msnd_pinnacle_mixer.c new file mode 100644 index 0000000..494058a --- /dev/null +++ b/sound/isa/msnd/msnd_pinnacle_mixer.c @@ -0,0 +1,343 @@ +/*************************************************************************** + msnd_pinnacle_mixer.c - description + ------------------- + begin : Fre Jun 7 2002 + copyright : (C) 2002 by karsten wiese + email : annabellesgarden@yahoo.de + ***************************************************************************/ + +/*************************************************************************** + * * + * This program is free software; you can redistribute it and/or modify * + * it under the terms of the GNU General Public License as published by * + * the Free Software Foundation; either version 2 of the License, or * + * (at your option) any later version. * + * * + ***************************************************************************/ + +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/control.h> +#include "msnd.h" +#include "msnd_pinnacle.h" + + +#define MSND_MIXER_VOLUME 0 +#define MSND_MIXER_PCM 1 +#define MSND_MIXER_AUX 2 /* Input source 1 (aux1) */ +#define MSND_MIXER_IMIX 3 /* Recording monitor */ +#define MSND_MIXER_SYNTH 4 +#define MSND_MIXER_SPEAKER 5 +#define MSND_MIXER_LINE 6 +#define MSND_MIXER_MIC 7 +#define MSND_MIXER_RECLEV 11 /* Recording level */ +#define MSND_MIXER_IGAIN 12 /* Input gain */ +#define MSND_MIXER_OGAIN 13 /* Output gain */ +#define MSND_MIXER_DIGITAL 17 /* Digital (input) 1 */ + +/* Device mask bits */ + +#define MSND_MASK_VOLUME (1 << MSND_MIXER_VOLUME) +#define MSND_MASK_SYNTH (1 << MSND_MIXER_SYNTH) +#define MSND_MASK_PCM (1 << MSND_MIXER_PCM) +#define MSND_MASK_SPEAKER (1 << MSND_MIXER_SPEAKER) +#define MSND_MASK_LINE (1 << MSND_MIXER_LINE) +#define MSND_MASK_MIC (1 << MSND_MIXER_MIC) +#define MSND_MASK_IMIX (1 << MSND_MIXER_IMIX) +#define MSND_MASK_RECLEV (1 << MSND_MIXER_RECLEV) +#define MSND_MASK_IGAIN (1 << MSND_MIXER_IGAIN) +#define MSND_MASK_OGAIN (1 << MSND_MIXER_OGAIN) +#define MSND_MASK_AUX (1 << MSND_MIXER_AUX) +#define MSND_MASK_DIGITAL (1 << MSND_MIXER_DIGITAL) + +static int snd_msndmix_info_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static char *texts[3] = { + "Analog", "MASS", "SPDIF", + }; + struct snd_msnd *chip = snd_kcontrol_chip(kcontrol); + unsigned items = test_bit(F_HAVEDIGITAL, &chip->flags) ? 3 : 2; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = items; + if (uinfo->value.enumerated.item >= items) + uinfo->value.enumerated.item = items - 1; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_msndmix_get_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *chip = snd_kcontrol_chip(kcontrol); + /* MSND_MASK_IMIX is the default */ + ucontrol->value.enumerated.item[0] = 0; + + if (chip->recsrc & MSND_MASK_SYNTH) { + ucontrol->value.enumerated.item[0] = 1; + } else if ((chip->recsrc & MSND_MASK_DIGITAL) && + test_bit(F_HAVEDIGITAL, &chip->flags)) { + ucontrol->value.enumerated.item[0] = 2; + } + + + return 0; +} + +static int snd_msndmix_set_mux(struct snd_msnd *chip, int val) +{ + unsigned newrecsrc; + int change; + unsigned char msndbyte; + + switch (val) { + case 0: + newrecsrc = MSND_MASK_IMIX; + msndbyte = HDEXAR_SET_ANA_IN; + break; + case 1: + newrecsrc = MSND_MASK_SYNTH; + msndbyte = HDEXAR_SET_SYNTH_IN; + break; + case 2: + newrecsrc = MSND_MASK_DIGITAL; + msndbyte = HDEXAR_SET_DAT_IN; + break; + default: + return -EINVAL; + } + change = newrecsrc != chip->recsrc; + if (change) { + change = 0; + if (!snd_msnd_send_word(chip, 0, 0, msndbyte)) + if (!snd_msnd_send_dsp_cmd(chip, HDEX_AUX_REQ)) { + chip->recsrc = newrecsrc; + change = 1; + } + } + return change; +} + +static int snd_msndmix_put_mux(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + return snd_msndmix_set_mux(msnd, ucontrol->value.enumerated.item[0]); +} + + +static int snd_msndmix_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 100; + return 0; +} + +static int snd_msndmix_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + int addr = kcontrol->private_value; + unsigned long flags; + + spin_lock_irqsave(&msnd->mixer_lock, flags); + ucontrol->value.integer.value[0] = msnd->left_levels[addr] * 100; + ucontrol->value.integer.value[0] /= 0xFFFF; + ucontrol->value.integer.value[1] = msnd->right_levels[addr] * 100; + ucontrol->value.integer.value[1] /= 0xFFFF; + spin_unlock_irqrestore(&msnd->mixer_lock, flags); + return 0; +} + +#define update_volm(a, b) \ + do { \ + writew((dev->left_levels[a] >> 1) * \ + readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev->SMA + SMA_##b##Left); \ + writew((dev->right_levels[a] >> 1) * \ + readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev->SMA + SMA_##b##Right); \ + } while (0); + +#define update_potm(d, s, ar) \ + do { \ + writeb((dev->left_levels[d] >> 8) * \ + readw(dev->SMA + SMA_wCurrMastVolLeft) / 0xffff, \ + dev->SMA + SMA_##s##Left); \ + writeb((dev->right_levels[d] >> 8) * \ + readw(dev->SMA + SMA_wCurrMastVolRight) / 0xffff, \ + dev->SMA + SMA_##s##Right); \ + if (snd_msnd_send_word(dev, 0, 0, ar) == 0) \ + snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ); \ + } while (0); + +#define update_pot(d, s, ar) \ + do { \ + writeb(dev->left_levels[d] >> 8, \ + dev->SMA + SMA_##s##Left); \ + writeb(dev->right_levels[d] >> 8, \ + dev->SMA + SMA_##s##Right); \ + if (snd_msnd_send_word(dev, 0, 0, ar) == 0) \ + snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ); \ + } while (0); + + +static int snd_msndmix_set(struct snd_msnd *dev, int d, int left, int right) +{ + int bLeft, bRight; + int wLeft, wRight; + int updatemaster = 0; + + if (d >= LEVEL_ENTRIES) + return -EINVAL; + + bLeft = left * 0xff / 100; + wLeft = left * 0xffff / 100; + + bRight = right * 0xff / 100; + wRight = right * 0xffff / 100; + + dev->left_levels[d] = wLeft; + dev->right_levels[d] = wRight; + + switch (d) { + /* master volume unscaled controls */ + case MSND_MIXER_LINE: /* line pot control */ + /* scaled by IMIX in digital mix */ + writeb(bLeft, dev->SMA + SMA_bInPotPosLeft); + writeb(bRight, dev->SMA + SMA_bInPotPosRight); + if (snd_msnd_send_word(dev, 0, 0, HDEXAR_IN_SET_POTS) == 0) + snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ); + break; + case MSND_MIXER_MIC: /* mic pot control */ + if (dev->type == msndClassic) + return -EINVAL; + /* scaled by IMIX in digital mix */ + writeb(bLeft, dev->SMA + SMA_bMicPotPosLeft); + writeb(bRight, dev->SMA + SMA_bMicPotPosRight); + if (snd_msnd_send_word(dev, 0, 0, HDEXAR_MIC_SET_POTS) == 0) + snd_msnd_send_dsp_cmd(dev, HDEX_AUX_REQ); + break; + case MSND_MIXER_VOLUME: /* master volume */ + writew(wLeft, dev->SMA + SMA_wCurrMastVolLeft); + writew(wRight, dev->SMA + SMA_wCurrMastVolRight); + /* fall through */ + + case MSND_MIXER_AUX: /* aux pot control */ + /* scaled by master volume */ + /* fall through */ + + /* digital controls */ + case MSND_MIXER_SYNTH: /* synth vol (dsp mix) */ + case MSND_MIXER_PCM: /* pcm vol (dsp mix) */ + case MSND_MIXER_IMIX: /* input monitor (dsp mix) */ + /* scaled by master volume */ + updatemaster = 1; + break; + + default: + return -EINVAL; + } + + if (updatemaster) { + /* update master volume scaled controls */ + update_volm(MSND_MIXER_PCM, wCurrPlayVol); + update_volm(MSND_MIXER_IMIX, wCurrInVol); + if (dev->type == msndPinnacle) + update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol); + update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); + } + + return 0; +} + +static int snd_msndmix_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_msnd *msnd = snd_kcontrol_chip(kcontrol); + int change, addr = kcontrol->private_value; + int left, right; + unsigned long flags; + + left = ucontrol->value.integer.value[0] % 101; + right = ucontrol->value.integer.value[1] % 101; + spin_lock_irqsave(&msnd->mixer_lock, flags); + change = msnd->left_levels[addr] != left + || msnd->right_levels[addr] != right; + snd_msndmix_set(msnd, addr, left, right); + spin_unlock_irqrestore(&msnd->mixer_lock, flags); + return change; +} + + +#define DUMMY_VOLUME(xname, xindex, addr) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ + .info = snd_msndmix_volume_info, \ + .get = snd_msndmix_volume_get, .put = snd_msndmix_volume_put, \ + .private_value = addr } + + +static struct snd_kcontrol_new snd_msnd_controls[] = { +DUMMY_VOLUME("Master Volume", 0, MSND_MIXER_VOLUME), +DUMMY_VOLUME("PCM Volume", 0, MSND_MIXER_PCM), +DUMMY_VOLUME("Aux Volume", 0, MSND_MIXER_AUX), +DUMMY_VOLUME("Line Volume", 0, MSND_MIXER_LINE), +DUMMY_VOLUME("Mic Volume", 0, MSND_MIXER_MIC), +DUMMY_VOLUME("Monitor", 0, MSND_MIXER_IMIX), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_msndmix_info_mux, + .get = snd_msndmix_get_mux, + .put = snd_msndmix_put_mux, +} +}; + + +int __devinit snd_msndmix_new(struct snd_card *card) +{ + struct snd_msnd *chip = card->private_data; + unsigned int idx; + int err; + + if (snd_BUG_ON(!chip)) + return -EINVAL; + spin_lock_init(&chip->mixer_lock); + strcpy(card->mixername, "MSND Pinnacle Mixer"); + + for (idx = 0; idx < ARRAY_SIZE(snd_msnd_controls); idx++) + err = snd_ctl_add(card, + snd_ctl_new1(snd_msnd_controls + idx, chip)); + if (err < 0) + return err; + + return 0; +} +EXPORT_SYMBOL(snd_msndmix_new); + +void snd_msndmix_setup(struct snd_msnd *dev) +{ + update_pot(MSND_MIXER_LINE, bInPotPos, HDEXAR_IN_SET_POTS); + update_potm(MSND_MIXER_AUX, bAuxPotPos, HDEXAR_AUX_SET_POTS); + update_volm(MSND_MIXER_PCM, wCurrPlayVol); + update_volm(MSND_MIXER_IMIX, wCurrInVol); + if (dev->type == msndPinnacle) { + update_pot(MSND_MIXER_MIC, bMicPotPos, HDEXAR_MIC_SET_POTS); + update_volm(MSND_MIXER_SYNTH, wCurrMHdrVol); + } +} +EXPORT_SYMBOL(snd_msndmix_setup); + +int snd_msndmix_force_recsrc(struct snd_msnd *dev, int recsrc) +{ + dev->recsrc = -1; + return snd_msndmix_set_mux(dev, recsrc); +} +EXPORT_SYMBOL(snd_msndmix_force_recsrc); diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 645491a..0481a55 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -179,12 +179,13 @@ static unsigned char __snd_opl3sa2_read(struct snd_opl3sa2 *chip, unsigned char unsigned char result; #if 0 outb(0x1d, port); /* password */ - printk("read [0x%lx] = 0x%x\n", port, inb(port)); + printk(KERN_DEBUG "read [0x%lx] = 0x%x\n", port, inb(port)); #endif outb(reg, chip->port); /* register */ result = inb(chip->port + 1); #if 0 - printk("read [0x%lx] = 0x%x [0x%x]\n", port, result, inb(port)); + printk(KERN_DEBUG "read [0x%lx] = 0x%x [0x%x]\n", + port, result, inb(port)); #endif return result; } @@ -233,7 +234,10 @@ static int __devinit snd_opl3sa2_detect(struct snd_card *card) snd_printk(KERN_ERR PFX "can't grab port 0x%lx\n", port); return -EBUSY; } - // snd_printk("REG 0A = 0x%x\n", snd_opl3sa2_read(chip, 0x0a)); + /* + snd_printk(KERN_DEBUG "REG 0A = 0x%x\n", + snd_opl3sa2_read(chip, 0x0a)); + */ chip->version = 0; tmp = snd_opl3sa2_read(chip, OPL3SA2_MISC); if (tmp == 0xff) { @@ -477,6 +481,7 @@ OPL3SA2_DOUBLE_TLV("Master Playback Volume", 0, 0x07, 0x08, 0, 0, 15, 1, OPL3SA2_SINGLE("Mic Playback Switch", 0, 0x09, 7, 1, 1), OPL3SA2_SINGLE_TLV("Mic Playback Volume", 0, 0x09, 0, 31, 1, db_scale_5bit_12db_max), +OPL3SA2_SINGLE("ZV Port Switch", 0, 0x02, 0, 1, 0), }; static struct snd_kcontrol_new snd_opl3sa2_tone_controls[] = { @@ -550,21 +555,27 @@ static int __devinit snd_opl3sa2_mixer(struct snd_card *card) #ifdef CONFIG_PM static int snd_opl3sa2_suspend(struct snd_card *card, pm_message_t state) { - struct snd_opl3sa2 *chip = card->private_data; + if (card) { + struct snd_opl3sa2 *chip = card->private_data; - snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); - chip->wss->suspend(chip->wss); - /* power down */ - snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3); + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + chip->wss->suspend(chip->wss); + /* power down */ + snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D3); + } return 0; } static int snd_opl3sa2_resume(struct snd_card *card) { - struct snd_opl3sa2 *chip = card->private_data; + struct snd_opl3sa2 *chip; int i; + if (!card) + return 0; + + chip = card->private_data; /* power up */ snd_opl3sa2_write(chip, OPL3SA2_PM_CTRL, OPL3SA2_PM_D0); @@ -613,7 +624,7 @@ static void snd_opl3sa2_free(struct snd_card *card) { struct snd_opl3sa2 *chip = card->private_data; if (chip->irq >= 0) - free_irq(chip->irq, (void *)chip); + free_irq(chip->irq, card); release_and_free_resource(chip->res_port); } @@ -628,7 +639,7 @@ static int snd_opl3sa2_card_new(int dev, struct snd_card **cardp) if (err < 0) return err; strcpy(card->driver, "OPL3SA2"); - strcpy(card->shortname, "Yamaha OPL3-SA2"); + strcpy(card->shortname, "Yamaha OPL3-SA"); chip = card->private_data; spin_lock_init(&chip->reg_lock); chip->irq = -1; diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index cd6e60a..5cd5553 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -252,7 +252,7 @@ static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", hardware); + snd_printk(KERN_ERR "chip %d not supported\n", hardware); return -ENODEV; } return 0; @@ -294,7 +294,7 @@ static unsigned char snd_opti9xx_read(struct snd_opti9xx *chip, #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", chip->hardware); + snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); } spin_unlock_irqrestore(&chip->lock, flags); @@ -336,7 +336,7 @@ static void snd_opti9xx_write(struct snd_opti9xx *chip, unsigned char reg, #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", chip->hardware); + snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); } spin_unlock_irqrestore(&chip->lock, flags); @@ -412,7 +412,7 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) #endif /* OPTi93X */ default: - snd_printk("chip %d not supported\n", chip->hardware); + snd_printk(KERN_ERR "chip %d not supported\n", chip->hardware); return -EINVAL; } @@ -430,7 +430,8 @@ static int __devinit snd_opti9xx_configure(struct snd_opti9xx *chip) wss_base_bits = 0x02; break; default: - snd_printk("WSS port 0x%lx not valid\n", chip->wss_base); + snd_printk(KERN_WARNING "WSS port 0x%lx not valid\n", + chip->wss_base); goto __skip_base; } snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(1), wss_base_bits << 4, 0x30); @@ -455,7 +456,7 @@ __skip_base: irq_bits = 0x04; break; default: - snd_printk("WSS irq # %d not valid\n", chip->irq); + snd_printk(KERN_WARNING "WSS irq # %d not valid\n", chip->irq); goto __skip_resources; } @@ -470,13 +471,14 @@ __skip_base: dma_bits = 0x03; break; default: - snd_printk("WSS dma1 # %d not valid\n", chip->dma1); + snd_printk(KERN_WARNING "WSS dma1 # %d not valid\n", + chip->dma1); goto __skip_resources; } #if defined(CS4231) || defined(OPTi93X) if (chip->dma1 == chip->dma2) { - snd_printk("don't want to share dmas\n"); + snd_printk(KERN_ERR "don't want to share dmas\n"); return -EBUSY; } @@ -485,7 +487,8 @@ __skip_base: case 1: break; default: - snd_printk("WSS dma2 # %d not valid\n", chip->dma2); + snd_printk(KERN_WARNING "WSS dma2 # %d not valid\n", + chip->dma2); goto __skip_resources; } dma_bits |= 0x04; @@ -516,7 +519,8 @@ __skip_resources: mpu_port_bits = 0x00; break; default: - snd_printk("MPU-401 port 0x%lx not valid\n", + snd_printk(KERN_WARNING + "MPU-401 port 0x%lx not valid\n", chip->mpu_port); goto __skip_mpu; } @@ -535,7 +539,7 @@ __skip_resources: mpu_irq_bits = 0x01; break; default: - snd_printk("MPU-401 irq # %d not valid\n", + snd_printk(KERN_WARNING "MPU-401 irq # %d not valid\n", chip->mpu_irq); goto __skip_mpu; } @@ -726,7 +730,7 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) if (chip->wss_base == SNDRV_AUTO_PORT) { chip->wss_base = snd_legacy_find_free_ioport(possible_ports, 4); if (chip->wss_base < 0) { - snd_printk("unable to find a free WSS port\n"); + snd_printk(KERN_ERR "unable to find a free WSS port\n"); return -EBUSY; } } @@ -815,14 +819,8 @@ static int __devinit snd_opti9xx_probe(struct snd_card *card) chip->fm_port, chip->fm_port + 4 - 1); } if (opl3) { -#ifdef CS4231 - const int t1dev = 1; -#else - const int t1dev = 0; -#endif - if ((error = snd_opl3_timer_new(opl3, t1dev, t1dev+1)) < 0) - return error; - if ((error = snd_opl3_hwdep_new(opl3, 0, 1, &synth)) < 0) + error = snd_opl3_hwdep_new(opl3, 0, 1, &synth); + if (error < 0) return error; } } @@ -900,7 +898,7 @@ static int __devinit snd_opti9xx_isa_probe(struct device *devptr, #if defined(CS4231) || defined(OPTi93X) if (dma2 == SNDRV_AUTO_DMA) { if ((dma2 = snd_legacy_find_free_dma(possible_dma2s[dma1 % 4])) < 0) { - snd_printk("unable to find a free DMA2\n"); + snd_printk(KERN_ERR "unable to find a free DMA2\n"); return -EBUSY; } } diff --git a/sound/isa/sb/sb_mixer.c b/sound/isa/sb/sb_mixer.c index 406a431..475220b 100644 --- a/sound/isa/sb/sb_mixer.c +++ b/sound/isa/sb/sb_mixer.c @@ -182,7 +182,7 @@ static int snd_sbmixer_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int snd_dt019x_input_sw_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[5] = { + static const char *texts[5] = { "CD", "Mic", "Line", "Synth", "Master" }; @@ -269,12 +269,73 @@ static int snd_dt019x_input_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl } /* + * ALS4000 mono recording control switch + */ + +static int snd_als4k_mono_capture_route_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *texts[3] = { + "L chan only", "R chan only", "L ch/2 + R ch/2" + }; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 3; + if (uinfo->value.enumerated.item > 2) + uinfo->value.enumerated.item = 2; + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); + return 0; +} + +static int snd_als4k_mono_capture_route_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sb *sb = snd_kcontrol_chip(kcontrol); + unsigned long flags; + unsigned char oval; + + spin_lock_irqsave(&sb->mixer_lock, flags); + oval = snd_sbmixer_read(sb, SB_ALS4000_MONO_IO_CTRL); + spin_unlock_irqrestore(&sb->mixer_lock, flags); + oval >>= 6; + if (oval > 2) + oval = 2; + + ucontrol->value.enumerated.item[0] = oval; + return 0; +} + +static int snd_als4k_mono_capture_route_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_sb *sb = snd_kcontrol_chip(kcontrol); + unsigned long flags; + int change; + unsigned char nval, oval; + + if (ucontrol->value.enumerated.item[0] > 2) + return -EINVAL; + spin_lock_irqsave(&sb->mixer_lock, flags); + oval = snd_sbmixer_read(sb, SB_ALS4000_MONO_IO_CTRL); + + nval = (oval & ~(3 << 6)) + | (ucontrol->value.enumerated.item[0] << 6); + change = nval != oval; + if (change) + snd_sbmixer_write(sb, SB_ALS4000_MONO_IO_CTRL, nval); + spin_unlock_irqrestore(&sb->mixer_lock, flags); + return change; +} + +/* * SBPRO input multiplexer */ static int snd_sb8mixer_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[3] = { + static const char *texts[3] = { "Mic", "CD", "Line" }; @@ -442,6 +503,12 @@ int snd_sbmixer_add_ctl(struct snd_sb *chip, const char *name, int index, int ty .get = snd_dt019x_input_sw_get, .put = snd_dt019x_input_sw_put, }, + [SB_MIX_MONO_CAPTURE_ALS4K] = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .info = snd_als4k_mono_capture_route_info, + .get = snd_als4k_mono_capture_route_get, + .put = snd_als4k_mono_capture_route_put, + }, }; struct snd_kcontrol *ctl; int err; @@ -636,6 +703,8 @@ static struct sbmix_elem snd_dt019x_ctl_capture_source = }; static struct sbmix_elem *snd_dt019x_controls[] = { + /* ALS4000 below has some parts which we might be lacking, + * e.g. snd_als4000_ctl_mono_playback_switch - check it! */ &snd_dt019x_ctl_master_play_vol, &snd_dt019x_ctl_pcm_play_vol, &snd_dt019x_ctl_synth_play_vol, @@ -666,18 +735,21 @@ static unsigned char snd_dt019x_init_values[][2] = { /* * ALS4000 specific mixer elements */ -/* FIXME: SB_ALS4000_MONO_IO_CTRL needs output select ctrl! */ static struct sbmix_elem snd_als4000_ctl_master_mono_playback_switch = SB_SINGLE("Master Mono Playback Switch", SB_ALS4000_MONO_IO_CTRL, 5, 1); -static struct sbmix_elem snd_als4000_ctl_master_mono_capture_route = - SB_SINGLE("Master Mono Capture Route", SB_ALS4000_MONO_IO_CTRL, 6, 0x03); -/* FIXME: mono playback switch also available on DT019X? */ +static struct sbmix_elem snd_als4k_ctl_master_mono_capture_route = { + .name = "Master Mono Capture Route", + .type = SB_MIX_MONO_CAPTURE_ALS4K + }; static struct sbmix_elem snd_als4000_ctl_mono_playback_switch = SB_SINGLE("Mono Playback Switch", SB_DT019X_OUTPUT_SW2, 0, 1); static struct sbmix_elem snd_als4000_ctl_mic_20db_boost = SB_SINGLE("Mic Boost (+20dB)", SB_ALS4000_MIC_IN_GAIN, 0, 0x03); -static struct sbmix_elem snd_als4000_ctl_mixer_loopback = - SB_SINGLE("Analog Loopback", SB_ALS4000_MIC_IN_GAIN, 7, 0x01); +static struct sbmix_elem snd_als4000_ctl_mixer_analog_loopback = + SB_SINGLE("Analog Loopback Switch", SB_ALS4000_MIC_IN_GAIN, 7, 0x01); +static struct sbmix_elem snd_als4000_ctl_mixer_digital_loopback = + SB_SINGLE("Digital Loopback Switch", + SB_ALS4000_CR3_CONFIGURATION, 7, 0x01); /* FIXME: functionality of 3D controls might be swapped, I didn't find * a description of how to identify what is supposed to be what */ static struct sbmix_elem snd_als4000_3d_control_switch = @@ -694,6 +766,9 @@ static struct sbmix_elem snd_als4000_3d_control_delay = SB_SINGLE("3D Control - Wide", SB_ALS4000_3D_TIME_DELAY, 0, 0x0f); static struct sbmix_elem snd_als4000_3d_control_poweroff_switch = SB_SINGLE("3D PowerOff Switch", SB_ALS4000_3D_TIME_DELAY, 4, 0x01); +static struct sbmix_elem snd_als4000_ctl_3db_freq_control_switch = + SB_SINGLE("Master Playback 8kHz / 20kHz LPF Switch", + SB_ALS4000_FMDAC, 5, 0x01); #ifdef NOT_AVAILABLE static struct sbmix_elem snd_als4000_ctl_fmdac = SB_SINGLE("FMDAC Switch (Option ?)", SB_ALS4000_FMDAC, 0, 0x01); @@ -702,35 +777,37 @@ static struct sbmix_elem snd_als4000_ctl_qsound = #endif static struct sbmix_elem *snd_als4000_controls[] = { - &snd_sb16_ctl_master_play_vol, - &snd_dt019x_ctl_pcm_play_switch, - &snd_sb16_ctl_pcm_play_vol, - &snd_sb16_ctl_synth_capture_route, - &snd_dt019x_ctl_synth_play_switch, - &snd_sb16_ctl_synth_play_vol, - &snd_sb16_ctl_cd_capture_route, - &snd_sb16_ctl_cd_play_switch, - &snd_sb16_ctl_cd_play_vol, - &snd_sb16_ctl_line_capture_route, - &snd_sb16_ctl_line_play_switch, - &snd_sb16_ctl_line_play_vol, - &snd_sb16_ctl_mic_capture_route, - &snd_als4000_ctl_mic_20db_boost, - &snd_sb16_ctl_auto_mic_gain, - &snd_sb16_ctl_mic_play_switch, - &snd_sb16_ctl_mic_play_vol, - &snd_sb16_ctl_pc_speaker_vol, - &snd_sb16_ctl_capture_vol, - &snd_sb16_ctl_play_vol, - &snd_als4000_ctl_master_mono_playback_switch, - &snd_als4000_ctl_master_mono_capture_route, - &snd_als4000_ctl_mono_playback_switch, - &snd_als4000_ctl_mixer_loopback, - &snd_als4000_3d_control_switch, - &snd_als4000_3d_control_ratio, - &snd_als4000_3d_control_freq, - &snd_als4000_3d_control_delay, - &snd_als4000_3d_control_poweroff_switch, + /* ALS4000a.PDF regs page */ + &snd_sb16_ctl_master_play_vol, /* MX30/31 12 */ + &snd_dt019x_ctl_pcm_play_switch, /* MX4C 16 */ + &snd_sb16_ctl_pcm_play_vol, /* MX32/33 12 */ + &snd_sb16_ctl_synth_capture_route, /* MX3D/3E 14 */ + &snd_dt019x_ctl_synth_play_switch, /* MX4C 16 */ + &snd_sb16_ctl_synth_play_vol, /* MX34/35 12/13 */ + &snd_sb16_ctl_cd_capture_route, /* MX3D/3E 14 */ + &snd_sb16_ctl_cd_play_switch, /* MX3C 14 */ + &snd_sb16_ctl_cd_play_vol, /* MX36/37 13 */ + &snd_sb16_ctl_line_capture_route, /* MX3D/3E 14 */ + &snd_sb16_ctl_line_play_switch, /* MX3C 14 */ + &snd_sb16_ctl_line_play_vol, /* MX38/39 13 */ + &snd_sb16_ctl_mic_capture_route, /* MX3D/3E 14 */ + &snd_als4000_ctl_mic_20db_boost, /* MX4D 16 */ + &snd_sb16_ctl_mic_play_switch, /* MX3C 14 */ + &snd_sb16_ctl_mic_play_vol, /* MX3A 13 */ + &snd_sb16_ctl_pc_speaker_vol, /* MX3B 14 */ + &snd_sb16_ctl_capture_vol, /* MX3F/40 15 */ + &snd_sb16_ctl_play_vol, /* MX41/42 15 */ + &snd_als4000_ctl_master_mono_playback_switch, /* MX4C 16 */ + &snd_als4k_ctl_master_mono_capture_route, /* MX4B 16 */ + &snd_als4000_ctl_mono_playback_switch, /* MX4C 16 */ + &snd_als4000_ctl_mixer_analog_loopback, /* MX4D 16 */ + &snd_als4000_ctl_mixer_digital_loopback, /* CR3 21 */ + &snd_als4000_3d_control_switch, /* MX50 17 */ + &snd_als4000_3d_control_ratio, /* MX50 17 */ + &snd_als4000_3d_control_freq, /* MX50 17 */ + &snd_als4000_3d_control_delay, /* MX51 18 */ + &snd_als4000_3d_control_poweroff_switch, /* MX51 18 */ + &snd_als4000_ctl_3db_freq_control_switch, /* MX4F 17 */ #ifdef NOT_AVAILABLE &snd_als4000_ctl_fmdac, &snd_als4000_ctl_qsound, @@ -905,13 +982,14 @@ static unsigned char dt019x_saved_regs[] = { }; static unsigned char als4000_saved_regs[] = { + /* please verify in dsheet whether regs to be added + are actually real H/W or just dummy */ SB_DSP4_MASTER_DEV, SB_DSP4_MASTER_DEV + 1, SB_DSP4_OUTPUT_SW, SB_DSP4_PCM_DEV, SB_DSP4_PCM_DEV + 1, SB_DSP4_INPUT_LEFT, SB_DSP4_INPUT_RIGHT, SB_DSP4_SYNTH_DEV, SB_DSP4_SYNTH_DEV + 1, SB_DSP4_CD_DEV, SB_DSP4_CD_DEV + 1, - SB_DSP4_MIC_AGC, SB_DSP4_MIC_DEV, SB_DSP4_SPEAKER_DEV, SB_DSP4_IGAIN_DEV, SB_DSP4_IGAIN_DEV + 1, @@ -919,8 +997,10 @@ static unsigned char als4000_saved_regs[] = { SB_DT019X_OUTPUT_SW2, SB_ALS4000_MONO_IO_CTRL, SB_ALS4000_MIC_IN_GAIN, + SB_ALS4000_FMDAC, SB_ALS4000_3D_SND_FX, SB_ALS4000_3D_TIME_DELAY, + SB_ALS4000_CR3_CONFIGURATION, }; static void save_mixer(struct snd_sb *chip, unsigned char *regs, int num_regs) diff --git a/sound/isa/sc6000.c b/sound/isa/sc6000.c index 7a14703..7820106 100644 --- a/sound/isa/sc6000.c +++ b/sound/isa/sc6000.c @@ -576,10 +576,6 @@ static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) snd_printk(KERN_ERR PFX "no OPL device at 0x%x-0x%x ?\n", 0x388, 0x388 + 2); } else { - err = snd_opl3_timer_new(opl3, 0, 1); - if (err < 0) - goto err_unmap2; - err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); if (err < 0) goto err_unmap2; diff --git a/sound/isa/sscape.c b/sound/isa/sscape.c index 4025fb5..6618712 100644 --- a/sound/isa/sscape.c +++ b/sound/isa/sscape.c @@ -89,9 +89,6 @@ MODULE_DEVICE_TABLE(pnp_card, sscape_pnpids); #endif -#define MPU401_IO(i) ((i) + 0) -#define MIDI_DATA_IO(i) ((i) + 0) -#define MIDI_CTRL_IO(i) ((i) + 1) #define HOST_CTRL_IO(i) ((i) + 2) #define HOST_DATA_IO(i) ((i) + 3) #define ODIE_ADDR_IO(i) ((i) + 4) @@ -129,9 +126,6 @@ enum GA_REG { #define DMA_8BIT 0x80 -#define AD1845_FREQ_SEL_MSB 0x16 -#define AD1845_FREQ_SEL_LSB 0x17 - enum card_type { SSCAPE, SSCAPE_PNP, @@ -141,8 +135,6 @@ enum card_type { struct soundscape { spinlock_t lock; unsigned io_base; - unsigned wss_base; - int codec_type; int ic_type; enum card_type type; struct resource *io_res; @@ -330,7 +322,7 @@ static int host_write_ctrl_unsafe(unsigned io_base, unsigned char data, */ static inline int verify_mpu401(const struct snd_mpu401 * mpu) { - return ((inb(MIDI_CTRL_IO(mpu->port)) & 0xc0) == 0x80); + return ((inb(MPU401C(mpu)) & 0xc0) == 0x80); } /* @@ -338,7 +330,7 @@ static inline int verify_mpu401(const struct snd_mpu401 * mpu) */ static inline void initialise_mpu401(const struct snd_mpu401 * mpu) { - outb(0, MIDI_DATA_IO(mpu->port)); + outb(0, MPU401D(mpu)); } /* @@ -396,20 +388,20 @@ static int sscape_wait_dma_unsafe(unsigned io_base, enum GA_REG reg, unsigned ti */ static int obp_startup_ack(struct soundscape *s, unsigned timeout) { - while (timeout != 0) { + unsigned long end_time = jiffies + msecs_to_jiffies(timeout); + + do { unsigned long flags; unsigned char x; - schedule_timeout_uninterruptible(1); - spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); spin_unlock_irqrestore(&s->lock, flags); if ((x & 0xfe) == 0xfe) return 1; - --timeout; - } /* while */ + msleep(10); + } while (time_before(jiffies, end_time)); return 0; } @@ -423,20 +415,20 @@ static int obp_startup_ack(struct soundscape *s, unsigned timeout) */ static int host_startup_ack(struct soundscape *s, unsigned timeout) { - while (timeout != 0) { + unsigned long end_time = jiffies + msecs_to_jiffies(timeout); + + do { unsigned long flags; unsigned char x; - schedule_timeout_uninterruptible(1); - spin_lock_irqsave(&s->lock, flags); x = inb(HOST_DATA_IO(s->io_base)); spin_unlock_irqrestore(&s->lock, flags); if (x == 0xfe) return 1; - --timeout; - } /* while */ + msleep(10); + } while (time_before(jiffies, end_time)); return 0; } @@ -532,10 +524,10 @@ static int upload_dma_data(struct soundscape *s, * give it 5 seconds (max) ... */ ret = 0; - if (!obp_startup_ack(s, 5)) { + if (!obp_startup_ack(s, 5000)) { snd_printk(KERN_ERR "sscape: No response from on-board processor after upload\n"); ret = -EAGAIN; - } else if (!host_startup_ack(s, 5)) { + } else if (!host_startup_ack(s, 5000)) { snd_printk(KERN_ERR "sscape: SoundScape failed to initialise\n"); ret = -EAGAIN; } @@ -732,13 +724,7 @@ static int sscape_midi_get(struct snd_kcontrol *kctl, unsigned long flags; spin_lock_irqsave(&s->lock, flags); - set_host_mode_unsafe(s->io_base); - - if (host_write_ctrl_unsafe(s->io_base, CMD_GET_MIDI_VOL, 100)) { - uctl->value.integer.value[0] = host_read_ctrl_unsafe(s->io_base, 100); - } - - set_midi_mode_unsafe(s->io_base); + uctl->value.integer.value[0] = s->midi_vol; spin_unlock_irqrestore(&s->lock, flags); return 0; } @@ -773,6 +759,7 @@ static int sscape_midi_put(struct snd_kcontrol *kctl, change = (host_write_ctrl_unsafe(s->io_base, CMD_SET_MIDI_VOL, 100) && host_write_ctrl_unsafe(s->io_base, ((unsigned char) uctl->value.integer. value[0]) & 127, 100) && host_write_ctrl_unsafe(s->io_base, CMD_XXX_MIDI_VOL, 100)); + s->midi_vol = (unsigned char) uctl->value.integer.value[0] & 127; __skip_change: /* @@ -815,12 +802,11 @@ static unsigned __devinit get_irq_config(int irq) * Perform certain arcane port-checks to see whether there * is a SoundScape board lurking behind the given ports. */ -static int __devinit detect_sscape(struct soundscape *s) +static int __devinit detect_sscape(struct soundscape *s, long wss_io) { unsigned long flags; unsigned d; int retval = 0; - int codec = s->wss_base; spin_lock_irqsave(&s->lock, flags); @@ -836,13 +822,11 @@ static int __devinit detect_sscape(struct soundscape *s) if ((d & 0x80) != 0) goto _done; - if (d == 0) { - s->codec_type = 1; + if (d == 0) s->ic_type = IC_ODIE; - } else if ((d & 0x60) != 0) { - s->codec_type = 2; + else if ((d & 0x60) != 0) s->ic_type = IC_OPUS; - } else + else goto _done; outb(0xfa, ODIE_ADDR_IO(s->io_base)); @@ -862,10 +846,10 @@ static int __devinit detect_sscape(struct soundscape *s) sscape_write_unsafe(s->io_base, GA_HMCTL_REG, d | 0xc0); if (s->type == SSCAPE_VIVO) - codec += 4; + wss_io += 4; /* wait for WSS codec */ for (d = 0; d < 500; d++) { - if ((inb(codec) & 0x80) == 0) + if ((inb(wss_io) & 0x80) == 0) break; spin_unlock_irqrestore(&s->lock, flags); msleep(1); @@ -955,82 +939,6 @@ static int __devinit create_mpu401(struct snd_card *card, int devnum, unsigned l /* - * Override for the CS4231 playback format function. - * The AD1845 has much simpler format and rate selection. - */ -static void ad1845_playback_format(struct snd_wss *chip, - struct snd_pcm_hw_params *params, - unsigned char format) -{ - unsigned long flags; - unsigned rate = params_rate(params); - - /* - * The AD1845 can't handle sample frequencies - * outside of 4 kHZ to 50 kHZ - */ - if (rate > 50000) - rate = 50000; - else if (rate < 4000) - rate = 4000; - - spin_lock_irqsave(&chip->reg_lock, flags); - - /* - * Program the AD1845 correctly for the playback stream. - * Note that we do NOT need to toggle the MCE bit because - * the PLAYBACK_ENABLE bit of the Interface Configuration - * register is set. - * - * NOTE: We seem to need to write to the MSB before the LSB - * to get the correct sample frequency. - */ - snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (format & 0xf0)); - snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); - snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); - - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -/* - * Override for the CS4231 capture format function. - * The AD1845 has much simpler format and rate selection. - */ -static void ad1845_capture_format(struct snd_wss *chip, - struct snd_pcm_hw_params *params, - unsigned char format) -{ - unsigned long flags; - unsigned rate = params_rate(params); - - /* - * The AD1845 can't handle sample frequencies - * outside of 4 kHZ to 50 kHZ - */ - if (rate > 50000) - rate = 50000; - else if (rate < 4000) - rate = 4000; - - spin_lock_irqsave(&chip->reg_lock, flags); - - /* - * Program the AD1845 correctly for the playback stream. - * Note that we do NOT need to toggle the MCE bit because - * the CAPTURE_ENABLE bit of the Interface Configuration - * register is set. - * - * NOTE: We seem to need to write to the MSB before the LSB - * to get the correct sample frequency. - */ - snd_wss_out(chip, CS4231_REC_FORMAT, (format & 0xf0)); - snd_wss_out(chip, AD1845_FREQ_SEL_MSB, (unsigned char) (rate >> 8)); - snd_wss_out(chip, AD1845_FREQ_SEL_LSB, (unsigned char) rate); - - spin_unlock_irqrestore(&chip->reg_lock, flags); -} - -/* * Create an AD1845 PCM subdevice on the SoundScape. The AD1845 * is very much like a CS4231, with a few extra bits. We will * try to support at least some of the extra bits by overriding @@ -1055,11 +963,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, unsigned long flags; struct snd_pcm *pcm; -#define AD1845_FREQ_SEL_ENABLE 0x08 - -#define AD1845_PWR_DOWN_CTRL 0x1b -#define AD1845_CRYS_CLOCK_SEL 0x1d - /* * It turns out that the PLAYBACK_ENABLE bit is set * by the lowlevel driver ... @@ -1074,7 +977,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, */ if (sscape->type != SSCAPE_VIVO) { - int val; /* * The input clock frequency on the SoundScape must * be 14.31818 MHz, because we must set this register @@ -1082,22 +984,10 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, */ snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, AD1845_CRYS_CLOCK_SEL, 0x20); + snd_wss_out(chip, AD1845_CLOCK, 0x20); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); - /* - * More custom configuration: - * a) select "mode 2" and provide a current drive of 8mA - * b) enable frequency selection (for capture/playback) - */ - spin_lock_irqsave(&chip->reg_lock, flags); - snd_wss_out(chip, CS4231_MISC_INFO, - CS4231_MODE2 | 0x10); - val = snd_wss_in(chip, AD1845_PWR_DOWN_CTRL); - snd_wss_out(chip, AD1845_PWR_DOWN_CTRL, - val | AD1845_FREQ_SEL_ENABLE); - spin_unlock_irqrestore(&chip->reg_lock, flags); } err = snd_wss_pcm(chip, 0, &pcm); @@ -1113,11 +1003,13 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, "for AD1845 chip\n"); goto _error; } - err = snd_wss_timer(chip, 0, NULL); - if (err < 0) { - snd_printk(KERN_ERR "sscape: No timer device " - "for AD1845 chip\n"); - goto _error; + if (chip->hardware != WSS_HW_AD1848) { + err = snd_wss_timer(chip, 0, NULL); + if (err < 0) { + snd_printk(KERN_ERR "sscape: No timer device " + "for AD1845 chip\n"); + goto _error; + } } if (sscape->type != SSCAPE_VIVO) { @@ -1128,8 +1020,6 @@ static int __devinit create_ad1845(struct snd_card *card, unsigned port, "MIDI mixer control\n"); goto _error; } - chip->set_playback_format = ad1845_playback_format; - chip->set_capture_format = ad1845_capture_format; } strcpy(card->driver, "SoundScape"); @@ -1157,7 +1047,6 @@ static int __devinit create_sscape(int dev, struct snd_card *card) unsigned dma_cfg; unsigned irq_cfg; unsigned mpu_irq_cfg; - unsigned xport; struct resource *io_res; struct resource *wss_res; unsigned long flags; @@ -1177,15 +1066,15 @@ static int __devinit create_sscape(int dev, struct snd_card *card) printk(KERN_ERR "sscape: Invalid IRQ %d\n", mpu_irq[dev]); return -ENXIO; } - xport = port[dev]; /* * Grab IO ports that we will need to probe so that we * can detect and control this hardware ... */ - io_res = request_region(xport, 8, "SoundScape"); + io_res = request_region(port[dev], 8, "SoundScape"); if (!io_res) { - snd_printk(KERN_ERR "sscape: can't grab port 0x%x\n", xport); + snd_printk(KERN_ERR + "sscape: can't grab port 0x%lx\n", port[dev]); return -EBUSY; } wss_res = NULL; @@ -1212,10 +1101,9 @@ static int __devinit create_sscape(int dev, struct snd_card *card) spin_lock_init(&sscape->fwlock); sscape->io_res = io_res; sscape->wss_res = wss_res; - sscape->io_base = xport; - sscape->wss_base = wss_port[dev]; + sscape->io_base = port[dev]; - if (!detect_sscape(sscape)) { + if (!detect_sscape(sscape, wss_port[dev])) { printk(KERN_ERR "sscape: hardware not detected at 0x%x\n", sscape->io_base); err = -ENODEV; goto _release_dma; @@ -1288,12 +1176,11 @@ static int __devinit create_sscape(int dev, struct snd_card *card) } #define MIDI_DEVNUM 0 if (sscape->type != SSCAPE_VIVO) { - err = create_mpu401(card, MIDI_DEVNUM, - MPU401_IO(xport), mpu_irq[dev]); + err = create_mpu401(card, MIDI_DEVNUM, port[dev], mpu_irq[dev]); if (err < 0) { printk(KERN_ERR "sscape: Failed to create " - "MPU-401 device at 0x%x\n", - MPU401_IO(xport)); + "MPU-401 device at 0x%lx\n", + port[dev]); goto _release_dma; } diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index 95898b2..a34ae7b 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -553,11 +553,11 @@ static int __devinit snd_wavefront_isa_match(struct device *pdev, return 0; #endif if (cs4232_pcm_port[dev] == SNDRV_AUTO_PORT) { - snd_printk("specify CS4232 port\n"); + snd_printk(KERN_ERR "specify CS4232 port\n"); return 0; } if (ics2115_port[dev] == SNDRV_AUTO_PORT) { - snd_printk("specify ICS2115 port\n"); + snd_printk(KERN_ERR "specify ICS2115 port\n"); return 0; } return 1; diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 4c41082..beb312c 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -633,7 +633,7 @@ wavefront_get_sample_status (snd_wavefront_t *dev, int assume_rom) wbuf[1] = i >> 7; if (snd_wavefront_cmd (dev, WFC_IDENTIFY_SAMPLE_TYPE, rbuf, wbuf)) { - snd_printk("cannot identify sample " + snd_printk(KERN_WARNING "cannot identify sample " "type of slot %d\n", i); dev->sample_status[i] = WF_ST_EMPTY; continue; diff --git a/sound/isa/wss/wss_lib.c b/sound/isa/wss/wss_lib.c index 3d6c5f2..5d2ba1b 100644 --- a/sound/isa/wss/wss_lib.c +++ b/sound/isa/wss/wss_lib.c @@ -181,25 +181,6 @@ static void snd_wss_wait(struct snd_wss *chip) udelay(100); } -static void snd_wss_outm(struct snd_wss *chip, unsigned char reg, - unsigned char mask, unsigned char value) -{ - unsigned char tmp = (chip->image[reg] & mask) | value; - - snd_wss_wait(chip); -#ifdef CONFIG_SND_DEBUG - if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); -#endif - chip->image[reg] = tmp; - if (!chip->calibrate_mute) { - wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - wmb(); - wss_outb(chip, CS4231P(REG), tmp); - mb(); - } -} - static void snd_wss_dout(struct snd_wss *chip, unsigned char reg, unsigned char value) { @@ -219,7 +200,8 @@ void snd_wss_out(struct snd_wss *chip, unsigned char reg, unsigned char value) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); + snd_printk(KERN_DEBUG "out: auto calibration time out " + "- reg = 0x%x, value = 0x%x\n", reg, value); #endif wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); wss_outb(chip, CS4231P(REG), value); @@ -235,7 +217,8 @@ unsigned char snd_wss_in(struct snd_wss *chip, unsigned char reg) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("in: auto calibration time out - reg = 0x%x\n", reg); + snd_printk(KERN_DEBUG "in: auto calibration time out " + "- reg = 0x%x\n", reg); #endif wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); mb(); @@ -252,7 +235,7 @@ void snd_cs4236_ext_out(struct snd_wss *chip, unsigned char reg, wss_outb(chip, CS4231P(REG), val); chip->eimage[CS4236_REG(reg)] = val; #if 0 - printk("ext out : reg = 0x%x, val = 0x%x\n", reg, val); + printk(KERN_DEBUG "ext out : reg = 0x%x, val = 0x%x\n", reg, val); #endif } EXPORT_SYMBOL(snd_cs4236_ext_out); @@ -268,7 +251,8 @@ unsigned char snd_cs4236_ext_in(struct snd_wss *chip, unsigned char reg) { unsigned char res; res = wss_inb(chip, CS4231P(REG)); - printk("ext in : reg = 0x%x, val = 0x%x\n", reg, res); + printk(KERN_DEBUG "ext in : reg = 0x%x, val = 0x%x\n", + reg, res); return res; } #endif @@ -394,13 +378,16 @@ void snd_wss_mce_up(struct snd_wss *chip) snd_wss_wait(chip); #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("mce_up - auto calibration time out (0)\n"); + snd_printk(KERN_DEBUG + "mce_up - auto calibration time out (0)\n"); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit |= CS4231_MCE; timeout = wss_inb(chip, CS4231P(REGSEL)); if (timeout == 0x80) - snd_printk("mce_up [0x%lx]: serious init problem - codec still busy\n", chip->port); + snd_printk(KERN_DEBUG "mce_up [0x%lx]: " + "serious init problem - codec still busy\n", + chip->port); if (!(timeout & CS4231_MCE)) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); @@ -419,7 +406,9 @@ void snd_wss_mce_down(struct snd_wss *chip) #ifdef CONFIG_SND_DEBUG if (wss_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) - snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL)); + snd_printk(KERN_DEBUG "mce_down [0x%lx] - " + "auto calibration time out (0)\n", + (long)CS4231P(REGSEL)); #endif spin_lock_irqsave(&chip->reg_lock, flags); chip->mce_bit &= ~CS4231_MCE; @@ -427,7 +416,9 @@ void snd_wss_mce_down(struct snd_wss *chip) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) - snd_printk("mce_down [0x%lx]: serious init problem - codec still busy\n", chip->port); + snd_printk(KERN_DEBUG "mce_down [0x%lx]: " + "serious init problem - codec still busy\n", + chip->port); if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & hw_mask)) return; @@ -565,7 +556,7 @@ static unsigned char snd_wss_get_format(struct snd_wss *chip, if (channels > 1) rformat |= CS4231_STEREO; #if 0 - snd_printk("get_format: 0x%x (mode=0x%x)\n", format, mode); + snd_printk(KERN_DEBUG "get_format: 0x%x (mode=0x%x)\n", format, mode); #endif return rformat; } @@ -587,7 +578,15 @@ static void snd_wss_calibrate_mute(struct snd_wss *chip, int mute) chip->image[CS4231_RIGHT_INPUT]); snd_wss_dout(chip, CS4231_LOOPBACK, chip->image[CS4231_LOOPBACK]); + } else { + snd_wss_dout(chip, CS4231_LEFT_INPUT, + 0); + snd_wss_dout(chip, CS4231_RIGHT_INPUT, + 0); + snd_wss_dout(chip, CS4231_LOOPBACK, + 0xfd); } + snd_wss_dout(chip, CS4231_AUX1_LEFT_INPUT, mute | chip->image[CS4231_AUX1_LEFT_INPUT]); snd_wss_dout(chip, CS4231_AUX1_RIGHT_INPUT, @@ -630,7 +629,6 @@ static void snd_wss_playback_format(struct snd_wss *chip, int full_calib = 1; mutex_lock(&chip->mce_mutex); - snd_wss_calibrate_mute(chip, 1); if (chip->hardware == WSS_HW_CS4231A || (chip->hardware & WSS_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); @@ -646,6 +644,24 @@ static void snd_wss_playback_format(struct snd_wss *chip, full_calib = 0; } spin_unlock_irqrestore(&chip->reg_lock, flags); + } else if (chip->hardware == WSS_HW_AD1845) { + unsigned rate = params_rate(params); + + /* + * Program the AD1845 correctly for the playback stream. + * Note that we do NOT need to toggle the MCE bit because + * the PLAYBACK_ENABLE bit of the Interface Configuration + * register is set. + * + * NOTE: We seem to need to write to the MSB before the LSB + * to get the correct sample frequency. + */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_wss_out(chip, CS4231_PLAYBK_FORMAT, (pdfr & 0xf0)); + snd_wss_out(chip, AD1845_UPR_FREQ_SEL, (rate >> 8) & 0xff); + snd_wss_out(chip, AD1845_LWR_FREQ_SEL, rate & 0xff); + full_calib = 0; + spin_unlock_irqrestore(&chip->reg_lock, flags); } if (full_calib) { snd_wss_mce_up(chip); @@ -663,7 +679,6 @@ static void snd_wss_playback_format(struct snd_wss *chip, udelay(100); /* this seems to help */ snd_wss_mce_down(chip); } - snd_wss_calibrate_mute(chip, 0); mutex_unlock(&chip->mce_mutex); } @@ -675,7 +690,6 @@ static void snd_wss_capture_format(struct snd_wss *chip, int full_calib = 1; mutex_lock(&chip->mce_mutex); - snd_wss_calibrate_mute(chip, 1); if (chip->hardware == WSS_HW_CS4231A || (chip->hardware & WSS_HW_CS4232_MASK)) { spin_lock_irqsave(&chip->reg_lock, flags); @@ -690,6 +704,24 @@ static void snd_wss_capture_format(struct snd_wss *chip, full_calib = 0; } spin_unlock_irqrestore(&chip->reg_lock, flags); + } else if (chip->hardware == WSS_HW_AD1845) { + unsigned rate = params_rate(params); + + /* + * Program the AD1845 correctly for the capture stream. + * Note that we do NOT need to toggle the MCE bit because + * the PLAYBACK_ENABLE bit of the Interface Configuration + * register is set. + * + * NOTE: We seem to need to write to the MSB before the LSB + * to get the correct sample frequency. + */ + spin_lock_irqsave(&chip->reg_lock, flags); + snd_wss_out(chip, CS4231_REC_FORMAT, (cdfr & 0xf0)); + snd_wss_out(chip, AD1845_UPR_FREQ_SEL, (rate >> 8) & 0xff); + snd_wss_out(chip, AD1845_LWR_FREQ_SEL, rate & 0xff); + full_calib = 0; + spin_unlock_irqrestore(&chip->reg_lock, flags); } if (full_calib) { snd_wss_mce_up(chip); @@ -714,7 +746,6 @@ static void snd_wss_capture_format(struct snd_wss *chip, spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); } - snd_wss_calibrate_mute(chip, 0); mutex_unlock(&chip->mce_mutex); } @@ -771,10 +802,11 @@ static void snd_wss_init(struct snd_wss *chip) { unsigned long flags; + snd_wss_calibrate_mute(chip, 1); snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (1)\n"); + snd_printk(KERN_DEBUG "init: (1)\n"); #endif snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -789,18 +821,20 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (2)\n"); + snd_printk(KERN_DEBUG "init: (2)\n"); #endif snd_wss_mce_up(chip); spin_lock_irqsave(&chip->reg_lock, flags); + chip->image[CS4231_IFACE_CTRL] &= ~CS4231_AUTOCALIB; + snd_wss_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); snd_wss_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (3) - afei = 0x%x\n", + snd_printk(KERN_DEBUG "init: (3) - afei = 0x%x\n", chip->image[CS4231_ALT_FEATURE_1]); #endif @@ -817,7 +851,7 @@ static void snd_wss_init(struct snd_wss *chip) snd_wss_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (4)\n"); + snd_printk(KERN_DEBUG "init: (4)\n"); #endif snd_wss_mce_up(chip); @@ -827,9 +861,10 @@ static void snd_wss_init(struct snd_wss *chip) chip->image[CS4231_REC_FORMAT]); spin_unlock_irqrestore(&chip->reg_lock, flags); snd_wss_mce_down(chip); + snd_wss_calibrate_mute(chip, 0); #ifdef SNDRV_DEBUG_MCE - snd_printk("init: (5)\n"); + snd_printk(KERN_DEBUG "init: (5)\n"); #endif } @@ -885,8 +920,6 @@ static void snd_wss_close(struct snd_wss *chip, unsigned int mode) mutex_unlock(&chip->open_mutex); return; } - snd_wss_calibrate_mute(chip, 1); - /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); if (!(chip->hardware & WSS_HW_AD1848_MASK)) @@ -919,8 +952,6 @@ static void snd_wss_close(struct snd_wss *chip, unsigned int mode) wss_outb(chip, CS4231P(STATUS), 0); /* clear IRQ */ spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_wss_calibrate_mute(chip, 0); - chip->mode = 0; mutex_unlock(&chip->open_mutex); } @@ -1113,7 +1144,7 @@ irqreturn_t snd_wss_interrupt(int irq, void *dev_id) if (chip->hardware & WSS_HW_AD1848_MASK) wss_outb(chip, CS4231P(STATUS), 0); else - snd_wss_outm(chip, CS4231_IRQ_STATUS, status, 0); + snd_wss_out(chip, CS4231_IRQ_STATUS, status); spin_unlock(&chip->reg_lock); return IRQ_HANDLED; } @@ -1278,7 +1309,8 @@ static int snd_wss_probe(struct snd_wss *chip) } else if (rev == 0x03) { chip->hardware = WSS_HW_CS4236B; } else { - snd_printk("unknown CS chip with version 0x%x\n", rev); + snd_printk(KERN_ERR + "unknown CS chip with version 0x%x\n", rev); return -ENODEV; /* unknown CS4231 chip? */ } } @@ -1314,6 +1346,10 @@ static int snd_wss_probe(struct snd_wss *chip) chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == WSS_HW_INTERWAVE ? 0xc2 : 0x01; } + /* enable fine grained frequency selection */ + if (chip->hardware == WSS_HW_AD1845) + chip->image[AD1845_PWR_DOWN] = 8; + ptr = (unsigned char *) &chip->image; regnum = (chip->hardware & WSS_HW_AD1848_MASK) ? 16 : 32; snd_wss_mce_down(chip); @@ -1342,7 +1378,10 @@ static int snd_wss_probe(struct snd_wss *chip) case 6: break; default: - snd_printk("unknown CS4235 chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4235 chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x0b) { /* CS4236/B */ switch (id >> 5) { @@ -1353,7 +1392,10 @@ static int snd_wss_probe(struct snd_wss *chip) chip->hardware = WSS_HW_CS4236B; break; default: - snd_printk("unknown CS4236 chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4236 chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x08) { /* CS4237B */ chip->hardware = WSS_HW_CS4237B; @@ -1364,7 +1406,10 @@ static int snd_wss_probe(struct snd_wss *chip) case 7: break; default: - snd_printk("unknown CS4237B chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4237B chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x09) { /* CS4238B */ chip->hardware = WSS_HW_CS4238B; @@ -1374,7 +1419,10 @@ static int snd_wss_probe(struct snd_wss *chip) case 7: break; default: - snd_printk("unknown CS4238B chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4238B chip " + "(enhanced version = 0x%x)\n", + id); } } else if ((id & 0x1f) == 0x1e) { /* CS4239 */ chip->hardware = WSS_HW_CS4239; @@ -1384,10 +1432,15 @@ static int snd_wss_probe(struct snd_wss *chip) case 6: break; default: - snd_printk("unknown CS4239 chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4239 chip " + "(enhanced version = 0x%x)\n", + id); } } else { - snd_printk("unknown CS4236/CS423xB chip (enhanced version = 0x%x)\n", id); + snd_printk(KERN_WARNING + "unknown CS4236/CS423xB chip " + "(enhanced version = 0x%x)\n", id); } } } @@ -1618,7 +1671,8 @@ static void snd_wss_resume(struct snd_wss *chip) wss_outb(chip, CS4231P(REGSEL), chip->mce_bit | (timeout & 0x1f)); spin_unlock_irqrestore(&chip->reg_lock, flags); if (timeout == 0x80) - snd_printk("down [0x%lx]: serious init problem - codec still busy\n", chip->port); + snd_printk(KERN_ERR "down [0x%lx]: serious init problem " + "- codec still busy\n", chip->port); if ((timeout & CS4231_MCE) == 0 || !(chip->hardware & (WSS_HW_CS4231_MASK | WSS_HW_CS4232_MASK))) { return; @@ -1628,7 +1682,7 @@ static void snd_wss_resume(struct snd_wss *chip) } #endif /* CONFIG_PM */ -static int snd_wss_free(struct snd_wss *chip) +int snd_wss_free(struct snd_wss *chip) { release_and_free_resource(chip->res_port); release_and_free_resource(chip->res_cport); @@ -1651,6 +1705,7 @@ static int snd_wss_free(struct snd_wss *chip) kfree(chip); return 0; } +EXPORT_SYMBOL(snd_wss_free); static int snd_wss_dev_free(struct snd_device *device) { @@ -1820,7 +1875,8 @@ int snd_wss_create(struct snd_card *card, #if 0 if (chip->hardware & WSS_HW_CS4232_MASK) { if (chip->res_cport == NULL) - snd_printk("CS4232 control port features are not accessible\n"); + snd_printk(KERN_ERR "CS4232 control port features are " + "not accessible\n"); } #endif diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index 99e1391b..3e763d6 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -679,7 +679,7 @@ au1000_init(void) return err; } - printk( KERN_INFO "ALSA AC97: Driver Initialized\n" ); + printk(KERN_INFO "ALSA AC97: Driver Initialized\n"); au1000_card = card; return 0; } diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 7cf9913..d12bd98 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -280,7 +280,7 @@ static void wait_for_calibration(ad1848_info * devc) while (timeout > 0 && (ad_read(devc, 11) & 0x20)) timeout--; if (ad_read(devc, 11) & 0x20) - if ( (devc->model != MD_1845) || (devc->model != MD_1845_SSCAPE)) + if ((devc->model != MD_1845) && (devc->model != MD_1845_SSCAPE)) printk(KERN_WARNING "ad1848: Auto calibration timed out(3).\n"); } @@ -2107,7 +2107,7 @@ int ad1848_control(int cmd, int arg) switch (cmd) { case AD1848_SET_XTAL: /* Change clock frequency of AD1845 (only ) */ - if (devc->model != MD_1845 || devc->model != MD_1845_SSCAPE) + if (devc->model != MD_1845 && devc->model != MD_1845_SSCAPE) return -EINVAL; spin_lock_irqsave(&devc->lock,flags); ad_enter_MCE(devc); diff --git a/sound/oss/au1550_ac97.c b/sound/oss/au1550_ac97.c index 81e1f44..4191acc 100644 --- a/sound/oss/au1550_ac97.c +++ b/sound/oss/au1550_ac97.c @@ -1627,7 +1627,9 @@ au1550_ioctl(struct inode *inode, struct file *file, unsigned int cmd, sizeof(abinfo)) ? -EFAULT : 0; case SNDCTL_DSP_NONBLOCK: + spin_lock(&file->f_lock); file->f_flags |= O_NONBLOCK; + spin_unlock(&file->f_lock); return 0; case SNDCTL_DSP_GETODELAY: diff --git a/sound/oss/audio.c b/sound/oss/audio.c index 89bd27a..b69c05b 100644 --- a/sound/oss/audio.c +++ b/sound/oss/audio.c @@ -433,7 +433,9 @@ int audio_ioctl(int dev, struct file *file, unsigned int cmd, void __user *arg) return dma_ioctl(dev, cmd, arg); case SNDCTL_DSP_NONBLOCK: + spin_lock(&file->f_lock); file->f_flags |= O_NONBLOCK; + spin_unlock(&file->f_lock); return 0; case SNDCTL_DSP_GETCAPS: diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index 1e90d76..1bfcf7e 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -439,7 +439,7 @@ int DMAbuf_sync(int dev) DMAbuf_launch_output(dev, dmap); adev->dmap_out->flags |= DMA_SYNCING; adev->dmap_out->underrun_count = 0; - while (!signal_pending(current) && n++ <= adev->dmap_out->nbufs && + while (!signal_pending(current) && n++ < adev->dmap_out->nbufs && adev->dmap_out->qlen && adev->dmap_out->underrun_count == 0) { long t = dmabuf_timeout(dmap); spin_unlock_irqrestore(&dmap->lock,flags); diff --git a/sound/oss/dmasound/dmasound_atari.c b/sound/oss/dmasound/dmasound_atari.c index 57d9f15..1f47741 100644 --- a/sound/oss/dmasound/dmasound_atari.c +++ b/sound/oss/dmasound/dmasound_atari.c @@ -847,23 +847,23 @@ static int __init AtaIrqInit(void) of events. So all we need to keep the music playing is to provide the sound hardware with new data upon an interrupt from timer A. */ - mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */ - mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ - mfp.tim_ct_a = 8; /* Turn on event counting. */ + st_mfp.tim_ct_a = 0; /* ++roman: Stop timer before programming! */ + st_mfp.tim_dt_a = 1; /* Cause interrupt after first event. */ + st_mfp.tim_ct_a = 8; /* Turn on event counting. */ /* Register interrupt handler. */ if (request_irq(IRQ_MFP_TIMA, AtaInterrupt, IRQ_TYPE_SLOW, "DMA sound", AtaInterrupt)) return 0; - mfp.int_en_a |= 0x20; /* Turn interrupt on. */ - mfp.int_mk_a |= 0x20; + st_mfp.int_en_a |= 0x20; /* Turn interrupt on. */ + st_mfp.int_mk_a |= 0x20; return 1; } #ifdef MODULE static void AtaIrqCleanUp(void) { - mfp.tim_ct_a = 0; /* stop timer */ - mfp.int_en_a &= ~0x20; /* turn interrupt off */ + st_mfp.tim_ct_a = 0; /* stop timer */ + st_mfp.int_en_a &= ~0x20; /* turn interrupt off */ free_irq(IRQ_MFP_TIMA, AtaInterrupt); } #endif /* MODULE */ @@ -1524,7 +1524,7 @@ static SETTINGS def_soft = { .speed = 8000 } ; -static MACHINE machTT = { +static __initdata MACHINE machTT = { .name = "Atari", .name2 = "TT", .owner = THIS_MODULE, @@ -1553,7 +1553,7 @@ static MACHINE machTT = { .capabilities = DSP_CAP_BATCH /* As per SNDCTL_DSP_GETCAPS */ }; -static MACHINE machFalcon = { +static __initdata MACHINE machFalcon = { .name = "Atari", .name2 = "FALCON", .dma_alloc = AtaAlloc, @@ -1599,7 +1599,7 @@ static int __init dmasound_atari_init(void) is_falcon = 0; } else return -ENODEV; - if ((mfp.int_en_a & mfp.int_mk_a & 0x20) == 0) + if ((st_mfp.int_en_a & st_mfp.int_mk_a & 0x20) == 0) return dmasound_init(); else { printk("DMA sound driver: Timer A interrupt already in use\n"); diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c index 25f3a22..7f377ec 100644 --- a/sound/oss/pas2_card.c +++ b/sound/oss/pas2_card.c @@ -156,9 +156,7 @@ static int __init config_pas_hw(struct address_info *hw_config) * 0x80 */ , 0xB88); - pas_write(0x80 - | joystick?0x40:0 - ,0xF388); + pas_write(0x80 | (joystick ? 0x40 : 0), 0xF388); if (pas_irq < 0 || pas_irq > 15) { diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 16ed069..83f5ee2 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -46,7 +46,7 @@ * load the driver as it did in previous versions. * 04-07-1999: Anthony Barbachan <barbcode@xmen.cis.fordham.edu> * Added module parameter pss_firmware to allow the user to tell - * the driver where the fireware file is located. The default + * the driver where the firmware file is located. The default * setting is the previous hardcoded setting "/etc/sound/pss_synth". * 00-03-03: Christoph Hellwig <chhellwig@infradead.org> * Adapted to module_init/module_exit @@ -457,10 +457,9 @@ static void pss_mixer_reset(pss_confdata *devc) } } -static int set_volume_mono(unsigned __user *p, int *aleft) +static int set_volume_mono(unsigned __user *p, unsigned int *aleft) { - int left; - unsigned volume; + unsigned int left, volume; if (get_user(volume, p)) return -EFAULT; @@ -471,10 +470,11 @@ static int set_volume_mono(unsigned __user *p, int *aleft) return 0; } -static int set_volume_stereo(unsigned __user *p, int *aleft, int *aright) +static int set_volume_stereo(unsigned __user *p, + unsigned int *aleft, + unsigned int *aright) { - int left, right; - unsigned volume; + unsigned int left, right, volume; if (get_user(volume, p)) return -EFAULT; diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 5c215f7..c798746 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -212,7 +212,6 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun { unsigned char event_rec[EV_SZ], ev_code; int p = 0, c, ev_size; - int err; int mode = translate_mode(file); dev = dev >> 4; @@ -285,7 +284,7 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun { if (!midi_opened[event_rec[2]]) { - int mode; + int err, mode; int dev = event_rec[2]; if (dev >= max_mididev || midi_devs[dev]==NULL) diff --git a/sound/oss/sh_dac_audio.c b/sound/oss/sh_dac_audio.c index e5d4239..78cfb66 100644 --- a/sound/oss/sh_dac_audio.c +++ b/sound/oss/sh_dac_audio.c @@ -135,7 +135,9 @@ static int dac_audio_ioctl(struct inode *inode, struct file *file, return put_user(AFMT_U8, (int *)arg); case SNDCTL_DSP_NONBLOCK: + spin_lock(&file->f_lock); file->f_flags |= O_NONBLOCK; + spin_unlock(&file->f_lock); return 0; case SNDCTL_DSP_GETCAPS: diff --git a/sound/oss/swarm_cs4297a.c b/sound/oss/swarm_cs4297a.c index 41562ec..1edab7b 100644 --- a/sound/oss/swarm_cs4297a.c +++ b/sound/oss/swarm_cs4297a.c @@ -2200,7 +2200,9 @@ static int cs4297a_ioctl(struct inode *inode, struct file *file, sizeof(abinfo)) ? -EFAULT : 0; case SNDCTL_DSP_NONBLOCK: + spin_lock(&file->f_lock); file->f_flags |= O_NONBLOCK; + spin_unlock(&file->f_lock); return 0; case SNDCTL_DSP_GETODELAY: diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 78b8acc..187f727 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -2673,7 +2673,9 @@ static int vwsnd_audio_do_ioctl(struct inode *inode, case SNDCTL_DSP_NONBLOCK: /* _SIO ('P',14) */ DBGX("SNDCTL_DSP_NONBLOCK\n"); + spin_lock(&file->f_lock); file->f_flags |= O_NONBLOCK; + spin_unlock(&file->f_lock); return 0; case SNDCTL_DSP_RESET: /* _SIO ('P', 0) */ diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index 6e3a184..93422e3 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -400,6 +400,26 @@ config SND_INDIGODJ To compile this driver as a module, choose M here: the module will be called snd-indigodj +config SND_INDIGOIOX + tristate "(Echoaudio) Indigo IOx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo IOx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigoiox + +config SND_INDIGODJX + tristate "(Echoaudio) Indigo DJx" + select FW_LOADER + select SND_PCM + help + Say 'Y' or 'M' to include support for Echoaudio Indigo DJx. + + To compile this driver as a module, choose M here: the module + will be called snd-indigodjx + config SND_EMU10K1 tristate "Emu10k1 (SB Live!, Audigy, E-mu APS)" select FW_LOADER @@ -487,7 +507,7 @@ config SND_FM801 config SND_FM801_TEA575X_BOOL bool "ForteMedia FM801 + TEA5757 tuner" depends on SND_FM801 - depends on VIDEO_V4L1=y || VIDEO_V4L1=SND_FM801 + depends on VIDEO_V4L2=y || VIDEO_V4L2=SND_FM801 help Say Y here to include support for soundcards based on the ForteMedia FM801 chip with a TEA5757 tuner connected to GPIO1-3 pins (Media @@ -744,8 +764,9 @@ config SND_VIRTUOSO select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the - Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X and - HDAV1.3 (Deluxe). + Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2, D2X, and + Essence STX. + Support for the HDAV1.3 (Deluxe) is very experimental. To compile this driver as a module, choose M here: the module will be called snd-virtuoso. diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index e2b843b..97ee127 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -143,6 +143,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x43525970, 0xfffffff8, "CS4202", NULL, NULL }, { 0x43585421, 0xffffffff, "HSD11246", NULL, NULL }, // SmartMC II { 0x43585428, 0xfffffff8, "Cx20468", patch_conexant, NULL }, // SmartAMC fixme: the mask might be different +{ 0x43585430, 0xffffffff, "Cx20468-31", patch_conexant, NULL }, { 0x43585431, 0xffffffff, "Cx20551", patch_cx20551, NULL }, { 0x44543031, 0xfffffff0, "DT0398", NULL, NULL }, { 0x454d4328, 0xffffffff, "EM28028", NULL, NULL }, // same as TR28028? @@ -383,7 +384,7 @@ int snd_ac97_update_bits(struct snd_ac97 *ac97, unsigned short reg, unsigned sho EXPORT_SYMBOL(snd_ac97_update_bits); -/* no lock version - see snd_ac97_updat_bits() */ +/* no lock version - see snd_ac97_update_bits() */ int snd_ac97_update_bits_nolock(struct snd_ac97 *ac97, unsigned short reg, unsigned short mask, unsigned short value) { @@ -1643,7 +1644,10 @@ static int snd_ac97_modem_build(struct snd_card *card, struct snd_ac97 * ac97) { int err, idx; - //printk("AC97_GPIO_CFG = %x\n",snd_ac97_read(ac97,AC97_GPIO_CFG)); + /* + printk(KERN_DEBUG "AC97_GPIO_CFG = %x\n", + snd_ac97_read(ac97,AC97_GPIO_CFG)); + */ snd_ac97_write(ac97, AC97_GPIO_CFG, 0xffff & ~(AC97_GPIO_LINE1_OH)); snd_ac97_write(ac97, AC97_GPIO_POLARITY, 0xffff & ~(AC97_GPIO_LINE1_OH)); snd_ac97_write(ac97, AC97_GPIO_STICKY, 0xffff); diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 060ea59..73b17d5 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -125,6 +125,8 @@ static void snd_ac97_proc_read_main(struct snd_ac97 *ac97, struct snd_info_buffe snd_iprintf(buffer, "PCI Subsys Device: 0x%04x\n\n", ac97->subsystem_device); + snd_iprintf(buffer, "Flags: %x\n", ac97->flags); + if ((ac97->ext_id & AC97_EI_REV_MASK) >= AC97_EI_REV_23) { val = snd_ac97_read(ac97, AC97_INT_PAGING); snd_ac97_update_bits(ac97, AC97_INT_PAGING, diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index d1f242b..8f5098f 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -909,8 +909,8 @@ snd_ad1889_create(struct snd_card *card, return err; /* check PCI availability (32bit DMA) */ - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { printk(KERN_ERR PFX "error setting 32-bit DMA mask.\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/ak4531_codec.c b/sound/pci/ak4531_codec.c index 0f819dd..fd135e3 100644 --- a/sound/pci/ak4531_codec.c +++ b/sound/pci/ak4531_codec.c @@ -51,7 +51,8 @@ static void snd_ak4531_dump(struct snd_ak4531 *ak4531) int idx; for (idx = 0; idx < 0x19; idx++) - printk("ak4531 0x%x: 0x%x\n", idx, ak4531->regs[idx]); + printk(KERN_DEBUG "ak4531 0x%x: 0x%x\n", + idx, ak4531->regs[idx]); } #endif diff --git a/sound/pci/ali5451/ali5451.c b/sound/pci/ali5451/ali5451.c index b36c551..c551006 100644 --- a/sound/pci/ali5451/ali5451.c +++ b/sound/pci/ali5451/ali5451.c @@ -2142,7 +2142,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec) { int err; - snd_ali_printk("resouces allocation ...\n"); + snd_ali_printk("resources allocation ...\n"); err = pci_request_regions(codec->pci, "ALI 5451"); if (err < 0) return err; @@ -2154,7 +2154,7 @@ static int __devinit snd_ali_resources(struct snd_ali *codec) return -EBUSY; } codec->irq = codec->pci->irq; - snd_ali_printk("resouces allocated.\n"); + snd_ali_printk("resources allocated.\n"); return 0; } static int snd_ali_dev_free(struct snd_device *device) @@ -2186,8 +2186,8 @@ static int __devinit snd_ali_create(struct snd_card *card, if (err < 0) return err; /* check, if we can restrict PCI DMA transfers to 31 bits */ - if (pci_set_dma_mask(pci, DMA_31BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_31BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(31)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(31)) < 0) { snd_printk(KERN_ERR "architecture does not support " "31bit PCI busmaster DMA\n"); pci_disable_device(pci); diff --git a/sound/pci/als300.c b/sound/pci/als300.c index f557c15..3aa35af 100644 --- a/sound/pci/als300.c +++ b/sound/pci/als300.c @@ -91,7 +91,7 @@ #define DEBUG_PLAY_REC 0 #if DEBUG_CALLS -#define snd_als300_dbgcalls(format, args...) printk(format, ##args) +#define snd_als300_dbgcalls(format, args...) printk(KERN_DEBUG format, ##args) #define snd_als300_dbgcallenter() printk(KERN_ERR "--> %s\n", __func__) #define snd_als300_dbgcallleave() printk(KERN_ERR "<-- %s\n", __func__) #else @@ -689,8 +689,8 @@ static int __devinit snd_als300_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { printk(KERN_ERR "error setting 28bit DMA mask\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index 542a0c6..3dbacde 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -872,8 +872,8 @@ static int __devinit snd_card_als4000_probe(struct pci_dev *pci, return err; } /* check, if we can restrict PCI DMA transfers to 24 bits */ - if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(24)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(24)) < 0) { snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 9ec1223..7b72c88 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -151,8 +151,8 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) // check PCI availability (DMA). if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { printk(KERN_ERR "error to set DMA mask\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index 649849e..f4aa8ff6 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -462,9 +462,10 @@ static void a3dsrc_ZeroSliceIO(a3dsrc_t * a) /* Reset Single A3D source. */ static void a3dsrc_ZeroState(a3dsrc_t * a) { - - //printk("vortex: ZeroState slice: %d, source %d\n", a->slice, a->source); - + /* + printk(KERN_DEBUG "vortex: ZeroState slice: %d, source %d\n", + a->slice, a->source); + */ a3dsrc_SetAtmosState(a, 0, 0, 0, 0); a3dsrc_SetHrtfState(a, A3dHrirZeros, A3dHrirZeros); a3dsrc_SetItdDline(a, A3dItdDlineZeros); diff --git a/sound/pci/au88x0/au88x0_core.c b/sound/pci/au88x0/au88x0_core.c index b070e57..3906f5a 100644 --- a/sound/pci/au88x0/au88x0_core.c +++ b/sound/pci/au88x0/au88x0_core.c @@ -1135,7 +1135,10 @@ vortex_adbdma_setbuffers(vortex_t * vortex, int adbdma, snd_pcm_sgbuf_get_addr(dma->substream, 0)); break; } - //printk("vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", dma->cfg0, dma->cfg1); + /* + printk(KERN_DEBUG "vortex: cfg0 = 0x%x\nvortex: cfg1=0x%x\n", + dma->cfg0, dma->cfg1); + */ hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG0 + (adbdma << 3), dma->cfg0); hwwrite(vortex->mmio, VORTEX_ADBDMA_BUFCFG1 + (adbdma << 3), dma->cfg1); @@ -1959,7 +1962,7 @@ vortex_connect_codecplay(vortex_t * vortex, int en, unsigned char mixers[]) ADB_CODECOUT(0 + 4)); vortex_connection_mix_adb(vortex, en, 0x11, mixers[3], ADB_CODECOUT(1 + 4)); - //printk("SDAC detected "); + /* printk(KERN_DEBUG "SDAC detected "); */ } #else // Use plain direct output to codec. @@ -2013,7 +2016,11 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) resmap[restype] |= (1 << i); else vortex->dma_adb[i].resources[restype] |= (1 << i); - //printk("vortex: ResManager: type %d out %d\n", restype, i); + /* + printk(KERN_DEBUG + "vortex: ResManager: type %d out %d\n", + restype, i); + */ return i; } } @@ -2024,7 +2031,11 @@ vortex_adb_checkinout(vortex_t * vortex, int resmap[], int out, int restype) for (i = 0; i < qty; i++) { if (resmap[restype] & (1 << i)) { resmap[restype] &= ~(1 << i); - //printk("vortex: ResManager: type %d in %d\n",restype, i); + /* + printk(KERN_DEBUG + "vortex: ResManager: type %d in %d\n", + restype, i); + */ return i; } } @@ -2789,7 +2800,7 @@ vortex_translateformat(vortex_t * vortex, char bits, char nch, int encod) { int a, this_194; - if ((bits != 8) || (bits != 16)) + if ((bits != 8) && (bits != 16)) return -1; switch (encod) { diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index 978b856..2805e34 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -213,38 +213,59 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, switch (reg) { /* Voice specific parameters */ case 0: /* running */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_RUN(wt), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_RUN(wt), (int)val); + */ hwwrite(vortex->mmio, WT_RUN(wt), val); return 0xc; break; case 1: /* param 0 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,0), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,0), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 0), val); return 0xc; break; case 2: /* param 1 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,1), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,1), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 1), val); return 0xc; break; case 3: /* param 2 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,2), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,2), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 2), val); return 0xc; break; case 4: /* param 3 */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_PARM(wt,3), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_PARM(wt,3), (int)val); + */ hwwrite(vortex->mmio, WT_PARM(wt, 3), val); return 0xc; break; case 6: /* mute */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_MUTE(wt), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_MUTE(wt), (int)val); + */ hwwrite(vortex->mmio, WT_MUTE(wt), val); return 0xc; break; case 0xb: { /* delay */ - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", WT_DELAY(wt,0), (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", + WT_DELAY(wt,0), (int)val); + */ hwwrite(vortex->mmio, WT_DELAY(wt, 3), val); hwwrite(vortex->mmio, WT_DELAY(wt, 2), val); hwwrite(vortex->mmio, WT_DELAY(wt, 1), val); @@ -272,7 +293,9 @@ vortex_wt_SetReg(vortex_t * vortex, unsigned char reg, int wt, return 0; break; } - //printk("vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); + /* + printk(KERN_DEBUG "vortex: WT SetReg(0x%x) = 0x%08x\n", ecx, (int)val); + */ hwwrite(vortex->mmio, ecx, val); return 1; } diff --git a/sound/pci/aw2/aw2-alsa.c b/sound/pci/aw2/aw2-alsa.c index eefcbf6..4d34bb0 100644 --- a/sound/pci/aw2/aw2-alsa.c +++ b/sound/pci/aw2/aw2-alsa.c @@ -165,7 +165,7 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable Audiowerk2 soundcard."); static struct pci_device_id snd_aw2_ids[] = { - {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, PCI_ANY_ID, PCI_ANY_ID, + {PCI_VENDOR_ID_SAA7146, PCI_DEVICE_ID_SAA7146, 0, 0, 0, 0, 0}, {0} }; @@ -279,8 +279,8 @@ static int __devinit snd_aw2_create(struct snd_card *card, pci_set_master(pci); /* check PCI availability (32bit DMA) */ - if ((pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) || - (pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0)) { + if ((pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0) || + (pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0)) { printk(KERN_ERR "aw2: Impossible to set 32bit mask DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 1df96e76..f290bc5 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -211,25 +211,25 @@ MODULE_SUPPORTED_DEVICE("{{Aztech,AZF3328}}"); #endif #if DEBUG_MIXER -#define snd_azf3328_dbgmixer(format, args...) printk(format, ##args) +#define snd_azf3328_dbgmixer(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbgmixer(format, args...) #endif #if DEBUG_PLAY_REC -#define snd_azf3328_dbgplay(format, args...) printk(KERN_ERR format, ##args) +#define snd_azf3328_dbgplay(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbgplay(format, args...) #endif #if DEBUG_MISC -#define snd_azf3328_dbgtimer(format, args...) printk(KERN_ERR format, ##args) +#define snd_azf3328_dbgtimer(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbgtimer(format, args...) #endif #if DEBUG_GAME -#define snd_azf3328_dbggame(format, args...) printk(KERN_ERR format, ##args) +#define snd_azf3328_dbggame(format, args...) printk(KERN_DEBUG format, ##args) #else #define snd_azf3328_dbggame(format, args...) #endif @@ -2125,8 +2125,8 @@ snd_azf3328_create(struct snd_card *card, chip->irq = -1; /* check if we can restrict PCI DMA transfers to 24 bits */ - if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(24)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(24)) < 0) { snd_printk(KERN_ERR "architecture does not support " "24bit PCI busmaster DMA\n" ); diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index b116456..bfac30f 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -255,6 +255,14 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .gpio_type = 2, .i2c_adc = 1, .spi_dac = 1 } , + /* Giga-byte GA-G1975X mobo + * Novell bnc#395807 + */ + /* FIXME: the GPIO and I2C setting aren't tested well */ + { .serial = 0x1458a006, + .name = "Giga-byte GA-G1975X", + .gpio_type = 1, + .i2c_adc = 1 }, /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". @@ -404,7 +412,9 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, } tmp = reg << 25 | value << 16; - // snd_printk("I2C-write:reg=0x%x, value=0x%x\n", reg, value); + /* + snd_printk(KERN_DEBUG "I2C-write:reg=0x%x, value=0x%x\n", reg, value); + */ /* Not sure what this I2C channel controls. */ /* snd_ca0106_ptr_write(emu, I2C_D0, 0, tmp); */ @@ -422,7 +432,7 @@ int snd_ca0106_i2c_write(struct snd_ca0106 *emu, /* Wait till the transaction ends */ while (1) { status = snd_ca0106_ptr_read(emu, I2C_A, 0); - //snd_printk("I2C:status=0x%x\n", status); + /*snd_printk(KERN_DEBUG "I2C:status=0x%x\n", status);*/ timeout++; if ((status & I2C_A_ADC_START) == 0) break; @@ -521,7 +531,10 @@ static int snd_ca0106_pcm_open_playback_channel(struct snd_pcm_substream *substr channel->number = channel_id; channel->use = 1; - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + /* + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); + */ //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) @@ -614,7 +627,10 @@ static int snd_ca0106_pcm_open_capture_channel(struct snd_pcm_substream *substre channel->number = channel_id; channel->use = 1; - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); + /* + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); + */ //channel->interrupt = snd_ca0106_pcm_channel_interrupt; channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) @@ -705,9 +721,20 @@ static int snd_ca0106_pcm_prepare_playback(struct snd_pcm_substream *substream) u32 reg71; int i; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, " + "channels=%d, buffer_size=%ld, period_size=%ld, " + "periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, + runtime->channels, runtime->buffer_size, + runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#endif /* debug */ /* Rate can be set per channel. */ /* reg40 control host to fifo */ /* reg71 controls DAC rate. */ @@ -799,9 +826,20 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) u32 reg71_set = 0; u32 reg71; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, " + "channels=%d, buffer_size=%ld, period_size=%ld, " + "periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, + runtime->channels, runtime->buffer_size, + runtime->period_size, runtime->periods, + frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->buffer.addr, emu->buffer.area, emu->buffer.bytes); +#endif /* debug */ /* reg71 controls ADC rate. */ switch (runtime->rate) { case 44100: @@ -846,7 +884,14 @@ static int snd_ca0106_pcm_prepare_capture(struct snd_pcm_substream *substream) } - //printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + /* + printk(KERN_DEBUG + "prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, " + "buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, + frames_to_bytes(runtime, 1)); + */ snd_ca0106_ptr_write(emu, 0x13, channel, 0); snd_ca0106_ptr_write(emu, CAPTURE_DMA_ADDR, channel, runtime->dma_addr); snd_ca0106_ptr_write(emu, CAPTURE_BUFFER_SIZE, channel, frames_to_bytes(runtime, runtime->buffer_size)<<16); // buffer size in bytes @@ -888,13 +933,13 @@ static int snd_ca0106_pcm_trigger_playback(struct snd_pcm_substream *substream, runtime = s->runtime; epcm = runtime->private_data; channel = epcm->channel_id; - /* snd_printk("channel=%d\n",channel); */ + /* snd_printk(KERN_DEBUG "channel=%d\n", channel); */ epcm->running = running; basic |= (0x1 << channel); extended |= (0x10 << channel); snd_pcm_trigger_done(s, substream); } - /* snd_printk("basic=0x%x, extended=0x%x\n",basic, extended); */ + /* snd_printk(KERN_DEBUG "basic=0x%x, extended=0x%x\n",basic, extended); */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -972,8 +1017,13 @@ snd_ca0106_pcm_pointer_playback(struct snd_pcm_substream *substream) ptr=ptr2; if (ptr >= runtime->buffer_size) ptr -= runtime->buffer_size; - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -995,8 +1045,13 @@ snd_ca0106_pcm_pointer_capture(struct snd_pcm_substream *substream) ptr=ptr2; if (ptr >= runtime->buffer_size) ptr -= runtime->buffer_size; - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -1181,8 +1236,12 @@ static irqreturn_t snd_ca0106_interrupt(int irq, void *dev_id) return IRQ_NONE; stat76 = snd_ca0106_ptr_read(chip, EXTENDED_INT, 0); - //snd_printk("interrupt status = 0x%08x, stat76=0x%08x\n", status, stat76); - //snd_printk("ptr=0x%08x\n",snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + /* + snd_printk(KERN_DEBUG "interrupt status = 0x%08x, stat76=0x%08x\n", + status, stat76); + snd_printk(KERN_DEBUG "ptr=0x%08x\n", + snd_ca0106_ptr_read(chip, PLAYBACK_POINTER, 0)); + */ mask = 0x11; /* 0x1 for one half, 0x10 for the other half period. */ for(i = 0; i < 4; i++) { pchannel = &(chip->playback_channels[i]); @@ -1470,7 +1529,7 @@ static void ca0106_init_chip(struct snd_ca0106 *chip, int resume) int size, n; size = ARRAY_SIZE(i2c_adc_init); - /* snd_printk("I2C:array size=0x%x\n", size); */ + /* snd_printk(KERN_DEBUG "I2C:array size=0x%x\n", size); */ for (n = 0; n < size; n++) snd_ca0106_i2c_write(chip, i2c_adc_init[n][0], i2c_adc_init[n][1]); @@ -1530,8 +1589,8 @@ static int __devinit snd_ca0106_create(int dev, struct snd_card *card, err = pci_enable_device(pci); if (err < 0) return err; - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { printk(KERN_ERR "error to set 32bit mask DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index b9b07f4..f6286f8 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -834,7 +834,11 @@ static snd_pcm_uframes_t snd_cs4281_pointer(struct snd_pcm_substream *substream) struct cs4281_dma *dma = runtime->private_data; struct cs4281 *chip = snd_pcm_substream_chip(substream); - // printk("DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, jiffies); + /* + printk(KERN_DEBUG "DCC = 0x%x, buffer_size = 0x%x, jiffies = %li\n", + snd_cs4281_peekBA0(chip, dma->regDCC), runtime->buffer_size, + jiffies); + */ return runtime->buffer_size - snd_cs4281_peekBA0(chip, dma->regDCC) - 1; } diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 8ab07aa..1be96ea 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -194,7 +194,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, * ACSDA = Status Data Register = 474h */ #if 0 - printk("e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg, + printk(KERN_DEBUG "e) reg = 0x%x, val = 0x%x, BA0_ACCAD = 0x%x\n", reg, snd_cs46xx_peekBA0(chip, BA0_ACSDA), snd_cs46xx_peekBA0(chip, BA0_ACCAD)); #endif @@ -428,8 +428,8 @@ static int cs46xx_wait_for_fifo(struct snd_cs46xx * chip,int retry_timeout) } if(status & SERBST_WBSY) { - snd_printk( KERN_ERR "cs46xx: failure waiting for FIFO command to complete\n"); - + snd_printk(KERN_ERR "cs46xx: failure waiting for " + "FIFO command to complete\n"); return -EINVAL; } diff --git a/sound/pci/cs46xx/cs46xx_lib.h b/sound/pci/cs46xx/cs46xx_lib.h index 018a7de..4eb55aa 100644 --- a/sound/pci/cs46xx/cs46xx_lib.h +++ b/sound/pci/cs46xx/cs46xx_lib.h @@ -62,7 +62,11 @@ static inline void snd_cs46xx_poke(struct snd_cs46xx *chip, unsigned long reg, u unsigned int bank = reg >> 16; unsigned int offset = reg & 0xffff; - /*if (bank == 0) printk("snd_cs46xx_poke: %04X - %08X\n",reg >> 2,val); */ + /* + if (bank == 0) + printk(KERN_DEBUG "snd_cs46xx_poke: %04X - %08X\n", + reg >> 2,val); + */ writel(val, chip->region.idx[bank+1].remap_addr + offset); } diff --git a/sound/pci/cs5535audio/cs5535audio.c b/sound/pci/cs5535audio/cs5535audio.c index ac1d72e..05f56e0 100644 --- a/sound/pci/cs5535audio/cs5535audio.c +++ b/sound/pci/cs5535audio/cs5535audio.c @@ -285,8 +285,8 @@ static int __devinit snd_cs5535audio_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { printk(KERN_WARNING "unable to get 32bit dma\n"); err = -ENXIO; goto pcifail; @@ -312,7 +312,7 @@ static int __devinit snd_cs5535audio_create(struct snd_card *card, if (request_irq(pci->irq, snd_cs5535audio_interrupt, IRQF_SHARED, "CS5535 Audio", cs5535au)) { - snd_printk("unable to grab IRQ %d\n", pci->irq); + snd_printk(KERN_ERR "unable to grab IRQ %d\n", pci->irq); err = -EBUSY; goto sndfail; } diff --git a/sound/pci/echoaudio/Makefile b/sound/pci/echoaudio/Makefile index 7b576ae..1361de7 100644 --- a/sound/pci/echoaudio/Makefile +++ b/sound/pci/echoaudio/Makefile @@ -15,6 +15,8 @@ snd-echo3g-objs := echo3g.o snd-indigo-objs := indigo.o snd-indigoio-objs := indigoio.o snd-indigodj-objs := indigodj.o +snd-indigoiox-objs := indigoiox.o +snd-indigodjx-objs := indigodjx.o obj-$(CONFIG_SND_DARLA20) += snd-darla20.o obj-$(CONFIG_SND_GINA20) += snd-gina20.o @@ -28,3 +30,5 @@ obj-$(CONFIG_SND_ECHO3G) += snd-echo3g.o obj-$(CONFIG_SND_INDIGO) += snd-indigo.o obj-$(CONFIG_SND_INDIGOIO) += snd-indigoio.o obj-$(CONFIG_SND_INDIGODJ) += snd-indigodj.o +obj-$(CONFIG_SND_INDIGOIOX) += snd-indigoiox.o +obj-$(CONFIG_SND_INDIGODJX) += snd-indigodjx.o diff --git a/sound/pci/echoaudio/echo3g_dsp.c b/sound/pci/echoaudio/echo3g_dsp.c index 417e25a..57967e5 100644 --- a/sound/pci/echoaudio/echo3g_dsp.c +++ b/sound/pci/echoaudio/echo3g_dsp.c @@ -56,7 +56,7 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) } chip->comm_page->e3g_frq_register = - __constant_cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2); + cpu_to_le32((E3G_MAGIC_NUMBER / 48000) - 2); chip->device_id = device_id; chip->subdevice_id = subdevice_id; chip->bad_board = TRUE; diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 9d015a7..da2065c 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -950,6 +950,8 @@ static int __devinit snd_echo_new_pcm(struct echoaudio *chip) Control interface ******************************************************************************/ +#ifndef ECHOCARD_HAS_VMIXER + /******************* PCM output volume *******************/ static int snd_echo_output_gain_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -1001,18 +1003,6 @@ static int snd_echo_output_gain_put(struct snd_kcontrol *kcontrol, return changed; } -#ifdef ECHOCARD_HAS_VMIXER -/* On Vmixer cards this one controls the line-out volume */ -static struct snd_kcontrol_new snd_echo_line_output_gain __devinitdata = { - .name = "Line Playback Volume", - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .access = SNDRV_CTL_ELEM_ACCESS_READWRITE | SNDRV_CTL_ELEM_ACCESS_TLV_READ, - .info = snd_echo_output_gain_info, - .get = snd_echo_output_gain_get, - .put = snd_echo_output_gain_put, - .tlv = {.p = db_scale_output_gain}, -}; -#else static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .name = "PCM Playback Volume", .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1022,6 +1012,7 @@ static struct snd_kcontrol_new snd_echo_pcm_output_gain __devinitdata = { .put = snd_echo_output_gain_put, .tlv = {.p = db_scale_output_gain}, }; + #endif @@ -2037,8 +2028,6 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, #ifdef ECHOCARD_HAS_VMIXER snd_echo_vmixer.count = num_pipes_out(chip) * num_busses_out(chip); - if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_line_output_gain, chip))) < 0) - goto ctl_error; if ((err = snd_ctl_add(chip->card, snd_ctl_new1(&snd_echo_vmixer, chip))) < 0) goto ctl_error; #else diff --git a/sound/pci/echoaudio/echoaudio.h b/sound/pci/echoaudio/echoaudio.h index 1c88e05..f9490ae 100644 --- a/sound/pci/echoaudio/echoaudio.h +++ b/sound/pci/echoaudio/echoaudio.h @@ -189,6 +189,9 @@ #define INDIGO 0x0090 #define INDIGO_IO 0x00a0 #define INDIGO_DJ 0x00b0 +#define DC8 0x00c0 +#define INDIGO_IOX 0x00d0 +#define INDIGO_DJX 0x00e0 #define ECHO3G 0x0100 diff --git a/sound/pci/echoaudio/echoaudio_3g.c b/sound/pci/echoaudio/echoaudio_3g.c index c3736bb..e32a748 100644 --- a/sound/pci/echoaudio/echoaudio_3g.c +++ b/sound/pci/echoaudio/echoaudio_3g.c @@ -40,8 +40,7 @@ static int check_asic_status(struct echoaudio *chip) if (wait_handshake(chip)) return -EIO; - chip->comm_page->ext_box_status = - __constant_cpu_to_le32(E3G_ASIC_NOT_LOADED); + chip->comm_page->ext_box_status = cpu_to_le32(E3G_ASIC_NOT_LOADED); chip->asic_loaded = FALSE; clear_handshake(chip); send_vector(chip, DSP_VC_TEST_ASIC); diff --git a/sound/pci/echoaudio/echoaudio_dsp.c b/sound/pci/echoaudio/echoaudio_dsp.c index be0e181..4df51ef 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.c +++ b/sound/pci/echoaudio/echoaudio_dsp.c @@ -926,11 +926,11 @@ static int init_dsp_comm_page(struct echoaudio *chip) /* Init the comm page */ chip->comm_page->comm_size = - __constant_cpu_to_le32(sizeof(struct comm_page)); + cpu_to_le32(sizeof(struct comm_page)); chip->comm_page->handshake = 0xffffffff; chip->comm_page->midi_out_free_count = - __constant_cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); - chip->comm_page->sample_rate = __constant_cpu_to_le32(44100); + cpu_to_le32(DSP_MIDI_OUT_FIFO_SIZE); + chip->comm_page->sample_rate = cpu_to_le32(44100); chip->sample_rate = 44100; /* Set line levels so we don't blast any inputs on startup */ diff --git a/sound/pci/echoaudio/echoaudio_dsp.h b/sound/pci/echoaudio/echoaudio_dsp.h index e352f3a..cb7d75a 100644 --- a/sound/pci/echoaudio/echoaudio_dsp.h +++ b/sound/pci/echoaudio/echoaudio_dsp.h @@ -576,8 +576,13 @@ SET_LAYLA24_FREQUENCY_REG command. #define E3G_ASIC_NOT_LOADED 0xffff #define E3G_BOX_TYPE_MASK 0xf0 -#define EXT_3GBOX_NC 0x01 -#define EXT_3GBOX_NOT_SET 0x02 +/* Indigo express control register values */ +#define INDIGO_EXPRESS_32000 0x02 +#define INDIGO_EXPRESS_44100 0x01 +#define INDIGO_EXPRESS_48000 0x00 +#define INDIGO_EXPRESS_DOUBLE_SPEED 0x10 +#define INDIGO_EXPRESS_QUAD_SPEED 0x04 +#define INDIGO_EXPRESS_CLOCK_MASK 0x17 /* diff --git a/sound/pci/echoaudio/gina20_dsp.c b/sound/pci/echoaudio/gina20_dsp.c index db6c952..3f1e747 100644 --- a/sound/pci/echoaudio/gina20_dsp.c +++ b/sound/pci/echoaudio/gina20_dsp.c @@ -208,10 +208,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof) DE_ACT(("set_professional_spdif %d\n", prof)); if (prof) chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + ~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); chip->professional_spdif = prof; return update_flags(chip); } diff --git a/sound/pci/echoaudio/indigo_dsp.c b/sound/pci/echoaudio/indigo_dsp.c index f05e39f..0b2cd9c 100644 --- a/sound/pci/echoaudio/indigo_dsp.c +++ b/sound/pci/echoaudio/indigo_dsp.c @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: all vchannels are routed - to the stereo output */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 0, 6, 0); - set_vmixer_gain(chip, 1, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff --git a/sound/pci/echoaudio/indigo_express_dsp.c b/sound/pci/echoaudio/indigo_express_dsp.c new file mode 100644 index 0000000..9ab625e --- /dev/null +++ b/sound/pci/echoaudio/indigo_express_dsp.c @@ -0,0 +1,119 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library is distributed in the hope that it will be useful, +but WITHOUT ANY WARRANTY; without even the implied warranty of +MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini@shiny.it> + +*************************************************************************/ + +static int set_sample_rate(struct echoaudio *chip, u32 rate) +{ + u32 clock, control_reg, old_control_reg; + + if (wait_handshake(chip)) + return -EIO; + + old_control_reg = le32_to_cpu(chip->comm_page->control_register); + control_reg = old_control_reg & ~INDIGO_EXPRESS_CLOCK_MASK; + + switch (rate) { + case 32000: + clock = INDIGO_EXPRESS_32000; + break; + case 44100: + clock = INDIGO_EXPRESS_44100; + break; + case 48000: + clock = INDIGO_EXPRESS_48000; + break; + case 64000: + clock = INDIGO_EXPRESS_32000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 88200: + clock = INDIGO_EXPRESS_44100|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + case 96000: + clock = INDIGO_EXPRESS_48000|INDIGO_EXPRESS_DOUBLE_SPEED; + break; + default: + return -EINVAL; + } + + control_reg |= clock; + if (control_reg != old_control_reg) { + chip->comm_page->control_register = cpu_to_le32(control_reg); + chip->sample_rate = rate; + clear_handshake(chip); + return send_vector(chip, DSP_VC_UPDATE_CLOCKS); + } + return 0; +} + + + +/* This function routes the sound from a virtual channel to a real output */ +static int set_vmixer_gain(struct echoaudio *chip, u16 output, u16 pipe, + int gain) +{ + int index; + + if (snd_BUG_ON(pipe >= num_pipes_out(chip) || + output >= num_busses_out(chip))) + return -EINVAL; + + if (wait_handshake(chip)) + return -EIO; + + chip->vmixer_gain[output][pipe] = gain; + index = output * num_pipes_out(chip) + pipe; + chip->comm_page->vmixer[index] = gain; + + DE_ACT(("set_vmixer_gain: pipe %d, out %d = %d\n", pipe, output, gain)); + return 0; +} + + + +/* Tell the DSP to read and update virtual mixer levels in comm page. */ +static int update_vmixer_level(struct echoaudio *chip) +{ + if (wait_handshake(chip)) + return -EIO; + clear_handshake(chip); + return send_vector(chip, DSP_VC_SET_VMIXER_GAIN); +} + + + +static u32 detect_input_clocks(const struct echoaudio *chip) +{ + return ECHO_CLOCK_BIT_INTERNAL; +} + + + +/* The IndigoIO has no ASIC. Just do nothing */ +static int load_asic(struct echoaudio *chip) +{ + return 0; +} diff --git a/sound/pci/echoaudio/indigodj_dsp.c b/sound/pci/echoaudio/indigodj_dsp.c index 90730a5..0839291 100644 --- a/sound/pci/echoaudio/indigodj_dsp.c +++ b/sound/pci/echoaudio/indigodj_dsp.c @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: vchannels 0-3 and - vchannels 4-7 are routed to real channels 0-4 */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 2, 2, 0); - set_vmixer_gain(chip, 3, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 2, 6, 0); - set_vmixer_gain(chip, 3, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c new file mode 100644 index 0000000..3482ef6 --- /dev/null +++ b/sound/pci/echoaudio/indigodjx.c @@ -0,0 +1,107 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini@shiny.it> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_DJX +#define ECHOCARD_NAME "Indigo DJx" +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 0 */ +#define PX_DIGITAL_IN 8 /* 0 */ +#define PX_NUM 8 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 4 */ +#define BX_DIGITAL_OUT 4 /* 0 */ +#define BX_ANALOG_IN 4 /* 0 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/firmware.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/asoundef.h> +#include <sound/initval.h> +#include <asm/atomic.h> +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_djx_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_DJX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_djx_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00E0, 0, 0, 0}, /* Indigo DJx*/ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 4, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigodjx_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" diff --git a/sound/pci/echoaudio/indigodjx_dsp.c b/sound/pci/echoaudio/indigodjx_dsp.c new file mode 100644 index 0000000..f591fc2 --- /dev/null +++ b/sound/pci/echoaudio/indigodjx_dsp.c @@ -0,0 +1,68 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library is distributed in the hope that it will be useful, +but WITHOUT ANY WARRANTY; without even the implied warranty of +MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini@shiny.it> + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo DJx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_DJX)) + return -ENODEV; + + err = init_dsp_comm_page(chip); + if (err < 0) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_DJX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + err = load_firmware(chip); + if (err < 0) + return err; + chip->bad_board = FALSE; + + err = init_line_levels(chip); + if (err < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff --git a/sound/pci/echoaudio/indigoio_dsp.c b/sound/pci/echoaudio/indigoio_dsp.c index a7e09ec..0604c8a 100644 --- a/sound/pci/echoaudio/indigoio_dsp.c +++ b/sound/pci/echoaudio/indigoio_dsp.c @@ -63,18 +63,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip)) < 0) return err; - /* Default routing of the virtual channels: all vchannels are routed - to the stereo output */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 0, 4, 0); - set_vmixer_gain(chip, 1, 5, 0); - set_vmixer_gain(chip, 0, 6, 0); - set_vmixer_gain(chip, 1, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c new file mode 100644 index 0000000..aebee27 --- /dev/null +++ b/sound/pci/echoaudio/indigoiox.c @@ -0,0 +1,109 @@ +/* + * ALSA driver for Echoaudio soundcards. + * Copyright (C) 2009 Giuliano Pochini <pochini@shiny.it> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA. + */ + +#define INDIGO_FAMILY +#define ECHOCARD_INDIGO_IOX +#define ECHOCARD_NAME "Indigo IOx" +#define ECHOCARD_HAS_MONITOR +#define ECHOCARD_HAS_SUPER_INTERLEAVE +#define ECHOCARD_HAS_VMIXER +#define ECHOCARD_HAS_STEREO_BIG_ENDIAN32 + +/* Pipe indexes */ +#define PX_ANALOG_OUT 0 /* 8 */ +#define PX_DIGITAL_OUT 8 /* 0 */ +#define PX_ANALOG_IN 8 /* 2 */ +#define PX_DIGITAL_IN 10 /* 0 */ +#define PX_NUM 10 + +/* Bus indexes */ +#define BX_ANALOG_OUT 0 /* 2 */ +#define BX_DIGITAL_OUT 2 /* 0 */ +#define BX_ANALOG_IN 2 /* 2 */ +#define BX_DIGITAL_IN 4 /* 0 */ +#define BX_NUM 4 + + +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/pci.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/firmware.h> +#include <linux/io.h> +#include <sound/core.h> +#include <sound/info.h> +#include <sound/control.h> +#include <sound/tlv.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/asoundef.h> +#include <sound/initval.h> +#include <asm/atomic.h> +#include "echoaudio.h" + +MODULE_FIRMWARE("ea/loader_dsp.fw"); +MODULE_FIRMWARE("ea/indigo_iox_dsp.fw"); + +#define FW_361_LOADER 0 +#define FW_INDIGO_IOX_DSP 1 + +static const struct firmware card_fw[] = { + {0, "loader_dsp.fw"}, + {0, "indigo_iox_dsp.fw"} +}; + +static struct pci_device_id snd_echo_ids[] = { + {0x1057, 0x3410, 0xECC0, 0x00D0, 0, 0, 0}, /* Indigo IOx */ + {0,} +}; + +static struct snd_pcm_hardware pcm_hardware_skel = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_3LE | + SNDRV_PCM_FMTBIT_S32_LE | + SNDRV_PCM_FMTBIT_S32_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + .rate_min = 32000, + .rate_max = 96000, + .channels_min = 1, + .channels_max = 8, + .buffer_bytes_max = 262144, + .period_bytes_min = 32, + .period_bytes_max = 131072, + .periods_min = 2, + .periods_max = 220, +}; + +#include "indigoiox_dsp.c" +#include "indigo_express_dsp.c" +#include "echoaudio_dsp.c" +#include "echoaudio.c" + diff --git a/sound/pci/echoaudio/indigoiox_dsp.c b/sound/pci/echoaudio/indigoiox_dsp.c new file mode 100644 index 0000000..f357521 --- /dev/null +++ b/sound/pci/echoaudio/indigoiox_dsp.c @@ -0,0 +1,68 @@ +/************************************************************************ + +This file is part of Echo Digital Audio's generic driver library. +Copyright Echo Digital Audio Corporation (c) 1998 - 2005 +All rights reserved +www.echoaudio.com + +This library is free software; you can redistribute it and/or +modify it under the terms of the GNU Lesser General Public +License as published by the Free Software Foundation; either +version 2.1 of the License, or (at your option) any later version. + +This library is distributed in the hope that it will be useful, +but WITHOUT ANY WARRANTY; without even the implied warranty of +MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU +Lesser General Public License for more details. + +You should have received a copy of the GNU Lesser General Public +License along with this library; if not, write to the Free Software +Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + +************************************************************************* + + Translation from C++ and adaptation for use in ALSA-Driver + were made by Giuliano Pochini <pochini@shiny.it> + +*************************************************************************/ + +static int update_vmixer_level(struct echoaudio *chip); +static int set_vmixer_gain(struct echoaudio *chip, u16 output, + u16 pipe, int gain); + + +static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) +{ + int err; + + DE_INIT(("init_hw() - Indigo IOx\n")); + if (snd_BUG_ON((subdevice_id & 0xfff0) != INDIGO_IOX)) + return -ENODEV; + + err = init_dsp_comm_page(chip); + if (err < 0) { + DE_INIT(("init_hw - could not initialize DSP comm page\n")); + return err; + } + + chip->device_id = device_id; + chip->subdevice_id = subdevice_id; + chip->bad_board = TRUE; + chip->dsp_code_to_load = &card_fw[FW_INDIGO_IOX_DSP]; + /* Since this card has no ASIC, mark it as loaded so everything + works OK */ + chip->asic_loaded = TRUE; + chip->input_clock_types = ECHO_CLOCK_BIT_INTERNAL; + + err = load_firmware(chip); + if (err < 0) + return err; + chip->bad_board = FALSE; + + err = init_line_levels(chip); + if (err < 0) + return err; + + DE_INIT(("init_hw done\n")); + return err; +} diff --git a/sound/pci/echoaudio/layla20_dsp.c b/sound/pci/echoaudio/layla20_dsp.c index ede75c6..83750e9 100644 --- a/sound/pci/echoaudio/layla20_dsp.c +++ b/sound/pci/echoaudio/layla20_dsp.c @@ -284,10 +284,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof) DE_ACT(("set_professional_spdif %d\n", prof)); if (prof) chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + ~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); chip->professional_spdif = prof; return update_flags(chip); } diff --git a/sound/pci/echoaudio/mia_dsp.c b/sound/pci/echoaudio/mia_dsp.c index 2273866..5514051 100644 --- a/sound/pci/echoaudio/mia_dsp.c +++ b/sound/pci/echoaudio/mia_dsp.c @@ -69,18 +69,6 @@ static int init_hw(struct echoaudio *chip, u16 device_id, u16 subdevice_id) if ((err = init_line_levels(chip))) return err; - /* Default routing of the virtual channels: vchannels 0-3 go to analog - outputs and vchannels 4-7 go to S/PDIF outputs */ - set_vmixer_gain(chip, 0, 0, 0); - set_vmixer_gain(chip, 1, 1, 0); - set_vmixer_gain(chip, 0, 2, 0); - set_vmixer_gain(chip, 1, 3, 0); - set_vmixer_gain(chip, 2, 4, 0); - set_vmixer_gain(chip, 3, 5, 0); - set_vmixer_gain(chip, 2, 6, 0); - set_vmixer_gain(chip, 3, 7, 0); - err = update_vmixer_level(chip); - DE_INIT(("init_hw done\n")); return err; } @@ -222,10 +210,10 @@ static int set_professional_spdif(struct echoaudio *chip, char prof) DE_ACT(("set_professional_spdif %d\n", prof)); if (prof) chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); + ~cpu_to_le32(DSP_FLAG_PROFESSIONAL_SPDIF); chip->professional_spdif = prof; return update_flags(chip); } diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index 77bf2a8..a953d14 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -44,10 +44,10 @@ static int enable_midi_input(struct echoaudio *chip, char enable) if (enable) { chip->mtc_state = MIDI_IN_STATE_NORMAL; chip->comm_page->flags |= - __constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT); + cpu_to_le32(DSP_FLAG_MIDI_INPUT); } else chip->comm_page->flags &= - ~__constant_cpu_to_le32(DSP_FLAG_MIDI_INPUT); + ~cpu_to_le32(DSP_FLAG_MIDI_INPUT); clear_handshake(chip); return send_vector(chip, DSP_VC_UPDATE_FLAGS); diff --git a/sound/pci/emu10k1/emu10k1_callback.c b/sound/pci/emu10k1/emu10k1_callback.c index 0e649dc..7ef949d9 100644 --- a/sound/pci/emu10k1/emu10k1_callback.c +++ b/sound/pci/emu10k1/emu10k1_callback.c @@ -103,7 +103,10 @@ snd_emu10k1_synth_get_voice(struct snd_emu10k1 *hw) int ch; vp = &emu->voices[best[i].voice]; if ((ch = vp->ch) < 0) { - //printk("synth_get_voice: ch < 0 (%d) ??", i); + /* + printk(KERN_WARNING + "synth_get_voice: ch < 0 (%d) ??", i); + */ continue; } vp->emu->num_voices--; @@ -335,7 +338,7 @@ start_voice(struct snd_emux_voice *vp) return -EINVAL; emem->map_locked++; if (snd_emu10k1_memblk_map(hw, emem) < 0) { - // printk("emu: cannot map!\n"); + /* printk(KERN_ERR "emu: cannot map!\n"); */ return -ENOMEM; } mapped_offset = snd_emu10k1_memblk_offset(emem) >> 1; diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 7958006..f18bd62 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -711,8 +711,7 @@ static int snd_emu1010_load_firmware(struct snd_emu10k1 *emu, const char *filena static int emu1010_firmware_thread(void *data) { struct snd_emu10k1 *emu = data; - int tmp, tmp2; - int reg; + u32 tmp, tmp2, reg; int err; for (;;) { @@ -758,7 +757,8 @@ static int emu1010_firmware_thread(void *data) snd_printk(KERN_INFO "emu1010: Audio Dock Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_DOCK_MAJOR_REV, &tmp); snd_emu1010_fpga_read(emu, EMU_DOCK_MINOR_REV, &tmp2); - snd_printk("Audio Dock ver:%d.%d\n", tmp, tmp2); + snd_printk(KERN_INFO "Audio Dock ver: %u.%u\n", + tmp, tmp2); /* Sync clocking between 1010 and Dock */ /* Allow DLL to settle */ msleep(10); @@ -804,8 +804,7 @@ static int emu1010_firmware_thread(void *data) static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) { unsigned int i; - int tmp, tmp2; - int reg; + u32 tmp, tmp2, reg; int err; const char *filename = NULL; @@ -887,7 +886,7 @@ static int snd_emu10k1_emu1010_init(struct snd_emu10k1 *emu) snd_printk(KERN_INFO "emu1010: Hana Firmware loaded\n"); snd_emu1010_fpga_read(emu, EMU_HANA_MAJOR_REV, &tmp); snd_emu1010_fpga_read(emu, EMU_HANA_MINOR_REV, &tmp2); - snd_printk("emu1010: Hana version: %d.%d\n", tmp, tmp2); + snd_printk(KERN_INFO "emu1010: Hana version: %u.%u\n", tmp, tmp2); /* Enable 48Volt power to Audio Dock */ snd_emu1010_fpga_write(emu, EMU_HANA_DOCK_PWR, EMU_HANA_DOCK_PWR_ON); @@ -1528,6 +1527,7 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0151_chip = 1, .spk71 = 1, .spdif_bug = 1, + .invert_shared_spdif = 1, /* digital/analog switch swapped */ .ac97_chip = 1} , {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x10021102, .driver = "Audigy2", .name = "SB Audigy 2 Platinum [SB0240P]", diff --git a/sound/pci/emu10k1/emu10k1x.c b/sound/pci/emu10k1/emu10k1x.c index 31542ad..1970f0e 100644 --- a/sound/pci/emu10k1/emu10k1x.c +++ b/sound/pci/emu10k1/emu10k1x.c @@ -897,8 +897,8 @@ static int __devinit snd_emu10k1x_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { snd_printk(KERN_ERR "error to set 28bit mask DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 7dba08f..191e1cd 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1519,7 +1519,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) /* A_PUT_STEREO_OUTPUT(A_EXTOUT_FRONT_L, A_EXTOUT_FRONT_R, playback + SND_EMU10K1_PLAYBACK_CHANNELS); */ if (emu->card_capabilities->emu_model) { /* EMU1010 Outputs from PCM Front, Rear, Center, LFE, Side */ - snd_printk("EMU outputs on\n"); + snd_printk(KERN_INFO "EMU outputs on\n"); for (z = 0; z < 8; z++) { if (emu->card_capabilities->ca0108_chip) { A_OP(icode, &ptr, iACC3, A3_EMU32OUT(z), A_GPR(playback + SND_EMU10K1_PLAYBACK_CHANNELS + z), A_C_00000000, A_C_00000000); @@ -1567,7 +1567,7 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) if (emu->card_capabilities->emu_model) { if (emu->card_capabilities->ca0108_chip) { - snd_printk("EMU2 inputs on\n"); + snd_printk(KERN_INFO "EMU2 inputs on\n"); for (z = 0; z < 0x10; z++) { snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, @@ -1575,10 +1575,13 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) A_FXBUS2(z*2) ); } } else { - snd_printk("EMU inputs on\n"); + snd_printk(KERN_INFO "EMU inputs on\n"); /* Capture 16 (originally 8) channels of S32_LE sound */ - /* printk("emufx.c: gpr=0x%x, tmp=0x%x\n",gpr, tmp); */ + /* + printk(KERN_DEBUG "emufx.c: gpr=0x%x, tmp=0x%x\n", + gpr, tmp); + */ /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ /* A_P16VIN(0) is delayed by one sample, * so all other A_P16VIN channels will need to also be delayed diff --git a/sound/pci/emu10k1/emupcm.c b/sound/pci/emu10k1/emupcm.c index cf9276dd..78f62fd 100644 --- a/sound/pci/emu10k1/emupcm.c +++ b/sound/pci/emu10k1/emupcm.c @@ -44,7 +44,7 @@ static void snd_emu10k1_pcm_interrupt(struct snd_emu10k1 *emu, if (epcm->substream == NULL) return; #if 0 - printk("IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n", + printk(KERN_DEBUG "IRQ: position = 0x%x, period = 0x%x, size = 0x%x\n", epcm->substream->runtime->hw->pointer(emu, epcm->substream), snd_pcm_lib_period_bytes(epcm->substream), snd_pcm_lib_buffer_bytes(epcm->substream)); @@ -146,7 +146,11 @@ static int snd_emu10k1_pcm_channel_alloc(struct snd_emu10k1_pcm * epcm, int voic 1, &epcm->extra); if (err < 0) { - /* printk("pcm_channel_alloc: failed extra: voices=%d, frame=%d\n", voices, frame); */ + /* + printk(KERN_DEBUG "pcm_channel_alloc: " + "failed extra: voices=%d, frame=%d\n", + voices, frame); + */ for (i = 0; i < voices; i++) { snd_emu10k1_voice_free(epcm->emu, epcm->voices[i]); epcm->voices[i] = NULL; @@ -737,7 +741,10 @@ static int snd_emu10k1_playback_trigger(struct snd_pcm_substream *substream, struct snd_emu10k1_pcm_mixer *mix; int result = 0; - /* printk("trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n", (int)emu, cmd, substream->ops->pointer(substream)); */ + /* + printk(KERN_DEBUG "trigger - emu10k1 = 0x%x, cmd = %i, pointer = %i\n", + (int)emu, cmd, substream->ops->pointer(substream)) + */ spin_lock(&emu->reg_lock); switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -786,7 +793,10 @@ static int snd_emu10k1_capture_trigger(struct snd_pcm_substream *substream, /* hmm this should cause full and half full interrupt to be raised? */ outl(epcm->capture_ipr, emu->port + IPR); snd_emu10k1_intr_enable(emu, epcm->capture_inte); - /* printk("adccr = 0x%x, adcbs = 0x%x\n", epcm->adccr, epcm->adcbs); */ + /* + printk(KERN_DEBUG "adccr = 0x%x, adcbs = 0x%x\n", + epcm->adccr, epcm->adcbs); + */ switch (epcm->type) { case CAPTURE_AC97ADC: snd_emu10k1_ptr_write(emu, ADCCR, 0, epcm->capture_cr_val); @@ -857,7 +867,11 @@ static snd_pcm_uframes_t snd_emu10k1_playback_pointer(struct snd_pcm_substream * ptr -= runtime->buffer_size; } #endif - /* printk("ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", ptr, runtime->buffer_size, runtime->period_size); */ + /* + printk(KERN_DEBUG + "ptr = 0x%x, buffer_size = 0x%x, period_size = 0x%x\n", + ptr, runtime->buffer_size, runtime->period_size); + */ return ptr; } @@ -1546,7 +1560,11 @@ static void snd_emu10k1_fx8010_playback_tram_poke1(unsigned short *dst_left, unsigned int count, unsigned int tram_shift) { - /* printk("tram_poke1: dst_left = 0x%p, dst_right = 0x%p, src = 0x%p, count = 0x%x\n", dst_left, dst_right, src, count); */ + /* + printk(KERN_DEBUG "tram_poke1: dst_left = 0x%p, dst_right = 0x%p, " + "src = 0x%p, count = 0x%x\n", + dst_left, dst_right, src, count); + */ if ((tram_shift & 1) == 0) { while (count--) { *dst_left-- = *src++; @@ -1623,7 +1641,12 @@ static int snd_emu10k1_fx8010_playback_prepare(struct snd_pcm_substream *substre struct snd_emu10k1_fx8010_pcm *pcm = &emu->fx8010.pcm[substream->number]; unsigned int i; - /* printk("prepare: etram_pages = 0x%p, dma_area = 0x%x, buffer_size = 0x%x (0x%x)\n", emu->fx8010.etram_pages, runtime->dma_area, runtime->buffer_size, runtime->buffer_size << 2); */ + /* + printk(KERN_DEBUG "prepare: etram_pages = 0x%p, dma_area = 0x%x, " + "buffer_size = 0x%x (0x%x)\n", + emu->fx8010.etram_pages, runtime->dma_area, + runtime->buffer_size, runtime->buffer_size << 2); + */ memset(&pcm->pcm_rec, 0, sizeof(pcm->pcm_rec)); pcm->pcm_rec.hw_buffer_size = pcm->buffer_size * 2; /* byte size */ pcm->pcm_rec.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); diff --git a/sound/pci/emu10k1/io.c b/sound/pci/emu10k1/io.c index b5a802b..4bfc31d 100644 --- a/sound/pci/emu10k1/io.c +++ b/sound/pci/emu10k1/io.c @@ -226,7 +226,9 @@ int snd_emu10k1_i2c_write(struct snd_emu10k1 *emu, break; if (timeout > 1000) { - snd_printk("emu10k1:I2C:timeout status=0x%x\n", status); + snd_printk(KERN_WARNING + "emu10k1:I2C:timeout status=0x%x\n", + status); break; } } diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 749a21b..e617aca 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -168,7 +168,7 @@ static void snd_p16v_pcm_free_substream(struct snd_pcm_runtime *runtime) struct snd_emu10k1_pcm *epcm = runtime->private_data; if (epcm) { - //snd_printk("epcm free: %p\n", epcm); + /* snd_printk(KERN_DEBUG "epcm free: %p\n", epcm); */ kfree(epcm); } } @@ -183,14 +183,16 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - //snd_printk("epcm kcalloc: %p\n", epcm); + /* snd_printk(KERN_DEBUG "epcm kcalloc: %p\n", epcm); */ if (epcm == NULL) return -ENOMEM; epcm->emu = emu; epcm->substream = substream; - //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); - + /* + snd_printk(KERN_DEBUG "epcm device=%d, channel_id=%d\n", + substream->pcm->device, channel_id); + */ runtime->private_data = epcm; runtime->private_free = snd_p16v_pcm_free_substream; @@ -200,10 +202,15 @@ static int snd_p16v_pcm_open_playback_channel(struct snd_pcm_substream *substrea channel->number = channel_id; channel->use=1; - //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use); - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); - //channel->interrupt = snd_p16v_pcm_channel_interrupt; - channel->epcm=epcm; +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "p16v: open channel_id=%d, channel=%p, use=0x%x\n", + channel_id, channel, channel->use); + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); +#endif /* debug */ + /* channel->interrupt = snd_p16v_pcm_channel_interrupt; */ + channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -224,14 +231,16 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream int err; epcm = kzalloc(sizeof(*epcm), GFP_KERNEL); - //snd_printk("epcm kcalloc: %p\n", epcm); + /* snd_printk(KERN_DEBUG "epcm kcalloc: %p\n", epcm); */ if (epcm == NULL) return -ENOMEM; epcm->emu = emu; epcm->substream = substream; - //snd_printk("epcm device=%d, channel_id=%d\n", substream->pcm->device, channel_id); - + /* + snd_printk(KERN_DEBUG "epcm device=%d, channel_id=%d\n", + substream->pcm->device, channel_id); + */ runtime->private_data = epcm; runtime->private_free = snd_p16v_pcm_free_substream; @@ -241,10 +250,15 @@ static int snd_p16v_pcm_open_capture_channel(struct snd_pcm_substream *substream channel->number = channel_id; channel->use=1; - //snd_printk("p16v: open channel_id=%d, channel=%p, use=0x%x\n", channel_id, channel, channel->use); - //printk("open:channel_id=%d, chip=%p, channel=%p\n",channel_id, chip, channel); - //channel->interrupt = snd_p16v_pcm_channel_interrupt; - channel->epcm=epcm; +#if 0 /* debug */ + snd_printk(KERN_DEBUG + "p16v: open channel_id=%d, channel=%p, use=0x%x\n", + channel_id, channel, channel->use); + printk(KERN_DEBUG "open:channel_id=%d, chip=%p, channel=%p\n", + channel_id, chip, channel); +#endif /* debug */ + /* channel->interrupt = snd_p16v_pcm_channel_interrupt; */ + channel->epcm = epcm; if ((err = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS)) < 0) return err; @@ -334,9 +348,19 @@ static int snd_p16v_pcm_prepare_playback(struct snd_pcm_substream *substream) int i; u32 tmp; - //snd_printk("prepare:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, periods=%u, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, runtime->periods, frames_to_bytes(runtime, 1)); - //snd_printk("dma_addr=%x, dma_area=%p, table_base=%p\n",runtime->dma_addr, runtime->dma_area, table_base); - //snd_printk("dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n",emu->p16v_buffer.addr, emu->p16v_buffer.area, emu->p16v_buffer.bytes); +#if 0 /* debug */ + snd_printk(KERN_DEBUG "prepare:channel_number=%d, rate=%d, " + "format=0x%x, channels=%d, buffer_size=%ld, " + "period_size=%ld, periods=%u, frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, + runtime->periods, frames_to_bytes(runtime, 1)); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, table_base=%p\n", + runtime->dma_addr, runtime->dma_area, table_base); + snd_printk(KERN_DEBUG "dma_addr=%x, dma_area=%p, dma_bytes(size)=%x\n", + emu->p16v_buffer.addr, emu->p16v_buffer.area, + emu->p16v_buffer.bytes); +#endif /* debug */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel); switch (runtime->rate) { case 44100: @@ -379,7 +403,15 @@ static int snd_p16v_pcm_prepare_capture(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; int channel = substream->pcm->device - emu->p16v_device_offset; u32 tmp; - //printk("prepare capture:channel_number=%d, rate=%d, format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, frames_to_bytes=%d\n",channel, runtime->rate, runtime->format, runtime->channels, runtime->buffer_size, runtime->period_size, frames_to_bytes(runtime, 1)); + + /* + printk(KERN_DEBUG "prepare capture:channel_number=%d, rate=%d, " + "format=0x%x, channels=%d, buffer_size=%ld, period_size=%ld, " + "frames_to_bytes=%d\n", + channel, runtime->rate, runtime->format, runtime->channels, + runtime->buffer_size, runtime->period_size, + frames_to_bytes(runtime, 1)); + */ tmp = snd_emu10k1_ptr_read(emu, A_SPDIF_SAMPLERATE, channel); switch (runtime->rate) { case 44100: @@ -459,13 +491,13 @@ static int snd_p16v_pcm_trigger_playback(struct snd_pcm_substream *substream, runtime = s->runtime; epcm = runtime->private_data; channel = substream->pcm->device-emu->p16v_device_offset; - //snd_printk("p16v channel=%d\n",channel); + /* snd_printk(KERN_DEBUG "p16v channel=%d\n", channel); */ epcm->running = running; basic |= (0x1<<channel); inte |= (INTE2_PLAYBACK_CH_0_LOOP<<channel); snd_pcm_trigger_done(s, substream); } - //snd_printk("basic=0x%x, inte=0x%x\n",basic, inte); + /* snd_printk(KERN_DEBUG "basic=0x%x, inte=0x%x\n", basic, inte); */ switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -558,8 +590,13 @@ snd_p16v_pcm_pointer_capture(struct snd_pcm_substream *substream) ptr -= runtime->buffer_size; printk(KERN_WARNING "buffer capture limited!\n"); } - //printk("ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", ptr1, ptr2, ptr, (int)runtime->buffer_size, (int)runtime->period_size, (int)runtime->frame_bits, (int)runtime->rate); - + /* + printk(KERN_DEBUG "ptr1 = 0x%lx, ptr2=0x%lx, ptr=0x%lx, " + "buffer_size = 0x%x, period_size = 0x%x, bits=%d, rate=%d\n", + ptr1, ptr2, ptr, (int)runtime->buffer_size, + (int)runtime->period_size, (int)runtime->frame_bits, + (int)runtime->rate); + */ return ptr; } @@ -592,7 +629,10 @@ int snd_p16v_free(struct snd_emu10k1 *chip) // release the data if (chip->p16v_buffer.area) { snd_dma_free_pages(&chip->p16v_buffer); - //snd_printk("period lables free: %p\n", &chip->p16v_buffer); + /* + snd_printk(KERN_DEBUG "period lables free: %p\n", + &chip->p16v_buffer); + */ } return 0; } @@ -604,7 +644,7 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * int err; int capture=1; - //snd_printk("snd_p16v_pcm called. device=%d\n", device); + /* snd_printk("KERN_DEBUG snd_p16v_pcm called. device=%d\n", device); */ emu->p16v_device_offset = device; if (rpcm) *rpcm = NULL; @@ -631,7 +671,10 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * snd_dma_pci_data(emu->pci), ((65536 - 64) * 8), ((65536 - 64) * 8))) < 0) return err; - //snd_printk("preallocate playback substream: err=%d\n", err); + /* + snd_printk(KERN_DEBUG + "preallocate playback substream: err=%d\n", err); + */ } for (substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; @@ -642,7 +685,10 @@ int __devinit snd_p16v_pcm(struct snd_emu10k1 *emu, int device, struct snd_pcm * snd_dma_pci_data(emu->pci), 65536 - 64, 65536 - 64)) < 0) return err; - //snd_printk("preallocate capture substream: err=%d\n", err); + /* + snd_printk(KERN_DEBUG + "preallocate capture substream: err=%d\n", err); + */ } if (rpcm) diff --git a/sound/pci/emu10k1/voice.c b/sound/pci/emu10k1/voice.c index d7300a1..20b8da2 100644 --- a/sound/pci/emu10k1/voice.c +++ b/sound/pci/emu10k1/voice.c @@ -53,7 +53,10 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, *rvoice = NULL; first_voice = last_voice = 0; for (i = emu->next_free_voice, j = 0; j < NUM_G ; i += number, j += number) { - // printk("i %d j %d next free %d!\n", i, j, emu->next_free_voice); + /* + printk(KERN_DEBUG "i %d j %d next free %d!\n", + i, j, emu->next_free_voice); + */ i %= NUM_G; /* stereo voices must be even/odd */ @@ -71,7 +74,7 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, } } if (!skip) { - // printk("allocated voice %d\n", i); + /* printk(KERN_DEBUG "allocated voice %d\n", i); */ first_voice = i; last_voice = (i + number) % NUM_G; emu->next_free_voice = last_voice; @@ -84,7 +87,10 @@ static int voice_alloc(struct snd_emu10k1 *emu, int type, int number, for (i = 0; i < number; i++) { voice = &emu->voices[(first_voice + i) % NUM_G]; - // printk("voice alloc - %i, %i of %i\n", voice->number, idx-first_voice+1, number); + /* + printk(kERN_DEBUG "voice alloc - %i, %i of %i\n", + voice->number, idx-first_voice+1, number); + */ voice->use = 1; switch (type) { case EMU10K1_PCM: diff --git a/sound/pci/ens1370.c b/sound/pci/ens1370.c index e00614c..18f4d1e 100644 --- a/sound/pci/ens1370.c +++ b/sound/pci/ens1370.c @@ -584,7 +584,8 @@ static void snd_es1370_codec_write(struct snd_ak4531 *ak4531, unsigned long end_time = jiffies + HZ / 10; #if 0 - printk("CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n", + printk(KERN_DEBUG + "CODEC WRITE: reg = 0x%x, val = 0x%x (0x%x), creg = 0x%x\n", reg, val, ES_1370_CODEC_WRITE(reg, val), ES_REG(ensoniq, 1370_CODEC)); #endif do { diff --git a/sound/pci/es1938.c b/sound/pci/es1938.c index 34a78af..fbd2ac0 100644 --- a/sound/pci/es1938.c +++ b/sound/pci/es1938.c @@ -1608,8 +1608,8 @@ static int __devinit snd_es1938_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; /* check, if we can restrict PCI DMA transfers to 24 bits */ - if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(24)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(24)) < 0) { snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; @@ -1673,18 +1673,22 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id) status = inb(SLIO_REG(chip, IRQCONTROL)); #if 0 - printk("Es1938debug - interrupt status: =0x%x\n", status); + printk(KERN_DEBUG "Es1938debug - interrupt status: =0x%x\n", status); #endif /* AUDIO 1 */ if (status & 0x10) { #if 0 - printk("Es1938debug - AUDIO channel 1 interrupt\n"); - printk("Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 interrupt\n"); + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 DMAC DMA count: %u\n", inw(SLDM_REG(chip, DMACOUNT))); - printk("Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 DMAC DMA base: %u\n", inl(SLDM_REG(chip, DMAADDR))); - printk("Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 1 DMAC DMA status: 0x%x\n", inl(SLDM_REG(chip, DMASTATUS))); #endif /* clear irq */ @@ -1699,10 +1703,13 @@ static irqreturn_t snd_es1938_interrupt(int irq, void *dev_id) /* AUDIO 2 */ if (status & 0x20) { #if 0 - printk("Es1938debug - AUDIO channel 2 interrupt\n"); - printk("Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 2 interrupt\n"); + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 2 DMAC DMA count: %u\n", inw(SLIO_REG(chip, AUDIO2DMACOUNT))); - printk("Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n", + printk(KERN_DEBUG + "Es1938debug - AUDIO channel 2 DMAC DMA base: %u\n", inl(SLIO_REG(chip, AUDIO2DMAADDR))); #endif diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index dc97e81..a11f453 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -2539,8 +2539,8 @@ static int __devinit snd_es1968_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; /* check, if we can restrict PCI DMA transfers to 28 bits */ - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 960fd79..4de5bac 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -138,6 +138,7 @@ void snd_hda_detach_beep_device(struct hda_codec *codec) input_unregister_device(beep->dev); kfree(beep); + codec->beep = NULL; } } EXPORT_SYMBOL_HDA(snd_hda_detach_beep_device); diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index b9679f0..51bf6a5 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -39,7 +39,7 @@ struct hda_beep { int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); #else -#define snd_hda_attach_beep_device(...) +#define snd_hda_attach_beep_device(...) 0 #define snd_hda_detach_beep_device(...) #endif #endif diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 3c596da..a4e5e59 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -487,7 +487,6 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, { struct hda_bus *bus; int err; - char qname[8]; static struct snd_device_ops dev_ops = { .dev_register = snd_hda_bus_dev_register, .dev_free = snd_hda_bus_dev_free, @@ -517,10 +516,12 @@ int /*__devinit*/ snd_hda_bus_new(struct snd_card *card, mutex_init(&bus->cmd_mutex); INIT_LIST_HEAD(&bus->codec_list); - snprintf(qname, sizeof(qname), "hda%d", card->number); - bus->workq = create_workqueue(qname); + snprintf(bus->workq_name, sizeof(bus->workq_name), + "hd-audio%d", card->number); + bus->workq = create_singlethread_workqueue(bus->workq_name); if (!bus->workq) { - snd_printk(KERN_ERR "cannot create workqueue %s\n", qname); + snd_printk(KERN_ERR "cannot create workqueue %s\n", + bus->workq_name); kfree(bus); return -ENOMEM; } @@ -646,9 +647,9 @@ static void /*__devinit*/ setup_fg_nodes(struct hda_codec *codec) total_nodes = snd_hda_get_sub_nodes(codec, AC_NODE_ROOT, &nid); for (i = 0; i < total_nodes; i++, nid++) { - unsigned int func; - func = snd_hda_param_read(codec, nid, AC_PAR_FUNCTION_TYPE); - switch (func & 0xff) { + codec->function_id = snd_hda_param_read(codec, nid, + AC_PAR_FUNCTION_TYPE) & 0xff; + switch (codec->function_id) { case AC_GRP_AUDIO_FUNCTION: codec->afg = nid; break; @@ -681,11 +682,140 @@ static int read_widget_caps(struct hda_codec *codec, hda_nid_t fg_node) return 0; } +/* read all pin default configurations and save codec->init_pins */ +static int read_pin_defaults(struct hda_codec *codec) +{ + int i; + hda_nid_t nid = codec->start_nid; + + for (i = 0; i < codec->num_nodes; i++, nid++) { + struct hda_pincfg *pin; + unsigned int wcaps = get_wcaps(codec, nid); + unsigned int wid_type = (wcaps & AC_WCAP_TYPE) >> + AC_WCAP_TYPE_SHIFT; + if (wid_type != AC_WID_PIN) + continue; + pin = snd_array_new(&codec->init_pins); + if (!pin) + return -ENOMEM; + pin->nid = nid; + pin->cfg = snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, 0); + } + return 0; +} + +/* look up the given pin config list and return the item matching with NID */ +static struct hda_pincfg *look_up_pincfg(struct hda_codec *codec, + struct snd_array *array, + hda_nid_t nid) +{ + int i; + for (i = 0; i < array->used; i++) { + struct hda_pincfg *pin = snd_array_elem(array, i); + if (pin->nid == nid) + return pin; + } + return NULL; +} + +/* write a config value for the given NID */ +static void set_pincfg(struct hda_codec *codec, hda_nid_t nid, + unsigned int cfg) +{ + int i; + for (i = 0; i < 4; i++) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, + cfg & 0xff); + cfg >>= 8; + } +} + +/* set the current pin config value for the given NID. + * the value is cached, and read via snd_hda_codec_get_pincfg() + */ +int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, + hda_nid_t nid, unsigned int cfg) +{ + struct hda_pincfg *pin; + unsigned int oldcfg; + + oldcfg = snd_hda_codec_get_pincfg(codec, nid); + pin = look_up_pincfg(codec, list, nid); + if (!pin) { + pin = snd_array_new(list); + if (!pin) + return -ENOMEM; + pin->nid = nid; + } + pin->cfg = cfg; + + /* change only when needed; e.g. if the pincfg is already present + * in user_pins[], don't write it + */ + cfg = snd_hda_codec_get_pincfg(codec, nid); + if (oldcfg != cfg) + set_pincfg(codec, nid, cfg); + return 0; +} + +int snd_hda_codec_set_pincfg(struct hda_codec *codec, + hda_nid_t nid, unsigned int cfg) +{ + return snd_hda_add_pincfg(codec, &codec->driver_pins, nid, cfg); +} +EXPORT_SYMBOL_HDA(snd_hda_codec_set_pincfg); + +/* get the current pin config value of the given pin NID */ +unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_pincfg *pin; + +#ifdef CONFIG_SND_HDA_HWDEP + pin = look_up_pincfg(codec, &codec->user_pins, nid); + if (pin) + return pin->cfg; +#endif + pin = look_up_pincfg(codec, &codec->driver_pins, nid); + if (pin) + return pin->cfg; + pin = look_up_pincfg(codec, &codec->init_pins, nid); + if (pin) + return pin->cfg; + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_codec_get_pincfg); + +/* restore all current pin configs */ +static void restore_pincfgs(struct hda_codec *codec) +{ + int i; + for (i = 0; i < codec->init_pins.used; i++) { + struct hda_pincfg *pin = snd_array_elem(&codec->init_pins, i); + set_pincfg(codec, pin->nid, + snd_hda_codec_get_pincfg(codec, pin->nid)); + } +} static void init_hda_cache(struct hda_cache_rec *cache, unsigned int record_size); static void free_hda_cache(struct hda_cache_rec *cache); +/* restore the initial pin cfgs and release all pincfg lists */ +static void restore_init_pincfgs(struct hda_codec *codec) +{ + /* first free driver_pins and user_pins, then call restore_pincfg + * so that only the values in init_pins are restored + */ + snd_array_free(&codec->driver_pins); +#ifdef CONFIG_SND_HDA_HWDEP + snd_array_free(&codec->user_pins); +#endif + restore_pincfgs(codec); + snd_array_free(&codec->init_pins); +} + /* * codec destructor */ @@ -693,6 +823,7 @@ static void snd_hda_codec_free(struct hda_codec *codec) { if (!codec) return; + restore_init_pincfgs(codec); #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); flush_workqueue(codec->bus->workq); @@ -711,6 +842,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state); + /** * snd_hda_codec_new - create a HDA codec * @bus: the bus to assign @@ -750,6 +884,8 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); snd_array_init(&codec->mixers, sizeof(struct snd_kcontrol *), 32); + snd_array_init(&codec->init_pins, sizeof(struct hda_pincfg), 16); + snd_array_init(&codec->driver_pins, sizeof(struct hda_pincfg), 16); if (codec->bus->modelname) { codec->modelname = kstrdup(codec->bus->modelname, GFP_KERNEL); if (!codec->modelname) { @@ -786,15 +922,18 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr setup_fg_nodes(codec); if (!codec->afg && !codec->mfg) { snd_printdd("hda_codec: no AFG or MFG node found\n"); - snd_hda_codec_free(codec); - return -ENODEV; + err = -ENODEV; + goto error; } - if (read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg) < 0) { + err = read_widget_caps(codec, codec->afg ? codec->afg : codec->mfg); + if (err < 0) { snd_printk(KERN_ERR "hda_codec: cannot malloc\n"); - snd_hda_codec_free(codec); - return -ENOMEM; + goto error; } + err = read_pin_defaults(codec); + if (err < 0) + goto error; if (!codec->subsystem_id) { hda_nid_t nid = codec->afg ? codec->afg : codec->mfg; @@ -805,12 +944,15 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr if (bus->modelname) codec->modelname = kstrdup(bus->modelname, GFP_KERNEL); + /* power-up all before initialization */ + hda_set_power_state(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_D0); + if (do_init) { err = snd_hda_codec_configure(codec); - if (err < 0) { - snd_hda_codec_free(codec); - return err; - } + if (err < 0) + goto error; } snd_hda_codec_proc_new(codec); @@ -823,6 +965,10 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr if (codecp) *codecp = codec; return 0; + + error: + snd_hda_codec_free(codec); + return err; } EXPORT_SYMBOL_HDA(snd_hda_codec_new); @@ -906,6 +1052,7 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); /* FIXME: more better hash key? */ #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) +#define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) #define INFO_AMP_CAPS (1<<0) #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) @@ -996,6 +1143,21 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, } EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_amp_info *info; + + info = get_alloc_amp_hash(codec, HDA_HASH_PINCAP_KEY(nid)); + if (!info) + return 0; + if (!info->head.val) { + info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + info->head.val |= INFO_AMP_CAPS; + } + return info->amp_caps; +} +EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); + /* * read the current volume to info * if the cache exists, read the cache value. @@ -1119,6 +1281,7 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, u16 nid = get_amp_nid(kcontrol); u8 chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); u32 caps; caps = query_amp_caps(codec, nid, dir); @@ -1130,6 +1293,8 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, kcontrol->id.name); return -EINVAL; } + if (ofs < caps) + caps -= ofs; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = chs == 3 ? 2 : 1; uinfo->value.integer.min = 0; @@ -1138,6 +1303,32 @@ int snd_hda_mixer_amp_volume_info(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_info); + +static inline unsigned int +read_amp_value(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, unsigned int ofs) +{ + unsigned int val; + val = snd_hda_codec_amp_read(codec, nid, ch, dir, idx); + val &= HDA_AMP_VOLMASK; + if (val >= ofs) + val -= ofs; + else + val = 0; + return val; +} + +static inline int +update_amp_value(struct hda_codec *codec, hda_nid_t nid, + int ch, int dir, int idx, unsigned int ofs, + unsigned int val) +{ + if (val > 0) + val += ofs; + return snd_hda_codec_amp_update(codec, nid, ch, dir, idx, + HDA_AMP_VOLMASK, val); +} + int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -1146,14 +1337,13 @@ int snd_hda_mixer_amp_volume_get(struct snd_kcontrol *kcontrol, int chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); int idx = get_amp_index(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); long *valp = ucontrol->value.integer.value; if (chs & 1) - *valp++ = snd_hda_codec_amp_read(codec, nid, 0, dir, idx) - & HDA_AMP_VOLMASK; + *valp++ = read_amp_value(codec, nid, 0, dir, idx, ofs); if (chs & 2) - *valp = snd_hda_codec_amp_read(codec, nid, 1, dir, idx) - & HDA_AMP_VOLMASK; + *valp = read_amp_value(codec, nid, 1, dir, idx, ofs); return 0; } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_volume_get); @@ -1166,18 +1356,17 @@ int snd_hda_mixer_amp_volume_put(struct snd_kcontrol *kcontrol, int chs = get_amp_channels(kcontrol); int dir = get_amp_direction(kcontrol); int idx = get_amp_index(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); long *valp = ucontrol->value.integer.value; int change = 0; snd_hda_power_up(codec); if (chs & 1) { - change = snd_hda_codec_amp_update(codec, nid, 0, dir, idx, - 0x7f, *valp); + change = update_amp_value(codec, nid, 0, dir, idx, ofs, *valp); valp++; } if (chs & 2) - change |= snd_hda_codec_amp_update(codec, nid, 1, dir, idx, - 0x7f, *valp); + change |= update_amp_value(codec, nid, 1, dir, idx, ofs, *valp); snd_hda_power_down(codec); return change; } @@ -1189,6 +1378,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, struct hda_codec *codec = snd_kcontrol_chip(kcontrol); hda_nid_t nid = get_amp_nid(kcontrol); int dir = get_amp_direction(kcontrol); + unsigned int ofs = get_amp_offset(kcontrol); u32 caps, val1, val2; if (size < 4 * sizeof(unsigned int)) @@ -1197,6 +1387,7 @@ int snd_hda_mixer_amp_tlv(struct snd_kcontrol *kcontrol, int op_flag, val2 = (caps & AC_AMPCAP_STEP_SIZE) >> AC_AMPCAP_STEP_SIZE_SHIFT; val2 = (val2 + 1) * 25; val1 = -((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT); + val1 += ofs; val1 = ((int)val1) * ((int)val2); if (put_user(SNDRV_CTL_TLVT_DB_SCALE, _tlv)) return -EFAULT; @@ -1267,7 +1458,6 @@ int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl) } EXPORT_SYMBOL_HDA(snd_hda_ctl_add); -#ifdef CONFIG_SND_HDA_RECONFIG /* Clear all controls assigned to the given codec */ void snd_hda_ctls_clear(struct hda_codec *codec) { @@ -1278,9 +1468,52 @@ void snd_hda_ctls_clear(struct hda_codec *codec) snd_array_free(&codec->mixers); } -void snd_hda_codec_reset(struct hda_codec *codec) +/* pseudo device locking + * toggle card->shutdown to allow/disallow the device access (as a hack) + */ +static int hda_lock_devices(struct snd_card *card) { - int i; + spin_lock(&card->files_lock); + if (card->shutdown) { + spin_unlock(&card->files_lock); + return -EINVAL; + } + card->shutdown = 1; + spin_unlock(&card->files_lock); + return 0; +} + +static void hda_unlock_devices(struct snd_card *card) +{ + spin_lock(&card->files_lock); + card->shutdown = 0; + spin_unlock(&card->files_lock); +} + +int snd_hda_codec_reset(struct hda_codec *codec) +{ + struct snd_card *card = codec->bus->card; + int i, pcm; + + if (hda_lock_devices(card) < 0) + return -EBUSY; + /* check whether the codec isn't used by any mixer or PCM streams */ + if (!list_empty(&card->ctl_files)) { + hda_unlock_devices(card); + return -EBUSY; + } + for (pcm = 0; pcm < codec->num_pcms; pcm++) { + struct hda_pcm *cpcm = &codec->pcm_info[pcm]; + if (!cpcm->pcm) + continue; + if (cpcm->pcm->streams[0].substream_opened || + cpcm->pcm->streams[1].substream_opened) { + hda_unlock_devices(card); + return -EBUSY; + } + } + + /* OK, let it free */ #ifdef CONFIG_SND_HDA_POWER_SAVE cancel_delayed_work(&codec->power_work); @@ -1290,8 +1523,7 @@ void snd_hda_codec_reset(struct hda_codec *codec) /* relase PCMs */ for (i = 0; i < codec->num_pcms; i++) { if (codec->pcm_info[i].pcm) { - snd_device_free(codec->bus->card, - codec->pcm_info[i].pcm); + snd_device_free(card, codec->pcm_info[i].pcm); clear_bit(codec->pcm_info[i].device, codec->bus->pcm_dev_bits); } @@ -1304,13 +1536,22 @@ void snd_hda_codec_reset(struct hda_codec *codec) free_hda_cache(&codec->cmd_cache); init_hda_cache(&codec->amp_cache, sizeof(struct hda_amp_info)); init_hda_cache(&codec->cmd_cache, sizeof(struct hda_cache_head)); + /* free only driver_pins so that init_pins + user_pins are restored */ + snd_array_free(&codec->driver_pins); + restore_pincfgs(codec); codec->num_pcms = 0; codec->pcm_info = NULL; codec->preset = NULL; + memset(&codec->patch_ops, 0, sizeof(codec->patch_ops)); + codec->slave_dig_outs = NULL; + codec->spdif_status_reset = 0; module_put(codec->owner); codec->owner = NULL; + + /* allow device access again */ + hda_unlock_devices(card); + return 0; } -#endif /* CONFIG_SND_HDA_RECONFIG */ /* create a virtual master control and add slaves */ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, @@ -1335,15 +1576,20 @@ int snd_hda_add_vmaster(struct hda_codec *codec, char *name, for (s = slaves; *s; s++) { struct snd_kcontrol *sctl; - - sctl = snd_hda_find_mixer_ctl(codec, *s); - if (!sctl) { - snd_printdd("Cannot find slave %s, skipped\n", *s); - continue; + int i = 0; + for (;;) { + sctl = _snd_hda_find_mixer_ctl(codec, *s, i); + if (!sctl) { + if (!i) + snd_printdd("Cannot find slave %s, " + "skipped\n", *s); + break; + } + err = snd_ctl_add_slave(kctl, sctl); + if (err < 0) + return err; + i++; } - err = snd_ctl_add_slave(kctl, sctl); - if (err < 0) - return err; } return 0; } @@ -1954,6 +2200,8 @@ int snd_hda_create_spdif_in_ctls(struct hda_codec *codec, hda_nid_t nid) } for (dig_mix = dig_in_ctls; dig_mix->name; dig_mix++) { kctl = snd_ctl_new1(dig_mix, codec); + if (!kctl) + return -ENOMEM; kctl->private_value = nid; err = snd_hda_ctl_add(codec, kctl); if (err < 0) @@ -2073,8 +2321,7 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, * don't power down the widget if it controls * eapd and EAPD_BTLENABLE is set. */ - pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_EAPD) { int eapd = snd_hda_codec_read(codec, nid, 0, @@ -2143,6 +2390,7 @@ static void hda_call_codec_resume(struct hda_codec *codec) hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); + restore_pincfgs(codec); /* restore all current pin configs */ hda_exec_init_verbs(codec); if (codec->patch_ops.resume) codec->patch_ops.resume(codec); @@ -2170,8 +2418,16 @@ int /*__devinit*/ snd_hda_build_controls(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { int err = snd_hda_codec_build_controls(codec); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot build controls" + "for #%d (error %d)\n", codec->addr, err); + err = snd_hda_codec_reset(codec); + if (err < 0) { + printk(KERN_ERR + "hda_codec: cannot revert codec\n"); + return err; + } + } } return 0; } @@ -2180,19 +2436,12 @@ EXPORT_SYMBOL_HDA(snd_hda_build_controls); int snd_hda_codec_build_controls(struct hda_codec *codec) { int err = 0; - /* fake as if already powered-on */ - hda_keep_power_on(codec); - /* then fire up */ - hda_set_power_state(codec, - codec->afg ? codec->afg : codec->mfg, - AC_PWRST_D0); hda_exec_init_verbs(codec); /* continue to initialize... */ if (codec->patch_ops.init) err = codec->patch_ops.init(codec); if (!err && codec->patch_ops.build_controls) err = codec->patch_ops.build_controls(codec); - snd_hda_power_down(codec); if (err < 0) return err; return 0; @@ -2305,12 +2554,11 @@ EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, u32 *ratesp, u64 *formatsp, unsigned int *bpsp) { - int i; - unsigned int val, streams; + unsigned int i, val, wcaps; val = 0; - if (nid != codec->afg && - (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) { + wcaps = get_wcaps(codec, nid); + if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) { val = snd_hda_param_read(codec, nid, AC_PAR_PCM); if (val == -1) return -EIO; @@ -2324,15 +2572,20 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, if (val & (1 << i)) rates |= rate_bits[i].alsa_bits; } + if (rates == 0) { + snd_printk(KERN_ERR "hda_codec: rates == 0 " + "(nid=0x%x, val=0x%x, ovrd=%i)\n", + nid, val, + (wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0); + return -EIO; + } *ratesp = rates; } if (formatsp || bpsp) { u64 formats = 0; - unsigned int bps; - unsigned int wcaps; + unsigned int streams, bps; - wcaps = get_wcaps(codec, nid); streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); if (streams == -1) return -EIO; @@ -2385,6 +2638,15 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, formats |= SNDRV_PCM_FMTBIT_U8; bps = 8; } + if (formats == 0) { + snd_printk(KERN_ERR "hda_codec: formats == 0 " + "(nid=0x%x, val=0x%x, ovrd=%i, " + "streams=0x%x)\n", + nid, val, + (wcaps & AC_WCAP_FORMAT_OVRD) ? 1 : 0, + streams); + return -EIO; + } if (formatsp) *formatsp = formats; if (bpsp) @@ -2500,12 +2762,16 @@ static int hda_pcm_default_cleanup(struct hda_pcm_stream *hinfo, static int set_pcm_default_values(struct hda_codec *codec, struct hda_pcm_stream *info) { + int err; + /* query support PCM information from the given NID */ if (info->nid && (!info->rates || !info->formats)) { - snd_hda_query_supported_pcm(codec, info->nid, + err = snd_hda_query_supported_pcm(codec, info->nid, info->rates ? NULL : &info->rates, info->formats ? NULL : &info->formats, info->maxbps ? NULL : &info->maxbps); + if (err < 0) + return err; } if (info->ops.open == NULL) info->ops.open = hda_pcm_default_open_close; @@ -2548,13 +2814,10 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) for (i = 0; i < ARRAY_SIZE(audio_idx); i++) { dev = audio_idx[i]; if (!test_bit(dev, bus->pcm_dev_bits)) - break; + goto ok; } - if (i >= ARRAY_SIZE(audio_idx)) { - snd_printk(KERN_WARNING "Too many audio devices\n"); - return -EAGAIN; - } - break; + snd_printk(KERN_WARNING "Too many audio devices\n"); + return -EAGAIN; case HDA_PCM_TYPE_SPDIF: case HDA_PCM_TYPE_HDMI: case HDA_PCM_TYPE_MODEM: @@ -2569,6 +2832,7 @@ static int get_empty_pcm_device(struct hda_bus *bus, int type) snd_printk(KERN_WARNING "Invalid PCM type %d\n", type); return -EINVAL; } + ok: set_bit(dev, bus->pcm_dev_bits); return dev; } @@ -2605,24 +2869,36 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) if (!codec->patch_ops.build_pcms) return 0; err = codec->patch_ops.build_pcms(codec); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot build PCMs" + "for #%d (error %d)\n", codec->addr, err); + err = snd_hda_codec_reset(codec); + if (err < 0) { + printk(KERN_ERR + "hda_codec: cannot revert codec\n"); + return err; + } + } } for (pcm = 0; pcm < codec->num_pcms; pcm++) { struct hda_pcm *cpcm = &codec->pcm_info[pcm]; int dev; if (!cpcm->stream[0].substreams && !cpcm->stream[1].substreams) - return 0; /* no substreams assigned */ + continue; /* no substreams assigned */ if (!cpcm->pcm) { dev = get_empty_pcm_device(codec->bus, cpcm->pcm_type); if (dev < 0) - return 0; + continue; /* no fatal error */ cpcm->device = dev; err = snd_hda_attach_pcm(codec, cpcm); - if (err < 0) - return err; + if (err < 0) { + printk(KERN_ERR "hda_codec: cannot attach " + "PCM stream %d for codec #%d\n", + dev, codec->addr); + continue; /* no fatal error */ + } } } return 0; @@ -2724,6 +3000,67 @@ int snd_hda_check_board_config(struct hda_codec *codec, EXPORT_SYMBOL_HDA(snd_hda_check_board_config); /** + * snd_hda_check_board_codec_sid_config - compare the current codec + subsystem ID with the + config table + + This is important for Gateway notebooks with SB450 HDA Audio + where the vendor ID of the PCI device is: + ATI Technologies Inc SB450 HDA Audio [1002:437b] + and the vendor/subvendor are found only at the codec. + + * @codec: the HDA codec + * @num_configs: number of config enums + * @models: array of model name strings + * @tbl: configuration table, terminated by null entries + * + * Compares the modelname or PCI subsystem id of the current codec with the + * given configuration table. If a matching entry is found, returns its + * config value (supposed to be 0 or positive). + * + * If no entries are matching, the function returns a negative value. + */ +int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, + int num_configs, const char **models, + const struct snd_pci_quirk *tbl) +{ + const struct snd_pci_quirk *q; + + /* Search for codec ID */ + for (q = tbl; q->subvendor; q++) { + unsigned long vendorid = (q->subdevice) | (q->subvendor << 16); + + if (vendorid == codec->subsystem_id) + break; + } + + if (!q->subvendor) + return -1; + + tbl = q; + + if (tbl->value >= 0 && tbl->value < num_configs) { +#ifdef CONFIG_SND_DEBUG_DETECT + char tmp[10]; + const char *model = NULL; + if (models) + model = models[tbl->value]; + if (!model) { + sprintf(tmp, "#%d", tbl->value); + model = tmp; + } + snd_printdd(KERN_INFO "hda_codec: model '%s' is selected " + "for config %x:%x (%s)\n", + model, tbl->subvendor, tbl->subdevice, + (tbl->name ? tbl->name : "Unknown device")); +#endif + return tbl->value; + } + return -1; +} +EXPORT_SYMBOL_HDA(snd_hda_check_board_codec_sid_config); + +/** * snd_hda_add_new_ctls - create controls from the array * @codec: the HDA codec * @knew: the array of struct snd_kcontrol_new @@ -2815,7 +3152,7 @@ void snd_hda_power_down(struct hda_codec *codec) return; if (power_save(codec)) { codec->power_transition = 1; /* avoid reentrance */ - schedule_delayed_work(&codec->power_work, + queue_delayed_work(codec->bus->workq, &codec->power_work, msecs_to_jiffies(power_save(codec) * 1000)); } } @@ -3026,6 +3363,16 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, } EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_prepare); +int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, + struct hda_multi_out *mout) +{ + mutex_lock(&codec->spdif_mutex); + cleanup_dig_out_stream(codec, mout->dig_out_nid); + mutex_unlock(&codec->spdif_mutex); + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_multi_out_dig_cleanup); + /* * release the digital out */ @@ -3252,8 +3599,7 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, if (ignore_nids && is_in_nid_list(nid, ignore_nids)) continue; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; loc = get_defcfg_location(def_conf); @@ -3329,10 +3675,22 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[AUTO_PIN_AUX] = nid; break; case AC_JACK_SPDIF_OUT: - cfg->dig_out_pin = nid; + case AC_JACK_DIG_OTHER_OUT: + if (cfg->dig_outs >= ARRAY_SIZE(cfg->dig_out_pins)) + continue; + cfg->dig_out_pins[cfg->dig_outs] = nid; + cfg->dig_out_type[cfg->dig_outs] = + (loc == AC_JACK_LOC_HDMI) ? + HDA_PCM_TYPE_HDMI : HDA_PCM_TYPE_SPDIF; + cfg->dig_outs++; break; case AC_JACK_SPDIF_IN: + case AC_JACK_DIG_OTHER_IN: cfg->dig_in_pin = nid; + if (loc == AC_JACK_LOC_HDMI) + cfg->dig_in_type = HDA_PCM_TYPE_HDMI; + else + cfg->dig_in_type = HDA_PCM_TYPE_SPDIF; break; } } @@ -3438,6 +3796,9 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->hp_pins[1], cfg->hp_pins[2], cfg->hp_pins[3], cfg->hp_pins[4]); snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); + if (cfg->dig_outs) + snd_printd(" dig-out=0x%x/0x%x\n", + cfg->dig_out_pins[0], cfg->dig_out_pins[1]); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], @@ -3446,6 +3807,8 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, cfg->input_pins[AUTO_PIN_FRONT_LINE], cfg->input_pins[AUTO_PIN_CD], cfg->input_pins[AUTO_PIN_AUX]); + if (cfg->dig_in_pin) + snd_printd(" dig-in=0x%x\n", cfg->dig_in_pin); return 0; } diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 5810ef5..2fdecf4 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -614,6 +614,7 @@ struct hda_bus { /* unsolicited event queue */ struct hda_bus_unsolicited *unsol; + char workq_name[16]; struct workqueue_struct *workq; /* common workqueue for codecs */ /* assigned PCMs */ @@ -738,6 +739,7 @@ struct hda_codec { hda_nid_t mfg; /* MFG node id */ /* ids */ + u32 function_id; u32 vendor_id; u32 subsystem_id; u32 revision_id; @@ -777,11 +779,14 @@ struct hda_codec { unsigned short spdif_ctls; /* SPDIF control bits */ unsigned int spdif_in_enable; /* SPDIF input enable? */ hda_nid_t *slave_dig_outs; /* optional digital out slave widgets */ + struct snd_array init_pins; /* initial (BIOS) pin configurations */ + struct snd_array driver_pins; /* pin configs set by codec parser */ #ifdef CONFIG_SND_HDA_HWDEP struct snd_hwdep *hwdep; /* assigned hwdep device */ struct snd_array init_verbs; /* additional init verbs */ struct snd_array hints; /* additional hints */ + struct snd_array user_pins; /* default pin configs to override */ #endif /* misc flags */ @@ -789,6 +794,9 @@ struct hda_codec { * status change * (e.g. Realtek codecs) */ + unsigned int pin_amp_workaround:1; /* pin out-amp takes index + * (e.g. Conexant codecs) + */ #ifdef CONFIG_SND_HDA_POWER_SAVE unsigned int power_on :1; /* current (global) power-state */ unsigned int power_transition :1; /* power-state in transition */ @@ -854,6 +862,18 @@ void snd_hda_codec_resume_cache(struct hda_codec *codec); #define snd_hda_sequence_write_cache snd_hda_sequence_write #endif +/* the struct for codec->pin_configs */ +struct hda_pincfg { + hda_nid_t nid; + unsigned int cfg; +}; + +unsigned int snd_hda_codec_get_pincfg(struct hda_codec *codec, hda_nid_t nid); +int snd_hda_codec_set_pincfg(struct hda_codec *codec, hda_nid_t nid, + unsigned int cfg); +int snd_hda_add_pincfg(struct hda_codec *codec, struct snd_array *list, + hda_nid_t nid, unsigned int cfg); /* for hwdep */ + /* * Mixer */ diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 65745e9..1d5797a 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -144,9 +144,9 @@ static int add_new_node(struct hda_codec *codec, struct hda_gspec *spec, hda_nid node->type = (node->wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; if (node->type == AC_WID_PIN) { - node->pin_caps = snd_hda_param_read(codec, node->nid, AC_PAR_PIN_CAP); + node->pin_caps = snd_hda_query_pin_caps(codec, node->nid); node->pin_ctl = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - node->def_cfg = snd_hda_codec_read(codec, node->nid, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + node->def_cfg = snd_hda_codec_get_pincfg(codec, node->nid); } if (node->wid_caps & AC_WCAP_OUT_AMP) { diff --git a/sound/pci/hda/hda_hwdep.c b/sound/pci/hda/hda_hwdep.c index 300ab40..1c57505 100644 --- a/sound/pci/hda/hda_hwdep.c +++ b/sound/pci/hda/hda_hwdep.c @@ -30,6 +30,12 @@ #include <sound/hda_hwdep.h> #include <sound/minors.h> +/* hint string pair */ +struct hda_hint { + const char *key; + const char *val; /* contained in the same alloc as key */ +}; + /* * write/read an out-of-bound verb */ @@ -99,16 +105,17 @@ static int hda_hwdep_open(struct snd_hwdep *hw, struct file *file) static void clear_hwdep_elements(struct hda_codec *codec) { - char **head; int i; /* clear init verbs */ snd_array_free(&codec->init_verbs); /* clear hints */ - head = codec->hints.list; - for (i = 0; i < codec->hints.used; i++, head++) - kfree(*head); + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + kfree(hint->key); /* we don't need to free hint->val */ + } snd_array_free(&codec->hints); + snd_array_free(&codec->user_pins); } static void hwdep_free(struct snd_hwdep *hwdep) @@ -140,7 +147,8 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) #endif snd_array_init(&codec->init_verbs, sizeof(struct hda_verb), 32); - snd_array_init(&codec->hints, sizeof(char *), 32); + snd_array_init(&codec->hints, sizeof(struct hda_hint), 32); + snd_array_init(&codec->user_pins, sizeof(struct hda_pincfg), 16); return 0; } @@ -153,7 +161,13 @@ int /*__devinit*/ snd_hda_create_hwdep(struct hda_codec *codec) static int clear_codec(struct hda_codec *codec) { - snd_hda_codec_reset(codec); + int err; + + err = snd_hda_codec_reset(codec); + if (err < 0) { + snd_printk(KERN_ERR "The codec is being used, can't free.\n"); + return err; + } clear_hwdep_elements(codec); return 0; } @@ -162,20 +176,29 @@ static int reconfig_codec(struct hda_codec *codec) { int err; + snd_hda_power_up(codec); snd_printk(KERN_INFO "hda-codec: reconfiguring\n"); - snd_hda_codec_reset(codec); + err = snd_hda_codec_reset(codec); + if (err < 0) { + snd_printk(KERN_ERR + "The codec is being used, can't reconfigure.\n"); + goto error; + } err = snd_hda_codec_configure(codec); if (err < 0) - return err; + goto error; /* rebuild PCMs */ err = snd_hda_codec_build_pcms(codec); if (err < 0) - return err; + goto error; /* rebuild mixers */ err = snd_hda_codec_build_controls(codec); if (err < 0) - return err; - return 0; + goto error; + err = snd_card_register(codec->bus->card); + error: + snd_hda_power_down(codec); + return err; } /* @@ -271,47 +294,195 @@ static ssize_t type##_store(struct device *dev, \ CODEC_ACTION_STORE(reconfig); CODEC_ACTION_STORE(clear); +static ssize_t init_verbs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int i, len = 0; + for (i = 0; i < codec->init_verbs.used; i++) { + struct hda_verb *v = snd_array_elem(&codec->init_verbs, i); + len += snprintf(buf + len, PAGE_SIZE - len, + "0x%02x 0x%03x 0x%04x\n", + v->nid, v->verb, v->param); + } + return len; +} + static ssize_t init_verbs_store(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; - char *p; - struct hda_verb verb, *v; + struct hda_verb *v; + int nid, verb, param; - verb.nid = simple_strtoul(buf, &p, 0); - verb.verb = simple_strtoul(p, &p, 0); - verb.param = simple_strtoul(p, &p, 0); - if (!verb.nid || !verb.verb || !verb.param) + if (sscanf(buf, "%i %i %i", &nid, &verb, ¶m) != 3) + return -EINVAL; + if (!nid || !verb) return -EINVAL; v = snd_array_new(&codec->init_verbs); if (!v) return -ENOMEM; - *v = verb; + v->nid = nid; + v->verb = verb; + v->param = param; return count; } +static ssize_t hints_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int i, len = 0; + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + len += snprintf(buf + len, PAGE_SIZE - len, + "%s = %s\n", hint->key, hint->val); + } + return len; +} + +static struct hda_hint *get_hint(struct hda_codec *codec, const char *key) +{ + int i; + + for (i = 0; i < codec->hints.used; i++) { + struct hda_hint *hint = snd_array_elem(&codec->hints, i); + if (!strcmp(hint->key, key)) + return hint; + } + return NULL; +} + +static void remove_trail_spaces(char *str) +{ + char *p; + if (!*str) + return; + p = str + strlen(str) - 1; + for (; isspace(*p); p--) { + *p = 0; + if (p == str) + return; + } +} + +#define MAX_HINTS 1024 + static ssize_t hints_store(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { struct snd_hwdep *hwdep = dev_get_drvdata(dev); struct hda_codec *codec = hwdep->private_data; - char *p; - char **hint; + char *key, *val; + struct hda_hint *hint; - if (!*buf || isspace(*buf) || *buf == '#' || *buf == '\n') + while (isspace(*buf)) + buf++; + if (!*buf || *buf == '#' || *buf == '\n') return count; - p = kstrndup_noeol(buf, 1024); - if (!p) + if (*buf == '=') + return -EINVAL; + key = kstrndup_noeol(buf, 1024); + if (!key) return -ENOMEM; - hint = snd_array_new(&codec->hints); + /* extract key and val */ + val = strchr(key, '='); + if (!val) { + kfree(key); + return -EINVAL; + } + *val++ = 0; + while (isspace(*val)) + val++; + remove_trail_spaces(key); + remove_trail_spaces(val); + hint = get_hint(codec, key); + if (hint) { + /* replace */ + kfree(hint->key); + hint->key = key; + hint->val = val; + return count; + } + /* allocate a new hint entry */ + if (codec->hints.used >= MAX_HINTS) + hint = NULL; + else + hint = snd_array_new(&codec->hints); if (!hint) { - kfree(p); + kfree(key); return -ENOMEM; } - *hint = p; + hint->key = key; + hint->val = val; + return count; +} + +static ssize_t pin_configs_show(struct hda_codec *codec, + struct snd_array *list, + char *buf) +{ + int i, len = 0; + for (i = 0; i < list->used; i++) { + struct hda_pincfg *pin = snd_array_elem(list, i); + len += sprintf(buf + len, "0x%02x 0x%08x\n", + pin->nid, pin->cfg); + } + return len; +} + +static ssize_t init_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->init_pins, buf); +} + +static ssize_t user_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->user_pins, buf); +} + +static ssize_t driver_pin_configs_show(struct device *dev, + struct device_attribute *attr, + char *buf) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + return pin_configs_show(codec, &codec->driver_pins, buf); +} + +#define MAX_PIN_CONFIGS 32 + +static ssize_t user_pin_configs_store(struct device *dev, + struct device_attribute *attr, + const char *buf, size_t count) +{ + struct snd_hwdep *hwdep = dev_get_drvdata(dev); + struct hda_codec *codec = hwdep->private_data; + int nid, cfg; + int err; + + if (sscanf(buf, "%i %i", &nid, &cfg) != 2) + return -EINVAL; + if (!nid) + return -EINVAL; + err = snd_hda_add_pincfg(codec, &codec->user_pins, nid, cfg); + if (err < 0) + return err; return count; } @@ -330,8 +501,11 @@ static struct device_attribute codec_attrs[] = { CODEC_ATTR_RO(mfg), CODEC_ATTR_RW(name), CODEC_ATTR_RW(modelname), - CODEC_ATTR_WO(init_verbs), - CODEC_ATTR_WO(hints), + CODEC_ATTR_RW(init_verbs), + CODEC_ATTR_RW(hints), + CODEC_ATTR_RO(init_pin_configs), + CODEC_ATTR_RW(user_pin_configs), + CODEC_ATTR_RO(driver_pin_configs), CODEC_ATTR_WO(reconfig), CODEC_ATTR_WO(clear), }; @@ -350,4 +524,29 @@ int snd_hda_hwdep_add_sysfs(struct hda_codec *codec) return 0; } +/* + * Look for hint string + */ +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key) +{ + struct hda_hint *hint = get_hint(codec, key); + return hint ? hint->val : NULL; +} +EXPORT_SYMBOL_HDA(snd_hda_get_hint); + +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) +{ + const char *p = snd_hda_get_hint(codec, key); + if (!p || !*p) + return -ENOENT; + switch (toupper(*p)) { + case 'T': /* true */ + case 'Y': /* yes */ + case '1': + return 1; + } + return 0; +} +EXPORT_SYMBOL_HDA(snd_hda_get_bool_hint); + #endif /* CONFIG_SND_HDA_RECONFIG */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index f960344..30829ee 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -381,6 +381,7 @@ struct azx { /* HD codec */ unsigned short codec_mask; + int codec_probe_mask; /* copied from probe_mask option */ struct hda_bus *bus; /* CORB/RIRB */ @@ -858,13 +859,18 @@ static void azx_stream_start(struct azx *chip, struct azx_dev *azx_dev) SD_CTL_DMA_START | SD_INT_MASK); } -/* stop a stream */ -static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) +/* stop DMA */ +static void azx_stream_clear(struct azx *chip, struct azx_dev *azx_dev) { - /* stop DMA */ azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & ~(SD_CTL_DMA_START | SD_INT_MASK)); azx_sd_writeb(azx_dev, SD_STS, SD_INT_MASK); /* to be sure */ +} + +/* stop a stream */ +static void azx_stream_stop(struct azx *chip, struct azx_dev *azx_dev) +{ + azx_stream_clear(chip, azx_dev); /* disable SIE */ azx_writeb(chip, INTCTL, azx_readb(chip, INTCTL) & ~(1 << azx_dev->index)); @@ -1075,8 +1081,7 @@ static int azx_setup_periods(struct azx *chip, azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); - period_bytes = snd_pcm_lib_period_bytes(substream); - azx_dev->period_bytes = period_bytes; + period_bytes = azx_dev->period_bytes; periods = azx_dev->bufsize / period_bytes; /* program the initial BDL entries */ @@ -1123,24 +1128,17 @@ static int azx_setup_periods(struct azx *chip, error: snd_printk(KERN_ERR "Too many BDL entries: buffer=%d, period=%d\n", azx_dev->bufsize, period_bytes); - /* reset */ - azx_sd_writel(azx_dev, SD_BDLPL, 0); - azx_sd_writel(azx_dev, SD_BDLPU, 0); return -EINVAL; } -/* - * set up the SD for streaming - */ -static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) +/* reset stream */ +static void azx_stream_reset(struct azx *chip, struct azx_dev *azx_dev) { unsigned char val; int timeout; - /* make sure the run bit is zero for SD */ - azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) & - ~SD_CTL_DMA_START); - /* reset stream */ + azx_stream_clear(chip, azx_dev); + azx_sd_writeb(azx_dev, SD_CTL, azx_sd_readb(azx_dev, SD_CTL) | SD_CTL_STREAM_RESET); udelay(3); @@ -1157,7 +1155,15 @@ static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) while (((val = azx_sd_readb(azx_dev, SD_CTL)) & SD_CTL_STREAM_RESET) && --timeout) ; +} +/* + * set up the SD for streaming + */ +static int azx_setup_controller(struct azx *chip, struct azx_dev *azx_dev) +{ + /* make sure the run bit is zero for SD */ + azx_stream_clear(chip, azx_dev); /* program the stream_tag */ azx_sd_writel(azx_dev, SD_CTL, (azx_sd_readl(azx_dev, SD_CTL) & ~SD_CTL_STREAM_TAG_MASK)| @@ -1228,7 +1234,6 @@ static unsigned int azx_max_codecs[AZX_NUM_DRIVERS] __devinitdata = { }; static int __devinit azx_codec_create(struct azx *chip, const char *model, - unsigned int codec_probe_mask, int no_init) { struct hda_bus_template bus_temp; @@ -1261,7 +1266,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, /* First try to probe all given codec slots */ for (c = 0; c < max_slots; c++) { - if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { + if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { if (probe_codec(chip, c) < 0) { /* Some BIOSen give you wrong codec addresses * that don't exist @@ -1285,7 +1290,7 @@ static int __devinit azx_codec_create(struct azx *chip, const char *model, /* Then create codec instances */ for (c = 0; c < max_slots; c++) { - if ((chip->codec_mask & (1 << c)) & codec_probe_mask) { + if ((chip->codec_mask & (1 << c)) & chip->codec_probe_mask) { struct hda_codec *codec; err = snd_hda_codec_new(chip->bus, c, !no_init, &codec); if (err < 0) @@ -1403,6 +1408,8 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) runtime->private_data = azx_dev; snd_pcm_set_sync(substream); mutex_unlock(&chip->open_mutex); + + azx_stream_reset(chip, azx_dev); return 0; } @@ -1429,6 +1436,11 @@ static int azx_pcm_close(struct snd_pcm_substream *substream) static int azx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { + struct azx_dev *azx_dev = get_azx_dev(substream); + + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); } @@ -1443,6 +1455,9 @@ static int azx_pcm_hw_free(struct snd_pcm_substream *substream) azx_sd_writel(azx_dev, SD_BDLPL, 0); azx_sd_writel(azx_dev, SD_BDLPU, 0); azx_sd_writel(azx_dev, SD_CTL, 0); + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; hinfo->ops.cleanup(hinfo, apcm->codec, substream); @@ -1456,23 +1471,37 @@ static int azx_pcm_prepare(struct snd_pcm_substream *substream) struct azx_dev *azx_dev = get_azx_dev(substream); struct hda_pcm_stream *hinfo = apcm->hinfo[substream->stream]; struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int bufsize, period_bytes, format_val; + int err; - azx_dev->bufsize = snd_pcm_lib_buffer_bytes(substream); - azx_dev->format_val = snd_hda_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hinfo->maxbps); - if (!azx_dev->format_val) { + format_val = snd_hda_calc_stream_format(runtime->rate, + runtime->channels, + runtime->format, + hinfo->maxbps); + if (!format_val) { snd_printk(KERN_ERR SFX "invalid format_val, rate=%d, ch=%d, format=%d\n", runtime->rate, runtime->channels, runtime->format); return -EINVAL; } + bufsize = snd_pcm_lib_buffer_bytes(substream); + period_bytes = snd_pcm_lib_period_bytes(substream); + snd_printdd("azx_pcm_prepare: bufsize=0x%x, format=0x%x\n", - azx_dev->bufsize, azx_dev->format_val); - if (azx_setup_periods(chip, substream, azx_dev) < 0) - return -EINVAL; + bufsize, format_val); + + if (bufsize != azx_dev->bufsize || + period_bytes != azx_dev->period_bytes || + format_val != azx_dev->format_val) { + azx_dev->bufsize = bufsize; + azx_dev->period_bytes = period_bytes; + azx_dev->format_val = format_val; + err = azx_setup_periods(chip, substream, azx_dev); + if (err < 0) + return err; + } + azx_setup_controller(chip, azx_dev); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) azx_dev->fifo_size = azx_sd_readw(azx_dev, SD_FIFOSIZE) + 1; @@ -1947,16 +1976,13 @@ static int azx_suspend(struct pci_dev *pci, pm_message_t state) return 0; } -static int azx_resume_early(struct pci_dev *pci) -{ - return pci_restore_state(pci); -} - static int azx_resume(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); struct azx *chip = card->private_data; + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); if (pci_enable_device(pci) < 0) { printk(KERN_ERR "hda-intel: pci_enable_device failed, " "disabling device\n"); @@ -2062,26 +2088,31 @@ static int __devinit check_position_fix(struct azx *chip, int fix) { const struct snd_pci_quirk *q; - /* Check VIA HD Audio Controller exist */ - if (chip->pci->vendor == PCI_VENDOR_ID_VIA && - chip->pci->device == VIA_HDAC_DEVICE_ID) { + switch (fix) { + case POS_FIX_LPIB: + case POS_FIX_POSBUF: + return fix; + } + + /* Check VIA/ATI HD Audio Controller exist */ + switch (chip->driver_type) { + case AZX_DRIVER_VIA: + case AZX_DRIVER_ATI: chip->via_dmapos_patch = 1; /* Use link position directly, avoid any transfer problem. */ return POS_FIX_LPIB; } chip->via_dmapos_patch = 0; - if (fix == POS_FIX_AUTO) { - q = snd_pci_quirk_lookup(chip->pci, position_fix_list); - if (q) { - printk(KERN_INFO - "hda_intel: position_fix set to %d " - "for device %04x:%04x\n", - q->value, q->subvendor, q->subdevice); - return q->value; - } + q = snd_pci_quirk_lookup(chip->pci, position_fix_list); + if (q) { + printk(KERN_INFO + "hda_intel: position_fix set to %d " + "for device %04x:%04x\n", + q->value, q->subvendor, q->subdevice); + return q->value; } - return fix; + return POS_FIX_AUTO; } /* @@ -2098,23 +2129,36 @@ static struct snd_pci_quirk probe_mask_list[] __devinitdata = { SND_PCI_QUIRK(0x1028, 0x20ac, "Dell Studio Desktop", 0x01), /* including bogus ALC268 in slot#2 that conflicts with ALC888 */ SND_PCI_QUIRK(0x17c0, 0x4085, "Medion MD96630", 0x01), + /* forced codec slots */ + SND_PCI_QUIRK(0x1046, 0x1262, "ASUS W5F", 0x103), {} }; +#define AZX_FORCE_CODEC_MASK 0x100 + static void __devinit check_probe_mask(struct azx *chip, int dev) { const struct snd_pci_quirk *q; - if (probe_mask[dev] == -1) { + chip->codec_probe_mask = probe_mask[dev]; + if (chip->codec_probe_mask == -1) { q = snd_pci_quirk_lookup(chip->pci, probe_mask_list); if (q) { printk(KERN_INFO "hda_intel: probe_mask set to 0x%x " "for device %04x:%04x\n", q->value, q->subvendor, q->subdevice); - probe_mask[dev] = q->value; + chip->codec_probe_mask = q->value; } } + + /* check forced option */ + if (chip->codec_probe_mask != -1 && + (chip->codec_probe_mask & AZX_FORCE_CODEC_MASK)) { + chip->codec_mask = chip->codec_probe_mask & 0xff; + printk(KERN_INFO "hda_intel: codec_mask forced to 0x%x\n", + chip->codec_mask); + } } @@ -2211,9 +2255,17 @@ static int __devinit azx_create(struct snd_card *card, struct pci_dev *pci, gcap = azx_readw(chip, GCAP); snd_printdd("chipset global capabilities = 0x%x\n", gcap); + /* ATI chips seems buggy about 64bit DMA addresses */ + if (chip->driver_type == AZX_DRIVER_ATI) + gcap &= ~0x01; + /* allow 64bit DMA address if supported by H/W */ if ((gcap & 0x01) && !pci_set_dma_mask(pci, DMA_64BIT_MASK)) pci_set_consistent_dma_mask(pci, DMA_64BIT_MASK); + else { + pci_set_dma_mask(pci, DMA_32BIT_MASK); + pci_set_consistent_dma_mask(pci, DMA_32BIT_MASK); + } /* read number of streams from GCAP register instead of using * hardcoded value @@ -2347,8 +2399,7 @@ static int __devinit azx_probe(struct pci_dev *pci, card->private_data = chip; /* create codec instances */ - err = azx_codec_create(chip, model[dev], probe_mask[dev], - probe_only[dev]); + err = azx_codec_create(chip, model[dev], probe_only[dev]); if (err < 0) goto out_free; @@ -2445,10 +2496,10 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x10de, 0x0ac1), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac2), .driver_data = AZX_DRIVER_NVIDIA }, { PCI_DEVICE(0x10de, 0x0ac3), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd4), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd5), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd6), .driver_data = AZX_DRIVER_NVIDIA }, - { PCI_DEVICE(0x10de, 0x0bd7), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d94), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d95), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d96), .driver_data = AZX_DRIVER_NVIDIA }, + { PCI_DEVICE(0x10de, 0x0d97), .driver_data = AZX_DRIVER_NVIDIA }, /* Teradici */ { PCI_DEVICE(0x6549, 0x1200), .driver_data = AZX_DRIVER_TERA }, /* AMD Generic, PCI class code and Vendor ID for HD Audio */ @@ -2468,7 +2519,6 @@ static struct pci_driver driver = { .remove = __devexit_p(azx_remove), #ifdef CONFIG_PM .suspend = azx_suspend, - .resume_early = azx_resume_early, .resume = azx_resume, #endif }; diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h index 6f2fe0f..8334901 100644 --- a/sound/pci/hda/hda_local.h +++ b/sound/pci/hda/hda_local.h @@ -26,8 +26,10 @@ /* * for mixer controls */ +#define HDA_COMPOSE_AMP_VAL_OFS(nid,chs,idx,dir,ofs) \ + ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19) | ((ofs)<<23)) #define HDA_COMPOSE_AMP_VAL(nid,chs,idx,dir) \ - ((nid) | ((chs)<<16) | ((dir)<<18) | ((idx)<<19)) + HDA_COMPOSE_AMP_VAL_OFS(nid, chs, idx, dir, 0) /* mono volume with index (index=0,1,...) (channel=1,2) */ #define HDA_CODEC_VOLUME_MONO_IDX(xname, xcidx, nid, channel, xindex, direction) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xcidx, \ @@ -96,7 +98,7 @@ struct snd_kcontrol *snd_hda_find_mixer_ctl(struct hda_codec *codec, const char *name); int snd_hda_add_vmaster(struct hda_codec *codec, char *name, unsigned int *tlv, const char **slaves); -void snd_hda_codec_reset(struct hda_codec *codec); +int snd_hda_codec_reset(struct hda_codec *codec); int snd_hda_codec_configure(struct hda_codec *codec); /* amp value bits */ @@ -134,7 +136,7 @@ extern struct hda_ctl_ops snd_hda_bind_sw; /* for bind-switch */ struct hda_bind_ctls { struct hda_ctl_ops *ops; - long values[]; + unsigned long values[]; }; int snd_hda_mixer_bind_ctls_info(struct snd_kcontrol *kcontrol, @@ -227,6 +229,7 @@ struct hda_multi_out { hda_nid_t hp_nid; /* optional DAC for HP, 0 when not exists */ hda_nid_t extra_out_nid[3]; /* optional DACs, 0 when not exists */ hda_nid_t dig_out_nid; /* digital out audio widget */ + hda_nid_t *slave_dig_outs; int max_channels; /* currently supported analog channels */ int dig_out_used; /* current usage of digital out (HDA_DIG_XXX) */ int no_share_stream; /* don't share a stream with multiple pins */ @@ -251,6 +254,8 @@ int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, unsigned int stream_tag, unsigned int format, struct snd_pcm_substream *substream); +int snd_hda_multi_out_dig_cleanup(struct hda_codec *codec, + struct hda_multi_out *mout); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, struct snd_pcm_substream *substream, @@ -296,6 +301,9 @@ void snd_print_pcm_bits(int pcm, char *buf, int buflen); int snd_hda_check_board_config(struct hda_codec *codec, int num_configs, const char **modelnames, const struct snd_pci_quirk *pci_list); +int snd_hda_check_board_codec_sid_config(struct hda_codec *codec, + int num_configs, const char **models, + const struct snd_pci_quirk *tbl); int snd_hda_add_new_ctls(struct hda_codec *codec, struct snd_kcontrol_new *knew); @@ -349,9 +357,12 @@ struct auto_pin_cfg { int line_out_type; /* AUTO_PIN_XXX_OUT */ hda_nid_t hp_pins[AUTO_CFG_MAX_OUTS]; hda_nid_t input_pins[AUTO_PIN_LAST]; - hda_nid_t dig_out_pin; + int dig_outs; + hda_nid_t dig_out_pins[2]; hda_nid_t dig_in_pin; hda_nid_t mono_out_pin; + int dig_out_type[2]; /* HDA_PCM_TYPE_XXX */ + int dig_in_type; /* HDA_PCM_TYPE_XXX */ }; #define get_defcfg_connect(cfg) \ @@ -400,6 +411,7 @@ static inline u32 get_wcaps(struct hda_codec *codec, hda_nid_t nid) u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps); +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid); int snd_hda_ctl_add(struct hda_codec *codec, struct snd_kcontrol *kctl); void snd_hda_ctls_clear(struct hda_codec *codec); @@ -422,6 +434,23 @@ static inline int snd_hda_hwdep_add_sysfs(struct hda_codec *codec) } #endif +#ifdef CONFIG_SND_HDA_RECONFIG +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key); +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key); +#else +static inline +const char *snd_hda_get_hint(struct hda_codec *codec, const char *key) +{ + return NULL; +} + +static inline +int snd_hda_get_bool_hint(struct hda_codec *codec, const char *key) +{ + return -ENOENT; +} +#endif + /* * power-management */ @@ -453,6 +482,7 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, #define get_amp_channels(kc) (((kc)->private_value >> 16) & 0x3) #define get_amp_direction(kc) (((kc)->private_value >> 18) & 0x1) #define get_amp_index(kc) (((kc)->private_value >> 19) & 0xf) +#define get_amp_offset(kc) (((kc)->private_value >> 23) & 0x3f) /* * CEA Short Audio Descriptor data diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7ca66d6..93d7499 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -399,7 +399,10 @@ static void print_conn_list(struct snd_info_buffer *buffer, { int c, curr = -1; - if (conn_len > 1 && wid_type != AC_WID_AUD_MIX) + if (conn_len > 1 && + wid_type != AC_WID_AUD_MIX && + wid_type != AC_WID_VOL_KNB && + wid_type != AC_WID_POWER) curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); snd_iprintf(buffer, " Connection: %d\n", conn_len); @@ -466,8 +469,9 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, "Codec: %s\n", codec->name ? codec->name : "Not Set"); snd_iprintf(buffer, "Address: %d\n", codec->addr); - snd_iprintf(buffer, "Vendor Id: 0x%x\n", codec->vendor_id); - snd_iprintf(buffer, "Subsystem Id: 0x%x\n", codec->subsystem_id); + snd_iprintf(buffer, "Function Id: 0x%x\n", codec->function_id); + snd_iprintf(buffer, "Vendor Id: 0x%08x\n", codec->vendor_id); + snd_iprintf(buffer, "Subsystem Id: 0x%08x\n", codec->subsystem_id); snd_iprintf(buffer, "Revision Id: 0x%x\n", codec->revision_id); if (codec->mfg) @@ -553,8 +557,14 @@ static void print_codec_info(struct snd_info_entry *entry, snd_iprintf(buffer, " Amp-Out caps: "); print_amp_caps(buffer, codec, nid, HDA_OUTPUT); snd_iprintf(buffer, " Amp-Out vals: "); - print_amp_vals(buffer, codec, nid, HDA_OUTPUT, - wid_caps & AC_WCAP_STEREO, 1); + if (wid_type == AC_WID_PIN && + codec->pin_amp_workaround) + print_amp_vals(buffer, codec, nid, HDA_OUTPUT, + wid_caps & AC_WCAP_STEREO, + conn_len); + else + print_amp_vals(buffer, codec, nid, HDA_OUTPUT, + wid_caps & AC_WCAP_STEREO, 1); } switch (wid_type) { diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 2e7371e..38ad3f7 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -27,11 +27,12 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" struct ad198x_spec { struct snd_kcontrol_new *mixers[5]; int num_mixers; - + unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL termination! */ @@ -154,6 +155,16 @@ static const char *ad_slave_sws[] = { static void ad198x_free_kctls(struct hda_codec *codec); +/* additional beep mixers; the actual parameters are overwritten at build */ +static struct snd_kcontrol_new ad_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_OUTPUT), + { } /* end */ +}; + +#define set_beep_amp(spec, nid, idx, dir) \ + ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 1, idx, dir)) /* mono */ + static int ad198x_build_controls(struct hda_codec *codec) { struct ad198x_spec *spec = codec->spec; @@ -181,6 +192,21 @@ static int ad198x_build_controls(struct hda_codec *codec) return err; } + /* create beep controls if needed */ + if (spec->beep_amp) { + struct snd_kcontrol_new *knew; + for (knew = ad_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, kctl); + if (err < 0) + return err; + } + } + /* if we have no master control, let's create it */ if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { unsigned int vmaster_tlv[4]; @@ -275,6 +301,14 @@ static int ad198x_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, format, substream); } +static int ad198x_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct ad198x_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + /* * Analog capture */ @@ -333,7 +367,8 @@ static struct hda_pcm_stream ad198x_pcm_digital_playback = { .ops = { .open = ad198x_dig_playback_pcm_open, .close = ad198x_dig_playback_pcm_close, - .prepare = ad198x_dig_playback_pcm_prepare + .prepare = ad198x_dig_playback_pcm_prepare, + .cleanup = ad198x_dig_playback_pcm_cleanup }, }; @@ -397,7 +432,8 @@ static void ad198x_free(struct hda_codec *codec) return; ad198x_free_kctls(codec); - kfree(codec->spec); + kfree(spec); + snd_hda_detach_beep_device(codec); } static struct hda_codec_ops ad198x_patch_ops = { @@ -536,8 +572,6 @@ static struct snd_kcontrol_new ad1986a_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), @@ -601,8 +635,7 @@ static struct snd_kcontrol_new ad1986a_laptop_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x18, 0x0, HDA_OUTPUT), + /* HDA_CODEC_VOLUME("Mono Playback Volume", 0x1e, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mono Playback Switch", 0x1e, 0x0, HDA_OUTPUT), */ HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), @@ -800,8 +833,6 @@ static struct snd_kcontrol_new ad1986a_laptop_automute_mixers[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0f, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x0, HDA_OUTPUT), { @@ -993,10 +1024,8 @@ static struct snd_pci_quirk ad1986a_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_SAMSUNG), - SND_PCI_QUIRK(0x144d, 0xc024, "Samsung R65", AD1986A_SAMSUNG), - SND_PCI_QUIRK(0x144d, 0xc026, "Samsung X11", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc027, "Samsung Q1", AD1986A_ULTRA), + SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc000, "Samsung", AD1986A_SAMSUNG), SND_PCI_QUIRK(0x144d, 0xc504, "Samsung Q35", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x1011, "Lenovo M55", AD1986A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), @@ -1018,15 +1047,14 @@ static struct hda_amp_list ad1986a_loopbacks[] = { static int is_jack_available(struct hda_codec *codec, hda_nid_t nid) { - unsigned int conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + unsigned int conf = snd_hda_codec_get_pincfg(codec, nid); return get_defcfg_connect(conf) != AC_JACK_PORT_NONE; } static int patch_ad1986a(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1034,6 +1062,13 @@ static int patch_ad1986a(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x19); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x18, 0, HDA_OUTPUT); + spec->multiout.max_channels = 6; spec->multiout.num_dacs = ARRAY_SIZE(ad1986a_dac_nids); spec->multiout.dac_nids = ad1986a_dac_nids; @@ -1213,8 +1248,6 @@ static struct snd_kcontrol_new ad1983_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x12, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x13, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x10, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x10, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Mic Boost", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x15, 0x0, HDA_OUTPUT), @@ -1285,6 +1318,7 @@ static struct hda_amp_list ad1983_loopbacks[] = { static int patch_ad1983(struct hda_codec *codec) { struct ad198x_spec *spec; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1292,6 +1326,13 @@ static int patch_ad1983(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1983_dac_nids); spec->multiout.dac_nids = ad1983_dac_nids; @@ -1361,8 +1402,6 @@ static struct snd_kcontrol_new ad1981_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_MONO("PC Speaker Playback Volume", 0x0d, 1, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_MONO("PC Speaker Playback Switch", 0x0d, 1, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x08, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), @@ -1407,8 +1446,8 @@ static struct hda_verb ad1981_init_verbs[] = { {0x1e, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, {0x1f, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, /* Mic boost: 0dB */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb000}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Record selector: Front mic */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x0}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, 0xb080}, @@ -1673,10 +1712,10 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = { SND_PCI_QUIRK(0x1014, 0x0597, "Lenovo Z60", AD1981_THINKPAD), SND_PCI_QUIRK(0x1014, 0x05b7, "Lenovo Z60m", AD1981_THINKPAD), /* All HP models */ - SND_PCI_QUIRK(0x103c, 0, "HP nx", AD1981_HP), + SND_PCI_QUIRK_VENDOR(0x103c, "HP nx", AD1981_HP), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba U205", AD1981_TOSHIBA), /* Lenovo Thinkpad T60/X60/Z6xx */ - SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1981_THINKPAD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1981_THINKPAD), /* HP nx6320 (reversed SSID, H/W bug) */ SND_PCI_QUIRK(0x30b0, 0x103c, "HP nx6320", AD1981_HP), {} @@ -1685,7 +1724,7 @@ static struct snd_pci_quirk ad1981_cfg_tbl[] = { static int patch_ad1981(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -1693,6 +1732,13 @@ static int patch_ad1981(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x0d, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1981_dac_nids); spec->multiout.dac_nids = ad1981_dac_nids; @@ -1885,8 +1931,8 @@ static hda_nid_t ad1988_capsrc_nids[3] = { #define AD1988_SPDIF_OUT_HDMI 0x0b #define AD1988_SPDIF_IN 0x07 -static hda_nid_t ad1989b_slave_dig_outs[2] = { - AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI +static hda_nid_t ad1989b_slave_dig_outs[] = { + AD1988_SPDIF_OUT, AD1988_SPDIF_OUT_HDMI, 0 }; static struct hda_input_mux ad1988_6stack_capture_source = { @@ -1979,9 +2025,6 @@ static struct snd_kcontrol_new ad1988_6stack_mixers2[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2025,9 +2068,6 @@ static struct snd_kcontrol_new ad1988_3stack_mixers2[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x4, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x4, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2057,9 +2097,6 @@ static struct snd_kcontrol_new ad1988_laptop_mixers[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x20, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Analog Mix Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Mix Playback Switch", 0x21, 0x0, HDA_OUTPUT), @@ -2288,10 +2325,6 @@ static struct hda_verb ad1988_capture_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, { } }; @@ -2399,10 +2432,6 @@ static struct hda_verb ad1988_3stack_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Analog Mix output amp */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } @@ -2474,10 +2503,6 @@ static struct hda_verb ad1988_laptop_init_verbs[] = { {0x0c, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0d, AC_VERB_SET_CONNECT_SEL, 0x1}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* ADCs; muted */ - {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Analog Mix output amp */ {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x1f}, /* 0dB */ { } @@ -2881,7 +2906,7 @@ static int ad1988_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = AD1988_SPDIF_OUT; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = AD1988_SPDIF_IN; @@ -2931,7 +2956,7 @@ static struct snd_pci_quirk ad1988_cfg_tbl[] = { static int patch_ad1988(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2951,7 +2976,7 @@ static int patch_ad1988(struct hda_codec *codec) if (board_config == AD1988_AUTO) { /* automatic parse from the BIOS config */ - int err = ad1988_parse_auto_config(codec); + err = ad1988_parse_auto_config(codec); if (err < 0) { ad198x_free(codec); return err; @@ -2961,6 +2986,13 @@ static int patch_ad1988(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + switch (board_config) { case AD1988_6STACK: case AD1988_6STACK_DIG: @@ -3117,12 +3149,6 @@ static struct snd_kcontrol_new ad1884_base_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - /* - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("Digital Beep Playback Volume", 0x10, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Digital Beep Playback Switch", 0x10, 0x0, HDA_OUTPUT), - */ HDA_CODEC_VOLUME("Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3195,10 +3221,10 @@ static struct hda_verb ad1884_init_verbs[] = { {0x0e, AC_VERB_SET_CONNECT_SEL, 0x1}, /* Port-B (front mic) pin */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-C (rear mic) pin */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Analog mixer; mute as default */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -3230,8 +3256,8 @@ static const char *ad1884_slave_vols[] = { "Mic Playback Volume", "CD Playback Volume", "Internal Mic Playback Volume", - "Docking Mic Playback Volume" - "Beep Playback Volume", + "Docking Mic Playback Volume", + /* "Beep Playback Volume", */ "IEC958 Playback Volume", NULL }; @@ -3239,6 +3265,7 @@ static const char *ad1884_slave_vols[] = { static int patch_ad1884(struct hda_codec *codec) { struct ad198x_spec *spec; + int err; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -3246,6 +3273,13 @@ static int patch_ad1884(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1884_dac_nids); spec->multiout.dac_nids = ad1884_dac_nids; @@ -3312,8 +3346,6 @@ static struct snd_kcontrol_new ad1984_thinkpad_mixers[] = { HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Docking Mic Boost", 0x25, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x0d, 0x0, HDA_OUTPUT), @@ -3349,7 +3381,7 @@ static struct hda_verb ad1984_thinkpad_init_verbs[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* docking mic boost */ - {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x25, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, /* Analog mixer - docking mic; mute as default */ {0x20, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* enable EAPD bit */ @@ -3370,10 +3402,6 @@ static struct snd_kcontrol_new ad1984_dell_desktop_mixers[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x20, 0x00, HDA_INPUT), HDA_CODEC_VOLUME("Line-In Playback Volume", 0x20, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line-In Playback Switch", 0x20, 0x01, HDA_INPUT), - /* - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x20, 0x03, HDA_INPUT), - */ HDA_CODEC_VOLUME("Line-In Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3459,7 +3487,7 @@ static const char *ad1984_models[AD1984_MODELS] = { static struct snd_pci_quirk ad1984_cfg_tbl[] = { /* Lenovo Thinkpad T61/X61 */ - SND_PCI_QUIRK(0x17aa, 0, "Lenovo Thinkpad", AD1984_THINKPAD), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1984_THINKPAD), SND_PCI_QUIRK(0x1028, 0x0214, "Dell T3400", AD1984_DELL_DESKTOP), {} }; @@ -3552,8 +3580,6 @@ static struct snd_kcontrol_new ad1884a_base_mixers[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x02, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Line Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x25, 0x0, HDA_OUTPUT), @@ -3613,10 +3639,10 @@ static struct hda_verb ad1884a_init_verbs[] = { {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Port-B (front mic) pin */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, - {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-C (rear line-in) pin */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* Port-E (rear mic) pin */ {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -3686,8 +3712,6 @@ static struct snd_kcontrol_new ad1884a_laptop_mixers[] = { HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x20, 0x04, HDA_INPUT), HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x20, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost", 0x25, 0x0, HDA_OUTPUT), @@ -3715,8 +3739,6 @@ static struct snd_kcontrol_new ad1884a_mobile_mixers[] = { HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Capture Volume", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Capture Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3827,8 +3849,6 @@ static struct snd_kcontrol_new ad1984a_thinkpad_mixers[] = { HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x20, 0x00, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Capture Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -3902,9 +3922,9 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), - SND_PCI_QUIRK(0x103c, 0x30e6, "HP 6730b", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30e7, "HP EliteBook 8530p", AD1884A_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x3614, "HP 6730s", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3600, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK(0x17aa, 0x20ac, "Thinkpad X300", AD1884A_THINKPAD), {} }; @@ -3912,7 +3932,7 @@ static struct snd_pci_quirk ad1884a_cfg_tbl[] = { static int patch_ad1884a(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -3920,6 +3940,13 @@ static int patch_ad1884a(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(ad1884a_dac_nids); spec->multiout.dac_nids = ad1884a_dac_nids; @@ -3957,6 +3984,14 @@ static int patch_ad1884a(struct hda_codec *codec) spec->multiout.dig_out_nid = 0; codec->patch_ops.unsol_event = ad1884a_hp_unsol_event; codec->patch_ops.init = ad1884a_hp_init; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x20, HDA_INPUT, + (0x17 << AC_AMPCAP_OFFSET_SHIFT) | + (0x17 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x05 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (1 << AC_AMPCAP_MUTE_SHIFT)); break; case AD1884A_THINKPAD: spec->mixers[0] = ad1984a_thinkpad_mixers; @@ -4074,8 +4109,6 @@ static struct snd_kcontrol_new ad1882_loopback_mixers[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), { } /* end */ }; @@ -4088,8 +4121,6 @@ static struct snd_kcontrol_new ad1882a_loopback_mixers[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x20, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x20, 0x06, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x20, 0x06, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x20, 0x07, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x20, 0x07, HDA_INPUT), HDA_CODEC_VOLUME("Digital Mic Boost", 0x1f, 0x0, HDA_INPUT), { } /* end */ }; @@ -4248,7 +4279,7 @@ static const char *ad1882_models[AD1986A_MODELS] = { static int patch_ad1882(struct hda_codec *codec) { struct ad198x_spec *spec; - int board_config; + int err, board_config; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4256,6 +4287,13 @@ static int patch_ad1882(struct hda_codec *codec) codec->spec = spec; + err = snd_hda_attach_beep_device(codec, 0x10); + if (err < 0) { + ad198x_free(codec); + return err; + } + set_beep_amp(spec, 0x10, 0, HDA_OUTPUT); + spec->multiout.max_channels = 6; spec->multiout.num_dacs = 3; spec->multiout.dac_nids = ad1882_dac_nids; diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c index f3ebe83..c921264 100644 --- a/sound/pci/hda/patch_cmedia.c +++ b/sound/pci/hda/patch_cmedia.c @@ -680,13 +680,13 @@ static int patch_cmi9880(struct hda_codec *codec) struct auto_pin_cfg cfg; /* collect pin default configuration */ - port_e = snd_hda_codec_read(codec, 0x0f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_f = snd_hda_codec_read(codec, 0x10, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_e = snd_hda_codec_get_pincfg(codec, 0x0f); + port_f = snd_hda_codec_get_pincfg(codec, 0x10); spec->front_panel = 1; if (get_defcfg_connect(port_e) == AC_JACK_PORT_NONE || get_defcfg_connect(port_f) == AC_JACK_PORT_NONE) { - port_g = snd_hda_codec_read(codec, 0x1f, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_h = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_g = snd_hda_codec_get_pincfg(codec, 0x1f); + port_h = snd_hda_codec_get_pincfg(codec, 0x20); spec->channel_modes = cmi9880_channel_modes; /* no front panel */ if (get_defcfg_connect(port_g) == AC_JACK_PORT_NONE || @@ -703,8 +703,8 @@ static int patch_cmi9880(struct hda_codec *codec) spec->multiout.max_channels = cmi9880_channel_modes[0].channels; } else { spec->input_mux = &cmi9880_basic_mux; - port_spdifi = snd_hda_codec_read(codec, 0x13, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - port_spdifo = snd_hda_codec_read(codec, 0x12, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + port_spdifi = snd_hda_codec_get_pincfg(codec, 0x13); + port_spdifo = snd_hda_codec_get_pincfg(codec, 0x12); if (get_defcfg_connect(port_spdifo) != AC_JACK_PORT_NONE) spec->multiout.dig_out_nid = CMI_DIG_OUT_NID; if (get_defcfg_connect(port_spdifi) != AC_JACK_PORT_NONE) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 75de40a..1f2ad76 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -58,6 +58,7 @@ struct conexant_spec { struct snd_kcontrol_new *mixers[5]; int num_mixers; + hda_nid_t vmaster_nid; const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -72,6 +73,7 @@ struct conexant_spec { */ unsigned int cur_eapd; unsigned int hp_present; + unsigned int no_auto_mic; unsigned int need_dac_fix; /* capture */ @@ -347,6 +349,7 @@ static int conexant_mux_enum_put(struct snd_kcontrol *kcontrol, &spec->cur_mux[adc_idx]); } +#ifdef CONFIG_SND_JACK static int conexant_add_jack(struct hda_codec *codec, hda_nid_t nid, int type) { @@ -394,7 +397,6 @@ static void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) static int conexant_init_jacks(struct hda_codec *codec) { -#ifdef CONFIG_SND_JACK struct conexant_spec *spec = codec->spec; int i; @@ -422,10 +424,19 @@ static int conexant_init_jacks(struct hda_codec *codec) ++hv; } } -#endif return 0; } +#else +static inline void conexant_report_jack(struct hda_codec *codec, hda_nid_t nid) +{ +} + +static inline int conexant_init_jacks(struct hda_codec *codec) +{ + return 0; +} +#endif static int conexant_init(struct hda_codec *codec) { @@ -452,6 +463,29 @@ static void conexant_free(struct hda_codec *codec) kfree(codec->spec); } +static struct snd_kcontrol_new cxt_capture_mixers[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = conexant_mux_enum_info, + .get = conexant_mux_enum_get, + .put = conexant_mux_enum_put + }, + {} +}; + +static const char *slave_vols[] = { + "Headphone Playback Volume", + "Speaker Playback Volume", + NULL +}; + +static const char *slave_sws[] = { + "Headphone Playback Switch", + "Speaker Playback Switch", + NULL +}; + static int conexant_build_controls(struct hda_codec *codec) { struct conexant_spec *spec = codec->spec; @@ -479,6 +513,32 @@ static int conexant_build_controls(struct hda_codec *codec) if (err < 0) return err; } + + /* if we have no master control, let's create it */ + if (spec->vmaster_nid && + !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + unsigned int vmaster_tlv[4]; + snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, + HDA_OUTPUT, vmaster_tlv); + err = snd_hda_add_vmaster(codec, "Master Playback Volume", + vmaster_tlv, slave_vols); + if (err < 0) + return err; + } + if (spec->vmaster_nid && + !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + err = snd_hda_add_vmaster(codec, "Master Playback Switch", + NULL, slave_sws); + if (err < 0) + return err; + } + + if (spec->input_mux) { + err = snd_hda_add_new_ctls(codec, cxt_capture_mixers); + if (err < 0) + return err; + } + return 0; } @@ -710,13 +770,6 @@ static void cxt5045_hp_unsol_event(struct hda_codec *codec, } static struct snd_kcontrol_new cxt5045_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), @@ -750,13 +803,6 @@ static struct snd_kcontrol_new cxt5045_benq_mixers[] = { }; static struct snd_kcontrol_new cxt5045_mixers_hp530[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, HDA_CODEC_VOLUME("Int Mic Capture Volume", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Capture Switch", 0x1a, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Ext Mic Capture Volume", 0x1a, 0x01, HDA_INPUT), @@ -993,15 +1039,9 @@ static const char *cxt5045_models[CXT5045_MODELS] = { }; static struct snd_pci_quirk cxt5045_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a5, "HP", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV Series", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2120", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30b7, "HP DV6000Z", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30bb, "HP DV8000", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30cd, "HP DV Series", CXT5045_LAPTOP_HPSENSE), - SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV9533EG", CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x103c, 0x30d5, "HP 530", CXT5045_LAPTOP_HP530), - SND_PCI_QUIRK(0x103c, 0x30d9, "HP Spartan", CXT5045_LAPTOP_HPSENSE), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", + CXT5045_LAPTOP_HPSENSE), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P105", CXT5045_LAPTOP_MICSENSE), SND_PCI_QUIRK(0x152d, 0x0753, "Benq R55E", CXT5045_BENQ), SND_PCI_QUIRK(0x1734, 0x10ad, "Fujitsu Si1520", CXT5045_LAPTOP_MICSENSE), @@ -1011,8 +1051,8 @@ static struct snd_pci_quirk cxt5045_cfg_tbl[] = { SND_PCI_QUIRK(0x1509, 0x1e40, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f05, "FIC", CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x1509, 0x2f06, "FIC", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1631, 0xc106, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE), - SND_PCI_QUIRK(0x1631, 0xc107, "Packard Bell", CXT5045_LAPTOP_HPMICSENSE), + SND_PCI_QUIRK_MASK(0x1631, 0xff00, 0xc100, "Packard Bell", + CXT5045_LAPTOP_HPMICSENSE), SND_PCI_QUIRK(0x8086, 0x2111, "Conexant Reference board", CXT5045_LAPTOP_HPSENSE), {} }; @@ -1026,6 +1066,7 @@ static int patch_cxt5045(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5045_dac_nids); @@ -1125,7 +1166,7 @@ static int patch_cxt5045(struct hda_codec *codec) /* Conexant 5047 specific */ #define CXT5047_SPDIF_OUT 0x11 -static hda_nid_t cxt5047_dac_nids[2] = { 0x10, 0x1c }; +static hda_nid_t cxt5047_dac_nids[1] = { 0x10 }; /* 0x1c */ static hda_nid_t cxt5047_adc_nids[1] = { 0x12 }; static hda_nid_t cxt5047_capsrc_nids[1] = { 0x1a }; @@ -1133,20 +1174,6 @@ static struct hda_channel_mode cxt5047_modes[1] = { { 2, NULL }, }; -static struct hda_input_mux cxt5047_capture_source = { - .num_items = 1, - .items = { - { "Mic", 0x2 }, - } -}; - -static struct hda_input_mux cxt5047_hp_capture_source = { - .num_items = 1, - .items = { - { "ExtMic", 0x2 }, - } -}; - static struct hda_input_mux cxt5047_toshiba_capture_source = { .num_items = 2, .items = { @@ -1170,7 +1197,11 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, * the headphone jack */ bits = (!spec->hp_present && spec->cur_eapd) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, + /* NOTE: Conexat codec needs the index for *OUTPUT* amp of + * pin widgets unlike other codecs. In this case, we need to + * set index 0x01 for the volume from the mixer amp 0x19. + */ + snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); bits = spec->cur_eapd ? 0 : HDA_AMP_MUTE; snd_hda_codec_amp_stereo(codec, 0x13, HDA_OUTPUT, 0, @@ -1178,16 +1209,6 @@ static int cxt5047_hp_master_sw_put(struct snd_kcontrol *kcontrol, return 1; } -/* bind volumes of both NID 0x13 (Headphones) and 0x1d (Speakers) */ -static struct hda_bind_ctls cxt5047_bind_master_vol = { - .ops = &snd_hda_bind_vol, - .values = { - HDA_COMPOSE_AMP_VAL(0x13, 3, 0, HDA_OUTPUT), - HDA_COMPOSE_AMP_VAL(0x1d, 3, 0, HDA_OUTPUT), - 0 - }, -}; - /* mute internal speaker if HP is plugged */ static void cxt5047_hp_automute(struct hda_codec *codec) { @@ -1198,27 +1219,8 @@ static void cxt5047_hp_automute(struct hda_codec *codec) AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = (spec->hp_present || !spec->cur_eapd) ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - /* Mute/Unmute PCM 2 for good measure - some systems need this */ - snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); -} - -/* mute internal speaker if HP is plugged */ -static void cxt5047_hp2_automute(struct hda_codec *codec) -{ - struct conexant_spec *spec = codec->spec; - unsigned int bits; - - spec->hp_present = snd_hda_codec_read(codec, 0x13, 0, - AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; - - bits = spec->hp_present ? HDA_AMP_MUTE : 0; - snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0, - HDA_AMP_MUTE, bits); - /* Mute/Unmute PCM 2 for good measure - some systems need this */ - snd_hda_codec_amp_stereo(codec, 0x1c, HDA_OUTPUT, 0, + /* See the note in cxt5047_hp_master_sw_put */ + snd_hda_codec_amp_stereo(codec, 0x1d, HDA_OUTPUT, 0x01, HDA_AMP_MUTE, bits); } @@ -1259,55 +1261,14 @@ static void cxt5047_hp_unsol_event(struct hda_codec *codec, } } -/* unsolicited event for HP jack sensing - non-EAPD systems */ -static void cxt5047_hp2_unsol_event(struct hda_codec *codec, - unsigned int res) -{ - res >>= 26; - switch (res) { - case CONEXANT_HP_EVENT: - cxt5047_hp2_automute(codec); - break; - case CONEXANT_MIC_EVENT: - cxt5047_hp_automic(codec); - break; - } -} - -static struct snd_kcontrol_new cxt5047_mixers[] = { - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Mic Gain Volume", 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("Mic Gain Switch", 0x1a, 0x0, HDA_OUTPUT), +static struct snd_kcontrol_new cxt5047_base_mixers[] = { + HDA_CODEC_VOLUME("Mic Playback Volume", 0x19, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x19, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x1a, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("PCM-2 Volume", 0x1c, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM-2 Switch", 0x1c, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x1d, 0x00, HDA_OUTPUT), - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x13, 0x00, HDA_OUTPUT), - - {} -}; - -static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), - HDA_BIND_VOL("Master Playback Volume", &cxt5047_bind_master_vol), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Master Playback Switch", @@ -1320,29 +1281,15 @@ static struct snd_kcontrol_new cxt5047_toshiba_mixers[] = { {} }; -static struct snd_kcontrol_new cxt5047_hp_mixers[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = conexant_mux_enum_info, - .get = conexant_mux_enum_get, - .put = conexant_mux_enum_put - }, - HDA_CODEC_VOLUME("Mic Bypass Capture Volume", 0x19, 0x02, HDA_INPUT), - HDA_CODEC_MUTE("Mic Bypass Capture Switch", 0x19,0x02,HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x12, 0x03, HDA_INPUT), - HDA_CODEC_VOLUME("PCM Volume", 0x10, 0x00, HDA_OUTPUT), - HDA_CODEC_MUTE("PCM Switch", 0x10, 0x00, HDA_OUTPUT), +static struct snd_kcontrol_new cxt5047_hp_spk_mixers[] = { + /* See the note in cxt5047_hp_master_sw_put */ + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x1d, 0x01, HDA_OUTPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x13, 0x00, HDA_OUTPUT), + {} +}; + +static struct snd_kcontrol_new cxt5047_hp_only_mixers[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x13, 0x00, HDA_OUTPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Master Playback Switch", - .info = cxt_eapd_info, - .get = cxt_eapd_get, - .put = cxt5047_hp_master_sw_put, - .private_value = 0x13, - }, { } /* end */ }; @@ -1353,8 +1300,8 @@ static struct hda_verb cxt5047_init_verbs[] = { {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN|AC_PINCTL_VREF_50 }, /* HP, Speaker */ {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, - {0x13, AC_VERB_SET_CONNECT_SEL,0x1}, - {0x1d, AC_VERB_SET_CONNECT_SEL,0x0}, + {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, /* mixer(0x19) */ + {0x1d, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mixer(0x19) */ /* Record selector: Mic */ {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, @@ -1374,30 +1321,7 @@ static struct hda_verb cxt5047_init_verbs[] = { /* configuration for Toshiba Laptops */ static struct hda_verb cxt5047_toshiba_init_verbs[] = { - {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0 }, /* default on */ - /* pin sensing on HP and Mic jacks */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, - /* Speaker routing */ - {0x1d, AC_VERB_SET_CONNECT_SEL,0x1}, - {} -}; - -/* configuration for HP Laptops */ -static struct hda_verb cxt5047_hp_init_verbs[] = { - /* pin sensing on HP jack */ - {0x13, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, - /* 0x13 is actually shared by both HP and speaker; - * setting the connection to 0 (=0x19) makes the master volume control - * working mysteriouslly... - */ - {0x13, AC_VERB_SET_CONNECT_SEL, 0x0}, - /* Record selector: Ext Mic */ - {0x12, AC_VERB_SET_CONNECT_SEL,0x03}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_INPUT|AC_AMP_SET_RIGHT|AC_AMP_SET_LEFT|0x17}, - /* Speaker routing */ - {0x1d, AC_VERB_SET_CONNECT_SEL,0x1}, + {0x13, AC_VERB_SET_EAPD_BTLENABLE, 0x0}, /* default off */ {} }; @@ -1562,10 +1486,9 @@ static const char *cxt5047_models[CXT5047_MODELS] = { }; static struct snd_pci_quirk cxt5047_cfg_tbl[] = { - SND_PCI_QUIRK(0x103c, 0x30a0, "HP DV1000", CXT5047_LAPTOP), SND_PCI_QUIRK(0x103c, 0x30a5, "HP DV5200T/DV8000T", CXT5047_LAPTOP_HP), - SND_PCI_QUIRK(0x103c, 0x30b2, "HP DV2000T/DV3000T", CXT5047_LAPTOP), - SND_PCI_QUIRK(0x103c, 0x30b5, "HP DV2000Z", CXT5047_LAPTOP), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x3000, "HP DV Series", + CXT5047_LAPTOP), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba P100", CXT5047_LAPTOP_EAPD), {} }; @@ -1579,6 +1502,7 @@ static int patch_cxt5047(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; spec->multiout.max_channels = 2; spec->multiout.num_dacs = ARRAY_SIZE(cxt5047_dac_nids); @@ -1587,9 +1511,8 @@ static int patch_cxt5047(struct hda_codec *codec) spec->num_adc_nids = 1; spec->adc_nids = cxt5047_adc_nids; spec->capsrc_nids = cxt5047_capsrc_nids; - spec->input_mux = &cxt5047_capture_source; spec->num_mixers = 1; - spec->mixers[0] = cxt5047_mixers; + spec->mixers[0] = cxt5047_base_mixers; spec->num_init_verbs = 1; spec->init_verbs[0] = cxt5047_init_verbs; spec->spdif_route = 0; @@ -1603,21 +1526,22 @@ static int patch_cxt5047(struct hda_codec *codec) cxt5047_cfg_tbl); switch (board_config) { case CXT5047_LAPTOP: - codec->patch_ops.unsol_event = cxt5047_hp2_unsol_event; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_spk_mixers; + codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; break; case CXT5047_LAPTOP_HP: - spec->input_mux = &cxt5047_hp_capture_source; - spec->num_init_verbs = 2; - spec->init_verbs[1] = cxt5047_hp_init_verbs; - spec->mixers[0] = cxt5047_hp_mixers; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_only_mixers; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; codec->patch_ops.init = cxt5047_hp_init; break; case CXT5047_LAPTOP_EAPD: spec->input_mux = &cxt5047_toshiba_capture_source; + spec->num_mixers = 2; + spec->mixers[1] = cxt5047_hp_spk_mixers; spec->num_init_verbs = 2; spec->init_verbs[1] = cxt5047_toshiba_init_verbs; - spec->mixers[0] = cxt5047_toshiba_mixers; codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; break; #ifdef CONFIG_SND_DEBUG @@ -1628,6 +1552,7 @@ static int patch_cxt5047(struct hda_codec *codec) codec->patch_ops.unsol_event = cxt5047_hp_unsol_event; #endif } + spec->vmaster_nid = 0x13; return 0; } @@ -1663,8 +1588,11 @@ static int cxt5051_hp_master_sw_put(struct snd_kcontrol *kcontrol, /* toggle input of built-in and mic jack appropriately */ static void cxt5051_portb_automic(struct hda_codec *codec) { + struct conexant_spec *spec = codec->spec; unsigned int present; + if (spec->no_auto_mic) + return; present = snd_hda_codec_read(codec, 0x17, 0, AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; @@ -1680,6 +1608,8 @@ static void cxt5051_portc_automic(struct hda_codec *codec) unsigned int present; hda_nid_t new_adc; + if (spec->no_auto_mic) + return; present = snd_hda_codec_read(codec, 0x18, 0, AC_VERB_GET_PIN_SENSE, 0) & AC_PINSENSE_PRESENCE; @@ -1766,6 +1696,22 @@ static struct snd_kcontrol_new cxt5051_hp_mixers[] = { {} }; +static struct snd_kcontrol_new cxt5051_hp_dv6736_mixers[] = { + HDA_CODEC_VOLUME("Mic Volume", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_MUTE("Mic Switch", 0x14, 0x00, HDA_INPUT), + HDA_CODEC_VOLUME("Master Playback Volume", 0x10, 0x00, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = cxt_eapd_info, + .get = cxt_eapd_get, + .put = cxt5051_hp_master_sw_put, + .private_value = 0x1a, + }, + + {} +}; + static struct hda_verb cxt5051_init_verbs[] = { /* Line in, Mic */ {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, @@ -1796,6 +1742,66 @@ static struct hda_verb cxt5051_init_verbs[] = { { } /* end */ }; +static struct hda_verb cxt5051_hp_dv6736_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x1}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + { } /* end */ +}; + +static struct hda_verb cxt5051_lenovo_x200_init_verbs[] = { + /* Line in, Mic */ + {0x17, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x03}, + /* SPK */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* HP, Amp */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Docking HP */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x19, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* DAC1 */ + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Record selector: Int mic */ + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1) | 0x44}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x44}, + /* SPDIF route: PCM */ + {0x1c, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* EAPD */ + {0x1a, AC_VERB_SET_EAPD_BTLENABLE, 0x2}, /* default on */ + {0x16, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + {0x17, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTB_EVENT}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CXT5051_PORTC_EVENT}, + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN|CONEXANT_HP_EVENT}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5051_init(struct hda_codec *codec) { @@ -1813,18 +1819,24 @@ static int cxt5051_init(struct hda_codec *codec) enum { CXT5051_LAPTOP, /* Laptops w/ EAPD support */ CXT5051_HP, /* no docking */ + CXT5051_HP_DV6736, /* HP without mic switch */ + CXT5051_LENOVO_X200, /* Lenovo X200 laptop */ CXT5051_MODELS }; static const char *cxt5051_models[CXT5051_MODELS] = { [CXT5051_LAPTOP] = "laptop", [CXT5051_HP] = "hp", + [CXT5051_HP_DV6736] = "hp-dv6736", + [CXT5051_LENOVO_X200] = "lenovo-x200", }; static struct snd_pci_quirk cxt5051_cfg_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x30cf, "HP DV6736", CXT5051_HP_DV6736), SND_PCI_QUIRK(0x14f1, 0x0101, "Conexant Reference board", CXT5051_LAPTOP), SND_PCI_QUIRK(0x14f1, 0x5051, "HP Spartan 1.1", CXT5051_HP), + SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo X200", CXT5051_LENOVO_X200), {} }; @@ -1837,6 +1849,7 @@ static int patch_cxt5051(struct hda_codec *codec) if (!spec) return -ENOMEM; codec->spec = spec; + codec->pin_amp_workaround = 1; codec->patch_ops = conexant_patch_ops; codec->patch_ops.init = cxt5051_init; @@ -1857,17 +1870,22 @@ static int patch_cxt5051(struct hda_codec *codec) spec->cur_adc = 0; spec->cur_adc_idx = 0; + codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; + board_config = snd_hda_check_board_config(codec, CXT5051_MODELS, cxt5051_models, cxt5051_cfg_tbl); switch (board_config) { case CXT5051_HP: - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; spec->mixers[0] = cxt5051_hp_mixers; break; - default: - case CXT5051_LAPTOP: - codec->patch_ops.unsol_event = cxt5051_hp_unsol_event; + case CXT5051_HP_DV6736: + spec->init_verbs[0] = cxt5051_hp_dv6736_init_verbs; + spec->mixers[0] = cxt5051_hp_dv6736_mixers; + spec->no_auto_mic = 1; + break; + case CXT5051_LENOVO_X200: + spec->init_verbs[0] = cxt5051_lenovo_x200_init_verbs; break; } diff --git a/sound/pci/hda/patch_intelhdmi.c b/sound/pci/hda/patch_intelhdmi.c index 3564f4e..fcc77fe 100644 --- a/sound/pci/hda/patch_intelhdmi.c +++ b/sound/pci/hda/patch_intelhdmi.c @@ -49,11 +49,6 @@ static struct hda_verb pinout_enable_verb[] = { {} /* terminator */ }; -static struct hda_verb pinout_disable_verb[] = { - {PIN_NID, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x00}, - {} -}; - static struct hda_verb unsolicited_response_verb[] = { {PIN_NID, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | INTEL_HDMI_EVENT_TAG}, @@ -248,10 +243,6 @@ static void hdmi_write_dip_byte(struct hda_codec *codec, hda_nid_t nid, static void hdmi_enable_output(struct hda_codec *codec) { - /* Enable Audio InfoFrame Transmission */ - hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); - snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, - AC_DIPXMIT_BEST); /* Unmute */ if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, PIN_NID, 0, @@ -260,17 +251,24 @@ static void hdmi_enable_output(struct hda_codec *codec) snd_hda_sequence_write(codec, pinout_enable_verb); } -static void hdmi_disable_output(struct hda_codec *codec) +/* + * Enable Audio InfoFrame Transmission + */ +static void hdmi_start_infoframe_trans(struct hda_codec *codec) { - snd_hda_sequence_write(codec, pinout_disable_verb); - if (get_wcaps(codec, PIN_NID) & AC_WCAP_OUT_AMP) - snd_hda_codec_write(codec, PIN_NID, 0, - AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_BEST); +} - /* - * FIXME: noises may arise when playing music after reloading the - * kernel module, until the next X restart or monitor repower. - */ +/* + * Disable Audio InfoFrame Transmission + */ +static void hdmi_stop_infoframe_trans(struct hda_codec *codec) +{ + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); + snd_hda_codec_write(codec, PIN_NID, 0, AC_VERB_SET_HDMI_DIP_XMIT, + AC_DIPXMIT_DISABLE); } static int hdmi_get_channel_count(struct hda_codec *codec) @@ -368,11 +366,16 @@ static void hdmi_fill_audio_infoframe(struct hda_codec *codec, struct hdmi_audio_infoframe *ai) { u8 *params = (u8 *)ai; + u8 sum = 0; int i; hdmi_debug_dip_size(codec); hdmi_clear_dip_buffers(codec); /* be paranoid */ + for (i = 0; i < sizeof(ai); i++) + sum += params[i]; + ai->checksum = - sum; + hdmi_set_dip_index(codec, PIN_NID, 0x0, 0x0); for (i = 0; i < sizeof(ai); i++) hdmi_write_dip_byte(codec, PIN_NID, params[i]); @@ -419,14 +422,18 @@ static int hdmi_setup_channel_allocation(struct hda_codec *codec, /* * CA defaults to 0 for basic stereo audio */ - if (!eld->eld_ver) - return 0; - if (!eld->spk_alloc) - return 0; if (channels <= 2) return 0; /* + * HDMI sink's ELD info cannot always be retrieved for now, e.g. + * in console or for audio devices. Assume the highest speakers + * configuration, to _not_ prohibit multi-channel audio playback. + */ + if (!eld->spk_alloc) + eld->spk_alloc = 0xffff; + + /* * expand ELD's speaker allocation mask * * ELD tells the speaker mask in a compact(paired) form, @@ -485,6 +492,7 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, hdmi_setup_channel_mapping(codec, &ai); hdmi_fill_audio_infoframe(codec, &ai); + hdmi_start_infoframe_trans(codec); } @@ -562,7 +570,7 @@ static int intel_hdmi_playback_pcm_close(struct hda_pcm_stream *hinfo, { struct intel_hdmi_spec *spec = codec->spec; - hdmi_disable_output(codec); + hdmi_stop_infoframe_trans(codec); return snd_hda_multi_out_dig_close(codec, &spec->multiout); } @@ -582,8 +590,6 @@ static int intel_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, substream); - hdmi_enable_output(codec); - return 0; } @@ -628,8 +634,7 @@ static int intel_hdmi_build_controls(struct hda_codec *codec) static int intel_hdmi_init(struct hda_codec *codec) { - /* disable audio output as early as possible */ - hdmi_disable_output(codec); + hdmi_enable_output(codec); snd_hda_sequence_write(codec, unsolicited_response_verb); @@ -679,6 +684,7 @@ static struct hda_codec_preset snd_hda_preset_intelhdmi[] = { { .id = 0x80862801, .name = "G45 DEVBLC", .patch = patch_intel_hdmi }, { .id = 0x80862802, .name = "G45 DEVCTG", .patch = patch_intel_hdmi }, { .id = 0x80862803, .name = "G45 DEVELK", .patch = patch_intel_hdmi }, + { .id = 0x80862804, .name = "G45 DEVIBX", .patch = patch_intel_hdmi }, { .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_intel_hdmi }, {} /* terminator */ }; @@ -687,6 +693,7 @@ MODULE_ALIAS("snd-hda-codec-id:808629fb"); MODULE_ALIAS("snd-hda-codec-id:80862801"); MODULE_ALIAS("snd-hda-codec-id:80862802"); MODULE_ALIAS("snd-hda-codec-id:80862803"); +MODULE_ALIAS("snd-hda-codec-id:80862804"); MODULE_ALIAS("snd-hda-codec-id:10951392"); MODULE_LICENSE("GPL"); diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 96952a3..d57d813 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -160,6 +160,7 @@ static int patch_nvhdmi(struct hda_codec *codec) */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "MCP78 HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de0006, .name = "MCP78 HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0007, .name = "MCP7A HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi }, @@ -167,6 +168,7 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { }; MODULE_ALIAS("snd-hda-codec-id:10de0002"); +MODULE_ALIAS("snd-hda-codec-id:10de0006"); MODULE_ALIAS("snd-hda-codec-id:10de0007"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ea4c88f..f35e58a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -30,6 +30,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include "hda_beep.h" #define ALC880_FRONT_EVENT 0x01 #define ALC880_DCVOL_EVENT 0x02 @@ -77,6 +78,7 @@ enum { ALC260_ACER, ALC260_WILL, ALC260_REPLACER_672V, + ALC260_FAVORIT100, #ifdef CONFIG_SND_DEBUG ALC260_TEST, #endif @@ -103,6 +105,7 @@ enum { ALC262_NEC, ALC262_TOSHIBA_S06, ALC262_TOSHIBA_RX1, + ALC262_TYAN, ALC262_AUTO, ALC262_MODEL_LAST /* last tag */ }; @@ -238,6 +241,13 @@ enum { ALC883_MODEL_LAST, }; +/* styles of capture selection */ +enum { + CAPT_MUX = 0, /* only mux based */ + CAPT_MIX, /* only mixer based */ + CAPT_1MUX_MIX, /* first mux and other mixers */ +}; + /* for GPIO Poll */ #define GPIO_MASK 0x03 @@ -246,6 +256,7 @@ struct alc_spec { struct snd_kcontrol_new *mixers[5]; /* mixer arrays */ unsigned int num_mixers; struct snd_kcontrol_new *cap_mixer; /* capture mixer */ + unsigned int beep_amp; /* beep amp value, set via set_beep_amp() */ const struct hda_verb *init_verbs[5]; /* initialization verbs * don't forget NULL @@ -269,13 +280,15 @@ struct alc_spec { * dig_out_nid and hp_nid are optional */ hda_nid_t alt_dac_nid; + hda_nid_t slave_dig_outs[3]; /* optional - for auto-parsing */ + int dig_out_type; /* capture */ unsigned int num_adc_nids; hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; hda_nid_t dig_in_nid; /* digital-in NID; optional */ - unsigned char is_mix_capture; /* matrix-style capture (non-mux) */ + int capture_style; /* capture style (CAPT_*) */ /* capture source */ unsigned int num_mux_defs; @@ -293,7 +306,7 @@ struct alc_spec { /* dynamic controls, init_verbs and input_mux */ struct auto_pin_cfg autocfg; struct snd_array kctls; - struct hda_input_mux private_imux; + struct hda_input_mux private_imux[3]; hda_nid_t private_dac_nids[AUTO_CFG_MAX_OUTS]; /* hooks */ @@ -305,6 +318,9 @@ struct alc_spec { unsigned int jack_present: 1; unsigned int master_sw: 1; + /* other flags */ + unsigned int no_analog :1; /* digital I/O only */ + /* for virtual master */ hda_nid_t vmaster_nid; #ifdef CONFIG_SND_HDA_POWER_SAVE @@ -314,13 +330,6 @@ struct alc_spec { /* for PLL fix */ hda_nid_t pll_nid; unsigned int pll_coef_idx, pll_coef_bit; - -#ifdef SND_HDA_NEEDS_RESUME -#define ALC_MAX_PINS 16 - unsigned int num_pins; - hda_nid_t pin_nids[ALC_MAX_PINS]; - unsigned int pin_cfgs[ALC_MAX_PINS]; -#endif }; /* @@ -336,6 +345,7 @@ struct alc_config_preset { hda_nid_t *dac_nids; hda_nid_t dig_out_nid; /* optional */ hda_nid_t hp_nid; /* optional */ + hda_nid_t *slave_dig_outs; unsigned int num_adc_nids; hda_nid_t *adc_nids; hda_nid_t *capsrc_nids; @@ -392,7 +402,8 @@ static int alc_mux_enum_put(struct snd_kcontrol *kcontrol, mux_idx = adc_idx >= spec->num_mux_defs ? 0 : adc_idx; imux = &spec->input_mux[mux_idx]; - if (spec->is_mix_capture) { + if (spec->capture_style && + !(spec->capture_style == CAPT_1MUX_MIX && !adc_idx)) { /* Matrix-mixer style (e.g. ALC882) */ unsigned int *cur_val = &spec->cur_mux[adc_idx]; unsigned int i, idx; @@ -750,6 +761,24 @@ static int alc_eapd_ctrl_put(struct snd_kcontrol *kcontrol, #endif /* CONFIG_SND_DEBUG */ /* + * set up the input pin config (depending on the given auto-pin type) + */ +static void alc_set_input_pin(struct hda_codec *codec, hda_nid_t nid, + int auto_pin_type) +{ + unsigned int val = PIN_IN; + + if (auto_pin_type <= AUTO_PIN_FRONT_MIC) { + unsigned int pincap; + pincap = snd_hda_query_pin_caps(codec, nid); + pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; + if (pincap & AC_PINCAP_VREF_80) + val = PIN_VREF80; + } + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, val); +} + +/* */ static void add_mixer(struct alc_spec *spec, struct snd_kcontrol_new *mix) { @@ -810,6 +839,7 @@ static void setup_preset(struct alc_spec *spec, spec->multiout.num_dacs = preset->num_dacs; spec->multiout.dac_nids = preset->dac_nids; spec->multiout.dig_out_nid = preset->dig_out_nid; + spec->multiout.slave_dig_outs = preset->slave_dig_outs; spec->multiout.hp_nid = preset->hp_nid; spec->num_mux_defs = preset->num_mux_defs; @@ -921,7 +951,7 @@ static void alc_mic_automute(struct hda_codec *codec) HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); } #else -#define alc_mic_automute(codec) /* NOP */ +#define alc_mic_automute(codec) do {} while(0) /* NOP */ #endif /* disabled */ /* unsolicited event for HP jack sensing */ @@ -952,7 +982,7 @@ static void alc888_coef_init(struct hda_codec *codec) snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0); tmp = snd_hda_codec_read(codec, 0x20, 0, AC_VERB_GET_PROC_COEF, 0); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); - if ((tmp & 0xf0) == 2) + if ((tmp & 0xf0) == 0x20) /* alc888S-VC */ snd_hda_codec_read(codec, 0x20, 0, AC_VERB_SET_PROC_COEF, 0x830); @@ -991,8 +1021,7 @@ static void alc_subsystem_id(struct hda_codec *codec, nid = 0x1d; if (codec->vendor_id == 0x10ec0260) nid = 0x17; - ass = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + ass = snd_hda_codec_get_pincfg(codec, nid); if (!(ass & 1) && !(ass & 0x100000)) return; if ((ass >> 30) != 1) /* no physical connection */ @@ -1037,6 +1066,7 @@ do_sku: case 0x10ec0267: case 0x10ec0268: case 0x10ec0269: + case 0x10ec0272: case 0x10ec0660: case 0x10ec0662: case 0x10ec0663: @@ -1065,6 +1095,7 @@ do_sku: case 0x10ec0882: case 0x10ec0883: case 0x10ec0885: + case 0x10ec0887: case 0x10ec0889: snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 7); @@ -1164,16 +1195,8 @@ static void alc_fix_pincfg(struct hda_codec *codec, return; cfg = pinfix[quirk->value]; - for (; cfg->nid; cfg++) { - int i; - u32 val = cfg->val; - for (i = 0; i < 4; i++) { - snd_hda_codec_write(codec, cfg->nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 + i, - val & 0xff); - val >>= 8; - } - } + for (; cfg->nid; cfg++) + snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } /* @@ -1373,8 +1396,6 @@ static struct snd_kcontrol_new alc888_base_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -1481,8 +1502,6 @@ static struct snd_kcontrol_new alc880_three_stack_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x3, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x19, 0x0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -1576,8 +1595,7 @@ static int alc_cap_sw_put(struct snd_kcontrol *kcontrol, snd_hda_mixer_amp_switch_put); } -#define DEFINE_CAPMIX(num) \ -static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ +#define _DEFINE_CAPMIX(num) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Capture Switch", \ @@ -1598,7 +1616,9 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ .get = alc_cap_vol_get, \ .put = alc_cap_vol_put, \ .tlv = { .c = alc_cap_vol_tlv }, \ - }, \ + } + +#define _DEFINE_CAPSRC(num) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ /* .name = "Capture Source", */ \ @@ -1607,15 +1627,28 @@ static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ .info = alc_mux_enum_info, \ .get = alc_mux_enum_get, \ .put = alc_mux_enum_put, \ - }, \ - { } /* end */ \ + } + +#define DEFINE_CAPMIX(num) \ +static struct snd_kcontrol_new alc_capture_mixer ## num[] = { \ + _DEFINE_CAPMIX(num), \ + _DEFINE_CAPSRC(num), \ + { } /* end */ \ +} + +#define DEFINE_CAPMIX_NOSRC(num) \ +static struct snd_kcontrol_new alc_capture_mixer_nosrc ## num[] = { \ + _DEFINE_CAPMIX(num), \ + { } /* end */ \ } /* up to three ADCs */ DEFINE_CAPMIX(1); DEFINE_CAPMIX(2); DEFINE_CAPMIX(3); - +DEFINE_CAPMIX_NOSRC(1); +DEFINE_CAPMIX_NOSRC(2); +DEFINE_CAPMIX_NOSRC(3); /* * ALC880 5-stack model @@ -1704,8 +1737,6 @@ static struct snd_kcontrol_new alc880_six_stack_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -1882,13 +1913,6 @@ static struct snd_kcontrol_new alc880_asus_w1v_mixer[] = { { } /* end */ }; -/* additional mixers to alc880_asus_mixer */ -static struct snd_kcontrol_new alc880_pcbeep_mixer[] = { - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ -}; - /* TCL S700 */ static struct snd_kcontrol_new alc880_tcl_s700_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), @@ -1921,8 +1945,6 @@ static struct snd_kcontrol_new alc880_uniwill_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Channel Mode", @@ -1997,6 +2019,13 @@ static const char *alc_slave_sws[] = { static void alc_free_kctls(struct hda_codec *codec); +/* additional beep mixers; the actual parameters are overwritten at build */ +static struct snd_kcontrol_new alc_beep_mixer[] = { + HDA_CODEC_VOLUME("Beep Playback Volume", 0, 0, HDA_INPUT), + HDA_CODEC_MUTE("Beep Playback Switch", 0, 0, HDA_INPUT), + { } /* end */ +}; + static int alc_build_controls(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -2018,11 +2047,13 @@ static int alc_build_controls(struct hda_codec *codec) spec->multiout.dig_out_nid); if (err < 0) return err; - err = snd_hda_create_spdif_share_sw(codec, - &spec->multiout); - if (err < 0) - return err; - spec->multiout.share_spdif = 1; + if (!spec->no_analog) { + err = snd_hda_create_spdif_share_sw(codec, + &spec->multiout); + if (err < 0) + return err; + spec->multiout.share_spdif = 1; + } } if (spec->dig_in_nid) { err = snd_hda_create_spdif_in_ctls(codec, spec->dig_in_nid); @@ -2030,8 +2061,24 @@ static int alc_build_controls(struct hda_codec *codec) return err; } + /* create beep controls if needed */ + if (spec->beep_amp) { + struct snd_kcontrol_new *knew; + for (knew = alc_beep_mixer; knew->name; knew++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_new1(knew, codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = spec->beep_amp; + err = snd_hda_ctl_add(codec, kctl); + if (err < 0) + return err; + } + } + /* if we have no master control, let's create it */ - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { + if (!spec->no_analog && + !snd_hda_find_mixer_ctl(codec, "Master Playback Volume")) { unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->vmaster_nid, HDA_OUTPUT, vmaster_tlv); @@ -2040,7 +2087,8 @@ static int alc_build_controls(struct hda_codec *codec) if (err < 0) return err; } - if (!snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { + if (!spec->no_analog && + !snd_hda_find_mixer_ctl(codec, "Master Playback Switch")) { err = snd_hda_add_vmaster(codec, "Master Playback Switch", NULL, alc_slave_sws); if (err < 0) @@ -2949,6 +2997,14 @@ static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static int alc880_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -3032,7 +3088,8 @@ static struct hda_pcm_stream alc880_pcm_digital_playback = { .ops = { .open = alc880_dig_playback_pcm_open, .close = alc880_dig_playback_pcm_close, - .prepare = alc880_dig_playback_pcm_prepare + .prepare = alc880_dig_playback_pcm_prepare, + .cleanup = alc880_dig_playback_pcm_cleanup }, }; @@ -3059,6 +3116,9 @@ static int alc_build_pcms(struct hda_codec *codec) codec->num_pcms = 1; codec->pcm_info = info; + if (spec->no_analog) + goto skip_analog; + info->name = spec->stream_name_analog; if (spec->stream_analog_playback) { if (snd_BUG_ON(!spec->multiout.dac_nids)) @@ -3082,12 +3142,17 @@ static int alc_build_pcms(struct hda_codec *codec) } } + skip_analog: /* SPDIF for stream index #1 */ if (spec->multiout.dig_out_nid || spec->dig_in_nid) { codec->num_pcms = 2; + codec->slave_dig_outs = spec->multiout.slave_dig_outs; info = spec->pcm_rec + 1; info->name = spec->stream_name_digital; - info->pcm_type = HDA_PCM_TYPE_SPDIF; + if (spec->dig_out_type) + info->pcm_type = spec->dig_out_type; + else + info->pcm_type = HDA_PCM_TYPE_SPDIF; if (spec->multiout.dig_out_nid && spec->stream_digital_playback) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = *(spec->stream_digital_playback); @@ -3102,6 +3167,9 @@ static int alc_build_pcms(struct hda_codec *codec) codec->spdif_status_reset = 1; } + if (spec->no_analog) + return 0; + /* If the use of more than one ADC is requested for the current * model, configure a second analog capture-only PCM. */ @@ -3160,65 +3228,17 @@ static void alc_free(struct hda_codec *codec) alc_free_kctls(codec); kfree(spec); - codec->spec = NULL; /* to be sure */ + snd_hda_detach_beep_device(codec); } #ifdef SND_HDA_NEEDS_RESUME -static void store_pin_configs(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - hda_nid_t nid, end_nid; - - end_nid = codec->start_nid + codec->num_nodes; - for (nid = codec->start_nid; nid < end_nid; nid++) { - unsigned int wid_caps = get_wcaps(codec, nid); - unsigned int wid_type = - (wid_caps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wid_type != AC_WID_PIN) - continue; - if (spec->num_pins >= ARRAY_SIZE(spec->pin_nids)) - break; - spec->pin_nids[spec->num_pins] = nid; - spec->pin_cfgs[spec->num_pins] = - snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); - spec->num_pins++; - } -} - -static void resume_pin_configs(struct hda_codec *codec) -{ - struct alc_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_pins; i++) { - hda_nid_t pin_nid = spec->pin_nids[i]; - unsigned int pin_config = spec->pin_cfgs[i]; - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - pin_config & 0x000000ff); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (pin_config & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (pin_config & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - pin_config >> 24); - } -} - static int alc_resume(struct hda_codec *codec) { - resume_pin_configs(codec); codec->patch_ops.init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); return 0; } -#else -#define store_pin_configs(codec) #endif /* @@ -3557,7 +3577,7 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8181, "ASUS P4GPL", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x8196, "ASUS P5GD1", ALC880_6ST), SND_PCI_QUIRK(0x1043, 0x81b4, "ASUS", ALC880_6ST), - SND_PCI_QUIRK(0x1043, 0, "ASUS", ALC880_ASUS), /* default ASUS */ + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS", ALC880_ASUS), /* default ASUS */ SND_PCI_QUIRK(0x104d, 0x81a0, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x104d, 0x81d6, "Sony", ALC880_3ST), SND_PCI_QUIRK(0x107b, 0x3032, "Gateway", ALC880_5ST), @@ -3600,7 +3620,8 @@ static struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x8086, 0xe400, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xe401, "Intel mobo", ALC880_5ST_DIG), SND_PCI_QUIRK(0x8086, 0xe402, "Intel mobo", ALC880_5ST_DIG), - SND_PCI_QUIRK(0x8086, 0, "Intel mobo", ALC880_3ST), /* default Intel */ + /* default Intel */ + SND_PCI_QUIRK_VENDOR(0x8086, "Intel mobo", ALC880_3ST), SND_PCI_QUIRK(0xa0a0, 0x0560, "AOpen i915GMm-HFS", ALC880_5ST_DIG), SND_PCI_QUIRK(0xe803, 0x1019, NULL, ALC880_6ST_DIG), {} @@ -3780,7 +3801,7 @@ static struct alc_config_preset alc880_presets[] = { .input_mux = &alc880_capture_source, }, [ALC880_UNIWILL_DIG] = { - .mixers = { alc880_asus_mixer, alc880_pcbeep_mixer }, + .mixers = { alc880_asus_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_pin_asus_init_verbs }, .num_dacs = ARRAY_SIZE(alc880_asus_dac_nids), @@ -3818,8 +3839,7 @@ static struct alc_config_preset alc880_presets[] = { .init_hook = alc880_uniwill_p53_hp_automute, }, [ALC880_FUJITSU] = { - .mixers = { alc880_fujitsu_mixer, - alc880_pcbeep_mixer, }, + .mixers = { alc880_fujitsu_mixer }, .init_verbs = { alc880_volume_init_verbs, alc880_uniwill_p53_init_verbs, alc880_beep_init_verbs }, @@ -4112,7 +4132,7 @@ static int new_analog_input(struct alc_spec *spec, hda_nid_t pin, static int alc880_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -4200,11 +4220,9 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc880_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - if (nid != ALC880_PIN_CD_NID) + alc_set_input_pin(codec, nid, i); + if (nid != ALC880_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -4219,7 +4237,7 @@ static void alc880_auto_init_analog_input(struct hda_codec *codec) static int alc880_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int err; + int i, err; static hda_nid_t alc880_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -4250,8 +4268,23 @@ static int alc880_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) - spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; + /* check multiple SPDIF-out (for recent codecs) */ + for (i = 0; i < spec->autocfg.dig_outs; i++) { + hda_nid_t dig_nid; + err = snd_hda_get_connections(codec, + spec->autocfg.dig_out_pins[i], + &dig_nid, 1); + if (err < 0) + continue; + if (!i) + spec->multiout.dig_out_nid = dig_nid; + else { + spec->multiout.slave_dig_outs = spec->slave_dig_outs; + spec->slave_dig_outs[i - 1] = dig_nid; + if (i == ARRAY_SIZE(spec->slave_dig_outs) - 1) + break; + } + } if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC880_DIGIN_NID; @@ -4261,9 +4294,8 @@ static int alc880_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc880_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; - store_pin_configs(codec); return 1; } @@ -4278,21 +4310,33 @@ static void alc880_auto_init(struct hda_codec *codec) alc_inithook(codec); } -/* - * OK, here we have finally the patch for ALC880 - */ - static void set_capture_mixer(struct alc_spec *spec) { - static struct snd_kcontrol_new *caps[3] = { - alc_capture_mixer1, - alc_capture_mixer2, - alc_capture_mixer3, + static struct snd_kcontrol_new *caps[2][3] = { + { alc_capture_mixer_nosrc1, + alc_capture_mixer_nosrc2, + alc_capture_mixer_nosrc3 }, + { alc_capture_mixer1, + alc_capture_mixer2, + alc_capture_mixer3 }, }; - if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) - spec->cap_mixer = caps[spec->num_adc_nids - 1]; + if (spec->num_adc_nids > 0 && spec->num_adc_nids <= 3) { + int mux; + if (spec->input_mux && spec->input_mux->num_items > 1) + mux = 1; + else + mux = 0; + spec->cap_mixer = caps[mux][spec->num_adc_nids - 1]; + } } +#define set_beep_amp(spec, nid, idx, dir) \ + ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) + +/* + * OK, here we have finally the patch for ALC880 + */ + static int patch_alc880(struct hda_codec *codec) { struct alc_spec *spec; @@ -4328,6 +4372,12 @@ static int patch_alc880(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC880_AUTO) setup_preset(spec, &alc880_presets[board_config]); @@ -4354,6 +4404,7 @@ static int patch_alc880(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -4461,6 +4512,26 @@ static struct hda_input_mux alc260_acer_capture_sources[2] = { }, }, }; + +/* Maxdata Favorit 100XS */ +static struct hda_input_mux alc260_favorit100_capture_sources[2] = { + { + .num_items = 2, + .items = { + { "Line/Mic", 0x0 }, + { "CD", 0x4 }, + }, + }, + { + .num_items = 3, + .items = { + { "Line/Mic", 0x0 }, + { "CD", 0x4 }, + { "Mixer", 0x5 }, + }, + }, +}; + /* * This is just place-holder, so there's something for alc_build_pcms to look * at when it calculates the maximum number of channels. ALC260 has no mixer @@ -4503,12 +4574,6 @@ static struct snd_kcontrol_new alc260_input_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new alc260_pc_beep_mixer[] = { - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x07, 0x05, HDA_INPUT), - { } /* end */ -}; - /* update HP, line and mono out pins according to the master switch */ static void alc260_hp_master_update(struct hda_codec *codec, hda_nid_t hp, hda_nid_t line, @@ -4700,8 +4765,6 @@ static struct snd_kcontrol_new alc260_fujitsu_mixer[] = { HDA_CODEC_VOLUME("Mic/Line Playback Volume", 0x07, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic/Line Playback Switch", 0x07, 0x0, HDA_INPUT), ALC_PIN_MODE("Mic/Line Jack Mode", 0x12, ALC_PIN_DIR_IN), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Speaker Playback Volume", 0x09, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Speaker Playback Switch", 0x09, 2, HDA_INPUT), { } /* end */ @@ -4746,8 +4809,18 @@ static struct snd_kcontrol_new alc260_acer_mixer[] = { HDA_CODEC_VOLUME("Line Playback Volume", 0x07, 0x02, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x07, 0x02, HDA_INPUT), ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), + { } /* end */ +}; + +/* Maxdata Favorit 100XS: one output and one input (0x12) jack + */ +static struct snd_kcontrol_new alc260_favorit100_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x08, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x08, 2, HDA_INPUT), + ALC_PIN_MODE("Output Jack Mode", 0x0f, ALC_PIN_DIR_INOUT), + HDA_CODEC_VOLUME("Line/Mic Playback Volume", 0x07, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Line/Mic Playback Switch", 0x07, 0x0, HDA_INPUT), + ALC_PIN_MODE("Line/Mic Jack Mode", 0x12, ALC_PIN_DIR_IN), { } /* end */ }; @@ -4765,8 +4838,6 @@ static struct snd_kcontrol_new alc260_will_mixer[] = { ALC_PIN_MODE("Line Jack Mode", 0x14, ALC_PIN_DIR_INOUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), { } /* end */ }; @@ -5124,6 +5195,89 @@ static struct hda_verb alc260_acer_init_verbs[] = { { } }; +/* Initialisation sequence for Maxdata Favorit 100XS + * (adapted from Acer init verbs). + */ +static struct hda_verb alc260_favorit100_init_verbs[] = { + /* GPIO 0 enables the output jack. + * Turn this on and rely on the standard mute + * methods whenever the user wants to turn these outputs off. + */ + {0x01, AC_VERB_SET_GPIO_MASK, 0x01}, + {0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01}, + {0x01, AC_VERB_SET_GPIO_DATA, 0x01}, + /* Line/Mic input jack is connected to Mic1 pin */ + {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF50}, + /* Ensure all other unused pins are disabled and muted. */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x13, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x13, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Disable digital (SPDIF) pins */ + {0x03, AC_VERB_SET_DIGI_CONVERT_1, 0}, + {0x06, AC_VERB_SET_DIGI_CONVERT_1, 0}, + + /* Ensure Mic1 and Line1 pin widgets take input from the OUT1 sum + * bus when acting as outputs. + */ + {0x0b, AC_VERB_SET_CONNECT_SEL, 0}, + {0x0d, AC_VERB_SET_CONNECT_SEL, 0}, + + /* Start with output sum widgets muted and their output gains at min */ + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x08, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + + /* Unmute Line-out pin widget amp left and right + * (no equiv mixer ctrl) + */ + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + /* Unmute Mic1 and Line1 pin widget input buffers since they start as + * inputs. If the pin mode is changed by the user the pin mode control + * will take care of enabling the pin's input/output buffers as needed. + * Therefore there's no need to enable the input buffer at this + * stage. + */ + {0x12, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + + /* Mute capture amp left and right */ + {0x04, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + /* Set ADC connection select to match default mixer setting - mic + * (on mic1 pin) + */ + {0x04, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Do similar with the second ADC: mute capture input amp and + * set ADC connection to mic to match ALSA's default state. + */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x05, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* Mute all inputs to mixer widget (even unconnected ones) */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* mic1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* mic2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, /* line1 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, /* line2 pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(4)}, /* CD pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(5)}, /* Beep-gen pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(6)}, /* Line-out pin */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(7)}, /* HP-pin pin */ + + { } +}; + static struct hda_verb alc260_will_verbs[] = { {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x0b, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -5270,8 +5424,6 @@ static struct snd_kcontrol_new alc260_test_mixer[] = { HDA_CODEC_MUTE("LINE2 Playback Switch", 0x07, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x07, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x07, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x07, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x07, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("LINE-OUT loopback Playback Volume", 0x07, 0x06, HDA_INPUT), HDA_CODEC_MUTE("LINE-OUT loopback Playback Switch", 0x07, 0x06, HDA_INPUT), HDA_CODEC_VOLUME("HP-OUT loopback Playback Volume", 0x07, 0x7, HDA_INPUT), @@ -5469,7 +5621,7 @@ static int alc260_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc260_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -5544,11 +5696,9 @@ static void alc260_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (nid >= 0x12) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - if (nid != ALC260_PIN_CD_NID) + alc_set_input_pin(codec, nid, i); + if (nid != ALC260_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -5621,7 +5771,7 @@ static int alc260_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC260_DIGOUT_NID; if (spec->kctls.list) add_mixer(spec, spec->kctls.list); @@ -5629,9 +5779,8 @@ static int alc260_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc260_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; - store_pin_configs(codec); return 1; } @@ -5668,6 +5817,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = { [ALC260_ACER] = "acer", [ALC260_WILL] = "will", [ALC260_REPLACER_672V] = "replacer", + [ALC260_FAVORIT100] = "favorit100", #ifdef CONFIG_SND_DEBUG [ALC260_TEST] = "test", #endif @@ -5677,6 +5827,7 @@ static const char *alc260_models[ALC260_MODEL_LAST] = { static struct snd_pci_quirk alc260_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x007b, "Acer C20x", ALC260_ACER), SND_PCI_QUIRK(0x1025, 0x008f, "Acer", ALC260_ACER), + SND_PCI_QUIRK(0x1509, 0x4540, "Favorit 100XS", ALC260_FAVORIT100), SND_PCI_QUIRK(0x103c, 0x2808, "HP d5700", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x280a, "HP d5750", ALC260_HP_3013), SND_PCI_QUIRK(0x103c, 0x3010, "HP", ALC260_HP_3013), @@ -5699,8 +5850,7 @@ static struct snd_pci_quirk alc260_cfg_tbl[] = { static struct alc_config_preset alc260_presets[] = { [ALC260_BASIC] = { .mixers = { alc260_base_output_mixer, - alc260_input_mixer, - alc260_pc_beep_mixer }, + alc260_input_mixer }, .init_verbs = { alc260_init_verbs }, .num_dacs = ARRAY_SIZE(alc260_dac_nids), .dac_nids = alc260_dac_nids, @@ -5779,6 +5929,18 @@ static struct alc_config_preset alc260_presets[] = { .num_mux_defs = ARRAY_SIZE(alc260_acer_capture_sources), .input_mux = alc260_acer_capture_sources, }, + [ALC260_FAVORIT100] = { + .mixers = { alc260_favorit100_mixer }, + .init_verbs = { alc260_favorit100_init_verbs }, + .num_dacs = ARRAY_SIZE(alc260_dac_nids), + .dac_nids = alc260_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc260_dual_adc_nids), + .adc_nids = alc260_dual_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc260_modes), + .channel_mode = alc260_modes, + .num_mux_defs = ARRAY_SIZE(alc260_favorit100_capture_sources), + .input_mux = alc260_favorit100_capture_sources, + }, [ALC260_WILL] = { .mixers = { alc260_will_mixer }, .init_verbs = { alc260_init_verbs, alc260_will_verbs }, @@ -5855,6 +6017,12 @@ static int patch_alc260(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC260_AUTO) setup_preset(spec, &alc260_presets[board_config]); @@ -5880,6 +6048,7 @@ static int patch_alc260(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x07, 0x05, HDA_INPUT); spec->vmaster_nid = 0x08; @@ -6051,8 +6220,6 @@ static struct snd_kcontrol_new alc882_base_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6079,8 +6246,6 @@ static struct snd_kcontrol_new alc882_w2jc_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6132,8 +6297,6 @@ static struct snd_kcontrol_new alc882_asus_a7m_mixer[] = { HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -6242,8 +6405,10 @@ static struct snd_kcontrol_new alc882_macpro_mixer[] = { HDA_CODEC_MUTE("Headphone Playback Switch", 0x18, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), + /* FIXME: this looks suspicious... HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x02, HDA_INPUT), + */ { } /* end */ }; @@ -6875,19 +7040,9 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - unsigned int vref; if (!nid) continue; - vref = PIN_IN; - if (1 /*i <= AUTO_PIN_FRONT_MIC*/) { - unsigned int pincap; - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); - if ((pincap >> AC_PINCAP_VREF_SHIFT) & - AC_PINCAP_VREF_80) - vref = PIN_VREF80; - } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, vref); + alc_set_input_pin(codec, nid, AUTO_PIN_FRONT_MIC /*i*/); if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, @@ -6898,18 +7053,21 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) static void alc882_auto_init_input_src(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; int c; for (c = 0; c < spec->num_adc_nids; c++) { hda_nid_t conn_list[HDA_MAX_NUM_INPUTS]; hda_nid_t nid = spec->capsrc_nids[c]; + unsigned int mux_idx; + const struct hda_input_mux *imux; int conns, mute, idx, item; conns = snd_hda_get_connections(codec, nid, conn_list, ARRAY_SIZE(conn_list)); if (conns < 0) continue; + mux_idx = c >= spec->num_mux_defs ? 0 : c; + imux = &spec->input_mux[mux_idx]; for (idx = 0; idx < conns; idx++) { /* if the current connection is the selected one, * unmute it as default - otherwise mute it @@ -6922,8 +7080,20 @@ static void alc882_auto_init_input_src(struct hda_codec *codec) break; } } - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_AMP_GAIN_MUTE, mute); + /* check if we have a selector or mixer + * we could check for the widget type instead, but + * just check for Amp-In presence (in case of mixer + * without amp-in there is something wrong, this + * function shouldn't be used or capsrc nid is wrong) + */ + if (get_wcaps(codec, nid) & AC_WCAP_IN_AMP) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + mute); + else if (mute != AMP_IN_MUTE(idx)) + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, + idx); } } } @@ -7012,12 +7182,15 @@ static int patch_alc882(struct hda_codec *codec) break; case 0x106b1000: /* iMac 24 */ case 0x106b2800: /* AppleTV */ + case 0x106b3e00: /* iMac 24 Aluminium */ board_config = ALC885_IMAC24; break; + case 0x106b00a0: /* MacBookPro3,1 - Another revision */ case 0x106b00a1: /* Macbook (might be wrong - PCI SSID?) */ case 0x106b00a4: /* MacbookPro4,1 */ case 0x106b2c00: /* Macbook Pro rev3 */ case 0x106b3600: /* Macbook 3.1 */ + case 0x106b3800: /* MacbookPro4,1 - latter revision */ board_config = ALC885_MBP3; break; default: @@ -7049,6 +7222,12 @@ static int patch_alc882(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); @@ -7069,7 +7248,7 @@ static int patch_alc882(struct hda_codec *codec) spec->stream_digital_playback = &alc882_pcm_digital_playback; spec->stream_digital_capture = &alc882_pcm_digital_capture; - spec->is_mix_capture = 1; /* matrix-style capture */ + spec->capture_style = CAPT_MIX; /* matrix-style capture */ if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -7086,6 +7265,7 @@ static int patch_alc882(struct hda_codec *codec) } } set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -7137,10 +7317,14 @@ static hda_nid_t alc883_adc_nids_rev[2] = { 0x09, 0x08 }; +#define alc889_adc_nids alc880_adc_nids + static hda_nid_t alc883_capsrc_nids[2] = { 0x23, 0x22 }; static hda_nid_t alc883_capsrc_nids_rev[2] = { 0x22, 0x23 }; +#define alc889_capsrc_nids alc882_capsrc_nids + /* input MUX */ /* FIXME: should be a matrix-type input source selection */ @@ -7358,8 +7542,6 @@ static struct snd_kcontrol_new alc883_base_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7422,8 +7604,6 @@ static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7447,8 +7627,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7473,8 +7651,6 @@ static struct snd_kcontrol_new alc883_3ST_6ch_intel_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7498,8 +7674,6 @@ static struct snd_kcontrol_new alc883_fivestack_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -7907,36 +8081,83 @@ static struct hda_verb alc888_lenovo_sky_verbs[] = { { } /* end */ }; +static struct hda_verb alc888_6st_dell_verbs[] = { + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + { } +}; + +static void alc888_3st_hp_front_automute(struct hda_codec *codec) +{ + unsigned int present, bits; + + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc888_3st_hp_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc888_3st_hp_front_automute(codec); + break; + } +} + static struct hda_verb alc888_3st_hp_verbs[] = { {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, /* Front: output 0 (0x0c) */ {0x16, AC_VERB_SET_CONNECT_SEL, 0x01}, /* Rear : output 1 (0x0d) */ {0x18, AC_VERB_SET_CONNECT_SEL, 0x02}, /* CLFE : output 2 (0x0e) */ - { } -}; - -static struct hda_verb alc888_6st_dell_verbs[] = { {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, - { } + { } /* end */ }; +/* + * 2ch mode + */ static struct hda_verb alc888_3st_hp_2ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, - { } + { } /* end */ }; +/* + * 4ch mode + */ +static struct hda_verb alc888_3st_hp_4ch_init[] = { + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ static struct hda_verb alc888_3st_hp_6ch_init[] = { { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, { 0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, { 0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, - { } + { 0x16, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ }; -static struct hda_channel_mode alc888_3st_hp_modes[2] = { +static struct hda_channel_mode alc888_3st_hp_modes[3] = { { 2, alc888_3st_hp_2ch_init }, + { 4, alc888_3st_hp_4ch_init }, { 6, alc888_3st_hp_6ch_init }, }; @@ -8197,7 +8418,7 @@ static void alc888_6st_dell_unsol_event(struct hda_codec *codec, { switch (res >> 26) { case ALC880_HP_EVENT: - printk("hp_event\n"); + /* printk(KERN_DEBUG "hp_event\n"); */ alc888_6st_dell_front_automute(codec); break; } @@ -8456,6 +8677,7 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1019, 0x6668, "ECS", ALC883_3ST_6ch_DIG), SND_PCI_QUIRK(0x1025, 0x006c, "Acer Aspire 9810", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0090, "Acer Aspire", ALC883_ACER_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x010a, "Acer Ferrari 5000", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0110, "Acer Aspire", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0112, "Acer Aspire 9303", ALC883_ACER_ASPIRE), SND_PCI_QUIRK(0x1025, 0x0121, "Acer Aspire 5920G", ALC883_ACER_ASPIRE), @@ -8463,21 +8685,29 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { ALC888_ACER_ASPIRE_4930G), SND_PCI_QUIRK(0x1025, 0x013f, "Acer Aspire 5930G", ALC888_ACER_ASPIRE_4930G), + SND_PCI_QUIRK(0x1025, 0x0157, "Acer X3200", ALC883_AUTO), + SND_PCI_QUIRK(0x1025, 0x0158, "Acer AX1700-U3700A", ALC883_AUTO), SND_PCI_QUIRK(0x1025, 0x015e, "Acer Aspire 6930G", ALC888_ACER_ASPIRE_4930G), - SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), /* default Acer */ + SND_PCI_QUIRK(0x1025, 0x0166, "Acer Aspire 6530G", + ALC888_ACER_ASPIRE_4930G), + /* default Acer */ + SND_PCI_QUIRK_VENDOR(0x1025, "Acer laptop", ALC883_ACER), SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a66, "HP Acacia", ALC888_3ST_HP), + SND_PCI_QUIRK(0x103c, 0x2a72, "HP Educ.ar", ALC888_3ST_HP), SND_PCI_QUIRK(0x1043, 0x1873, "Asus M90V", ALC888_ASUS_M90V), SND_PCI_QUIRK(0x1043, 0x8249, "Asus M2A-VM HDMI", ALC883_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1043, 0x8284, "Asus Z37E", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x82fe, "Asus P5Q-EM HDMI", ALC1200_ASUS_P5Q), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_ASUS_EEE1601), SND_PCI_QUIRK(0x105b, 0x0ce8, "Foxconn P35AX-S", ALC883_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x1071, 0x8227, "Mitac 82801H", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), @@ -8509,9 +8739,11 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), - SND_PCI_QUIRK(0x1558, 0, "Clevo laptop", ALC883_LAPTOP_EAPD), + SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1734, 0x1107, "FSC AMILO Xi2550", + ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x1108, "Fujitsu AMILO Pi2515", ALC883_FUJITSU_PI2515), SND_PCI_QUIRK(0x1734, 0x113d, "Fujitsu AMILO Xa3530", ALC888_FUJITSU_XA3530), @@ -8526,11 +8758,20 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = { SND_PCI_QUIRK(0x1991, 0x5625, "Haier W66", ALC883_HAIER_W66), SND_PCI_QUIRK(0x8086, 0x0001, "DG33BUC", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0x0002, "DG33FBC", ALC883_3ST_6ch_INTEL), + SND_PCI_QUIRK(0x8086, 0x2503, "82801H", ALC883_MITAC), SND_PCI_QUIRK(0x8086, 0x0022, "DX58SO", ALC883_3ST_6ch_INTEL), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), {} }; +static hda_nid_t alc883_slave_dig_outs[] = { + ALC1200_DIGOUT_NID, 0, +}; + +static hda_nid_t alc1200_slave_dig_outs[] = { + ALC883_DIGOUT_NID, 0, +}; + static struct alc_config_preset alc883_presets[] = { [ALC883_3ST_2ch_DIG] = { .mixers = { alc883_3ST_2ch_mixer }, @@ -8572,6 +8813,7 @@ static struct alc_config_preset alc883_presets[] = { .dac_nids = alc883_dac_nids, .dig_out_nid = ALC883_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc883_slave_dig_outs, .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes), .channel_mode = alc883_3ST_6ch_intel_modes, .need_dac_fix = 1, @@ -8766,6 +9008,8 @@ static struct alc_config_preset alc883_presets[] = { .channel_mode = alc888_3st_hp_modes, .need_dac_fix = 1, .input_mux = &alc883_capture_source, + .unsol_event = alc888_3st_hp_unsol_event, + .init_hook = alc888_3st_hp_front_automute, }, [ALC888_6ST_DELL] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer }, @@ -8871,6 +9115,7 @@ static struct alc_config_preset alc883_presets[] = { .dac_nids = alc883_dac_nids, .dig_out_nid = ALC1200_DIGOUT_NID, .dig_in_nid = ALC883_DIGIN_NID, + .slave_dig_outs = alc1200_slave_dig_outs, .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), .channel_mode = alc883_sixstack_modes, .input_mux = &alc883_capture_source, @@ -8938,11 +9183,9 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc883_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - (i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN)); - if (nid != ALC883_PIN_CD_NID) + alc_set_input_pin(codec, nid, i); + if (nid != ALC883_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -8957,6 +9200,8 @@ static int alc883_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; int err = alc880_parse_auto_config(codec); + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; if (err < 0) return err; @@ -8970,6 +9215,26 @@ static int alc883_parse_auto_config(struct hda_codec *codec) /* hack - override the init verbs */ spec->init_verbs[0] = alc883_auto_init_verbs; + /* setup input_mux for ALC889 */ + if (codec->vendor_id == 0x10ec0889) { + /* digital-mic input pin is excluded in alc880_auto_create..() + * because it's under 0x18 + */ + if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || + cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { + struct hda_input_mux *imux = &spec->private_imux[0]; + for (i = 1; i < 3; i++) + memcpy(&spec->private_imux[i], + &spec->private_imux[0], + sizeof(spec->private_imux[0])); + imux->items[imux->num_items].label = "Int DMic"; + imux->items[imux->num_items].index = 0x0b; + imux->num_items++; + spec->num_mux_defs = 3; + spec->input_mux = spec->private_imux; + } + } + return 1; /* config found */ } @@ -9021,6 +9286,12 @@ static int patch_alc883(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC883_AUTO) setup_preset(spec, &alc883_presets[board_config]); @@ -9033,14 +9304,36 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_name_analog = "ALC888 Analog"; spec->stream_name_digital = "ALC888 Digital"; } + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; case 0x10ec0889: spec->stream_name_analog = "ALC889 Analog"; spec->stream_name_digital = "ALC889 Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc889_adc_nids); + spec->adc_nids = alc889_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc889_capsrc_nids; + spec->capture_style = CAPT_1MUX_MIX; /* 1mux/Nmix-style + capture */ break; default: spec->stream_name_analog = "ALC883 Analog"; spec->stream_name_digital = "ALC883 Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ break; } @@ -9051,15 +9344,9 @@ static int patch_alc883(struct hda_codec *codec) spec->stream_digital_playback = &alc883_pcm_digital_playback; spec->stream_digital_capture = &alc883_pcm_digital_capture; - if (!spec->num_adc_nids) { - spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); - spec->adc_nids = alc883_adc_nids; - } - if (!spec->capsrc_nids) - spec->capsrc_nids = alc883_capsrc_nids; - spec->is_mix_capture = 1; /* matrix-style capture */ if (!spec->cap_mixer) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -9112,8 +9399,6 @@ static struct snd_kcontrol_new alc262_base_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */ HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_MONO("Mono Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), @@ -9134,8 +9419,6 @@ static struct snd_kcontrol_new alc262_hippo1_mixer[] = { HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), - /* HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), */ /*HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0D, 0x0, HDA_OUTPUT),*/ HDA_CODEC_MUTE("Headphone Playback Switch", 0x1b, 0x0, HDA_OUTPUT), { } /* end */ @@ -9244,8 +9527,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("AUX IN Playback Volume", 0x0b, 0x06, HDA_INPUT), HDA_CODEC_MUTE("AUX IN Playback Switch", 0x0b, 0x06, HDA_INPUT), { } /* end */ @@ -9274,8 +9555,6 @@ static struct snd_kcontrol_new alc262_HP_BPC_WildWest_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -9423,6 +9702,67 @@ static struct snd_kcontrol_new alc262_benq_t31_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new alc262_tyan_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Aux Playback Volume", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_MUTE("Aux Playback Switch", 0x0b, 0x06, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + { } /* end */ +}; + +static struct hda_verb alc262_tyan_verbs[] = { + /* Headphone automute */ + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT}, + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + + /* P11 AUX_IN, white 4-pin connector */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, 0xe1}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, 0x93}, + {0x14, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, 0x19}, + + {} +}; + +/* unsolicited event for HP jack sensing */ +static void alc262_tyan_automute(struct hda_codec *codec) +{ + unsigned int mute; + unsigned int present; + + snd_hda_codec_read(codec, 0x1b, 0, AC_VERB_SET_PIN_SENSE, 0); + present = snd_hda_codec_read(codec, 0x1b, 0, + AC_VERB_GET_PIN_SENSE, 0); + present = (present & 0x80000000) != 0; + if (present) { + /* mute line output on ATX panel */ + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, HDA_AMP_MUTE); + } else { + /* unmute line output if necessary */ + mute = snd_hda_codec_amp_read(codec, 0x1b, 0, HDA_OUTPUT, 0); + snd_hda_codec_amp_stereo(codec, 0x15, HDA_OUTPUT, 0, + HDA_AMP_MUTE, mute); + } +} + +static void alc262_tyan_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + if ((res >> 26) != ALC880_HP_EVENT) + return; + alc262_tyan_automute(codec); +} + #define alc262_capture_mixer alc882_capture_mixer #define alc262_capture_alt_mixer alc882_capture_alt_mixer @@ -9889,8 +10229,6 @@ static struct snd_kcontrol_new alc262_fujitsu_mixer[] = { }, HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Switch", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), @@ -10462,8 +10800,14 @@ static int alc262_parse_auto_config(struct hda_codec *codec) alc262_ignore); if (err < 0) return err; - if (!spec->autocfg.line_outs) + if (!spec->autocfg.line_outs) { + if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { + spec->multiout.max_channels = 2; + spec->no_analog = 1; + goto dig_only; + } return 0; /* can't find valid BIOS pin config */ + } err = alc262_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; @@ -10473,8 +10817,11 @@ static int alc262_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + dig_only: + if (spec->autocfg.dig_outs) { spec->multiout.dig_out_nid = ALC262_DIGOUT_NID; + spec->dig_out_type = spec->autocfg.dig_out_type[0]; + } if (spec->autocfg.dig_in_pin) spec->dig_in_nid = ALC262_DIGIN_NID; @@ -10483,13 +10830,12 @@ static int alc262_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc262_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -10531,20 +10877,19 @@ static const char *alc262_models[ALC262_MODEL_LAST] = { [ALC262_ULTRA] = "ultra", [ALC262_LENOVO_3000] = "lenovo-3000", [ALC262_NEC] = "nec", + [ALC262_TYAN] = "tyan", [ALC262_AUTO] = "auto", }; static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x1002, 0x437b, "Hippo", ALC262_HIPPO), SND_PCI_QUIRK(0x1033, 0x8895, "NEC Versa S9100", ALC262_NEC), - SND_PCI_QUIRK(0x103c, 0x12fe, "HP xw9400", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x12ff, "HP xw4550", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1306, "HP xw8600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1307, "HP xw6600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1308, "HP xw4600", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x1309, "HP xw4*00", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x130a, "HP xw6*00", ALC262_HP_BPC), - SND_PCI_QUIRK(0x103c, 0x130b, "HP xw8*00", ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1200, "HP xw series", + ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1300, "HP xw series", + ALC262_HP_BPC), + SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series", + ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), SND_PCI_QUIRK(0x103c, 0x2801, "HP D7000", ALC262_HP_BPC_D7000_WF), SND_PCI_QUIRK(0x103c, 0x2802, "HP D7000", ALC262_HP_BPC_D7000_WL), @@ -10562,17 +10907,18 @@ static struct snd_pci_quirk alc262_cfg_tbl[] = { SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x8203, "Sony UX-90", ALC262_HIPPO), SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x9015, "Sony 0x9015", ALC262_SONY_ASSAMD), - SND_PCI_QUIRK(0x104d, 0x9033, "Sony VAIO VGN-SR19XN", - ALC262_SONY_ASSAMD), + SND_PCI_QUIRK(0x104d, 0x9016, "Sony VAIO", ALC262_AUTO), /* dig-only */ + SND_PCI_QUIRK_MASK(0x104d, 0xff00, 0x9000, "Sony VAIO", + ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x1179, 0x0001, "Toshiba dynabook SS RX1", ALC262_TOSHIBA_RX1), SND_PCI_QUIRK(0x1179, 0xff7b, "Toshiba S06", ALC262_TOSHIBA_S06), SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), SND_PCI_QUIRK(0x10cf, 0x142d, "Fujitsu Lifebook E8410", ALC262_FUJITSU), - SND_PCI_QUIRK(0x144d, 0xc032, "Samsung Q1 Ultra", ALC262_ULTRA), - SND_PCI_QUIRK(0x144d, 0xc039, "Samsung Q1U EL", ALC262_ULTRA), + SND_PCI_QUIRK(0x10f1, 0x2915, "Tyan Thunder n6650W", ALC262_TYAN), + SND_PCI_QUIRK_MASK(0x144d, 0xff00, 0xc032, "Samsung Q1", + ALC262_ULTRA), + SND_PCI_QUIRK(0x144d, 0xc510, "Samsung Q45", ALC262_HIPPO), SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 y410", ALC262_LENOVO_3000), SND_PCI_QUIRK(0x17ff, 0x0560, "Benq ED8", ALC262_BENQ_ED8), SND_PCI_QUIRK(0x17ff, 0x058d, "Benq T31-16", ALC262_BENQ_T31), @@ -10788,6 +11134,19 @@ static struct alc_config_preset alc262_presets[] = { .unsol_event = alc262_hippo_unsol_event, .init_hook = alc262_hippo_automute, }, + [ALC262_TYAN] = { + .mixers = { alc262_tyan_mixer }, + .init_verbs = { alc262_init_verbs, alc262_tyan_verbs}, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x02, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_capture_source, + .unsol_event = alc262_tyan_unsol_event, + .init_hook = alc262_tyan_automute, + }, }; static int patch_alc262(struct hda_codec *codec) @@ -10840,6 +11199,14 @@ static int patch_alc262(struct hda_codec *codec) } } + if (!spec->no_analog) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + } + if (board_config != ALC262_AUTO) setup_preset(spec, &alc262_presets[board_config]); @@ -10851,7 +11218,7 @@ static int patch_alc262(struct hda_codec *codec) spec->stream_digital_playback = &alc262_pcm_digital_playback; spec->stream_digital_capture = &alc262_pcm_digital_capture; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); @@ -10868,8 +11235,10 @@ static int patch_alc262(struct hda_codec *codec) spec->capsrc_nids = alc262_capsrc_nids; } } - if (!spec->cap_mixer) + if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); + if (!spec->no_analog) + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x0c; @@ -11249,19 +11618,13 @@ static void alc267_quanta_il1_unsol_event(struct hda_codec *codec, static struct hda_verb alc268_base_init_verbs[] = { /* Unmute DAC0-1 and set vol = 0 */ {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* * Set up output mixers (0x0c - 0x0e) */ /* set vol=0 to output mixers */ {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0e, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -11280,9 +11643,7 @@ static struct hda_verb alc268_base_init_verbs[] = { {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* set PCBEEP vol = 0, mute connections */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -11304,10 +11665,8 @@ static struct hda_verb alc268_base_init_verbs[] = { */ static struct hda_verb alc268_volume_init_verbs[] = { /* set output DAC */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, + {0x03, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, @@ -11315,16 +11674,12 @@ static struct hda_verb alc268_volume_init_verbs[] = { {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, {0x1d, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x20}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* set PCBEEP vol = 0, mute connections */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -11523,7 +11878,7 @@ static int alc268_auto_create_multi_out_ctls(struct alc_spec *spec, static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -11617,9 +11972,14 @@ static int alc268_parse_auto_config(struct hda_codec *codec) alc268_ignore); if (err < 0) return err; - if (!spec->autocfg.line_outs) + if (!spec->autocfg.line_outs) { + if (spec->autocfg.dig_outs || spec->autocfg.dig_in_pin) { + spec->multiout.max_channels = 2; + spec->no_analog = 1; + goto dig_only; + } return 0; /* can't find valid BIOS pin config */ - + } err = alc268_auto_create_multi_out_ctls(spec, &spec->autocfg); if (err < 0) return err; @@ -11629,25 +11989,26 @@ static int alc268_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = 2; + dig_only: /* digital only support output */ - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) { spec->multiout.dig_out_nid = ALC268_DIGOUT_NID; - + spec->dig_out_type = spec->autocfg.dig_out_type[0]; + } if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - if (spec->autocfg.speaker_pins[0] != 0x1d) + if (!spec->no_analog && spec->autocfg.speaker_pins[0] != 0x1d) add_mixer(spec, alc268_beep_mixer); add_verb(spec, alc268_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -11709,7 +12070,7 @@ static struct snd_pci_quirk alc268_cfg_tbl[] = { static struct alc_config_preset alc268_presets[] = { [ALC267_QUANTA_IL1] = { - .mixers = { alc267_quanta_il1_mixer }, + .mixers = { alc267_quanta_il1_mixer, alc268_beep_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc267_quanta_il1_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -11791,7 +12152,8 @@ static struct alc_config_preset alc268_presets[] = { }, [ALC268_ACER_ASPIRE_ONE] = { .mixers = { alc268_acer_aspire_one_mixer, - alc268_capture_alt_mixer }, + alc268_beep_mixer, + alc268_capture_alt_mixer }, .init_verbs = { alc268_base_init_verbs, alc268_eapd_verbs, alc268_acer_aspire_one_verbs }, .num_dacs = ARRAY_SIZE(alc268_dac_nids), @@ -11860,7 +12222,7 @@ static int patch_alc268(struct hda_codec *codec) { struct alc_spec *spec; int board_config; - int err; + int i, has_beep, err; spec = kcalloc(1, sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -11909,15 +12271,30 @@ static int patch_alc268(struct hda_codec *codec) spec->stream_digital_playback = &alc268_pcm_digital_playback; - if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) - /* override the amp caps for beep generator */ - snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, + has_beep = 0; + for (i = 0; i < spec->num_mixers; i++) { + if (spec->mixers[i] == alc268_beep_mixer) { + has_beep = 1; + break; + } + } + + if (has_beep) { + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (!query_amp_caps(codec, 0x1d, HDA_INPUT)) + /* override the amp caps for beep generator */ + snd_hda_override_amp_caps(codec, 0x1d, HDA_INPUT, (0x0c << AC_AMPCAP_OFFSET_SHIFT) | (0x0c << AC_AMPCAP_NUM_STEPS_SHIFT) | (0x07 << AC_AMPCAP_STEP_SIZE_SHIFT) | (0 << AC_AMPCAP_MUTE_SHIFT)); + } - if (!spec->adc_nids && spec->input_mux) { + if (!spec->no_analog && !spec->adc_nids && spec->input_mux) { /* check whether NID 0x07 is valid */ unsigned int wcap = get_wcaps(codec, 0x07); int i; @@ -11998,8 +12375,6 @@ static struct snd_kcontrol_new alc269_base_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT), HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), @@ -12026,8 +12401,6 @@ static struct snd_kcontrol_new alc269_quanta_fl1_mixer[] = { HDA_CODEC_VOLUME("Internal Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_MUTE("Internal Mic Playback Switch", 0x0b, 0x01, HDA_INPUT), HDA_CODEC_VOLUME("Internal Mic Boost", 0x19, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), { } }; @@ -12051,8 +12424,6 @@ static struct snd_kcontrol_new alc269_lifebook_mixer[] = { HDA_CODEC_VOLUME("Dock Mic Playback Volume", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_MUTE("Dock Mic Playback Switch", 0x0b, 0x03, HDA_INPUT), HDA_CODEC_VOLUME("Dock Mic Boost", 0x1b, 0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x04, HDA_INPUT), { } }; @@ -12089,13 +12460,6 @@ static struct snd_kcontrol_new alc269_fujitsu_mixer[] = { { } /* end */ }; -/* beep control */ -static struct snd_kcontrol_new alc269_beep_mixer[] = { - HDA_CODEC_VOLUME("Beep Playback Volume", 0x0b, 0x4, HDA_INPUT), - HDA_CODEC_MUTE("Beep Playback Switch", 0x0b, 0x4, HDA_INPUT), - { } /* end */ -}; - static struct hda_verb alc269_quanta_fl1_verbs[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x01}, {0x12, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -12495,7 +12859,7 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, */ if (cfg->input_pins[AUTO_PIN_MIC] == 0x12 || cfg->input_pins[AUTO_PIN_FRONT_MIC] == 0x12) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; imux->items[imux->num_items].label = "Int Mic"; imux->items[imux->num_items].index = 0x05; imux->num_items++; @@ -12513,13 +12877,34 @@ static int alc269_auto_create_analog_input_ctls(struct alc_spec *spec, #define alc269_pcm_digital_playback alc880_pcm_digital_playback #define alc269_pcm_digital_capture alc880_pcm_digital_capture +static struct hda_pcm_stream alc269_44k_pcm_analog_playback = { + .substreams = 1, + .channels_min = 2, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ + /* NID is set in alc_build_pcms */ + .ops = { + .open = alc880_playback_pcm_open, + .prepare = alc880_playback_pcm_prepare, + .cleanup = alc880_playback_pcm_cleanup + }, +}; + +static struct hda_pcm_stream alc269_44k_pcm_analog_capture = { + .substreams = 1, + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_44100, /* fixed rate */ + /* NID is set in alc_build_pcms */ +}; + /* * BIOS auto configuration */ static int alc269_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; - int i, err; + int err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, @@ -12536,22 +12921,15 @@ static int alc269_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC269_DIGOUT_NID; if (spec->kctls.list) add_mixer(spec, spec->kctls.list); - /* create a beep mixer control if the pin 0x1d isn't assigned */ - for (i = 0; i < ARRAY_SIZE(spec->autocfg.input_pins); i++) - if (spec->autocfg.input_pins[i] == 0x1d) - break; - if (i >= ARRAY_SIZE(spec->autocfg.input_pins)) - add_mixer(spec, alc269_beep_mixer); - add_verb(spec, alc269_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; /* set default input source */ snd_hda_codec_write_cache(codec, alc269_capsrc_nids[0], 0, AC_VERB_SET_CONNECT_SEL, @@ -12561,10 +12939,9 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - if (!spec->cap_mixer) + if (!spec->cap_mixer && !spec->no_analog) set_capture_mixer(spec); - store_pin_configs(codec); return 1; } @@ -12661,7 +13038,7 @@ static struct alc_config_preset alc269_presets[] = { .init_hook = alc269_eeepc_dmic_inithook, }, [ALC269_FUJITSU] = { - .mixers = { alc269_fujitsu_mixer, alc269_beep_mixer }, + .mixers = { alc269_fujitsu_mixer }, .cap_mixer = alc269_epc_capture_mixer, .init_verbs = { alc269_init_verbs, alc269_eeepc_dmic_init_verbs }, @@ -12726,13 +13103,26 @@ static int patch_alc269(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC269_AUTO) setup_preset(spec, &alc269_presets[board_config]); spec->stream_name_analog = "ALC269 Analog"; - spec->stream_analog_playback = &alc269_pcm_analog_playback; - spec->stream_analog_capture = &alc269_pcm_analog_capture; - + if (codec->subsystem_id == 0x17aa3bf8) { + /* Due to a hardware problem on Lenovo Ideadpad, we need to + * fix the sample rate of analog I/O to 44.1kHz + */ + spec->stream_analog_playback = &alc269_44k_pcm_analog_playback; + spec->stream_analog_capture = &alc269_44k_pcm_analog_capture; + } else { + spec->stream_analog_playback = &alc269_pcm_analog_playback; + spec->stream_analog_capture = &alc269_pcm_analog_capture; + } spec->stream_name_digital = "ALC269 Digital"; spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; @@ -12742,6 +13132,7 @@ static int patch_alc269(struct hda_codec *codec) spec->capsrc_nids = alc269_capsrc_nids; if (!spec->cap_mixer) set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); codec->patch_ops = alc_patch_ops; if (board_config == ALC269_AUTO) @@ -12992,8 +13383,6 @@ static struct snd_kcontrol_new alc861_asus_mixer[] = { static struct snd_kcontrol_new alc861_asus_laptop_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x15, 0x0, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x15, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Playback Volume", 0x23, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE("PC Beep Playback Switch", 0x23, 0x0, HDA_OUTPUT), { } }; @@ -13467,7 +13856,7 @@ static int alc861_auto_create_hp_ctls(struct alc_spec *spec, hda_nid_t pin) static int alc861_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx, idx1; for (i = 0; i < AUTO_PIN_LAST; i++) { @@ -13554,12 +13943,8 @@ static void alc861_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (nid >= 0x0c && nid <= 0x11) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - } + if (nid >= 0x0c && nid <= 0x11) + alc_set_input_pin(codec, nid, i); } } @@ -13595,7 +13980,7 @@ static int alc861_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC861_DIGOUT_NID; if (spec->kctls.list) @@ -13604,13 +13989,12 @@ static int alc861_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861_auto_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; spec->adc_nids = alc861_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861_adc_nids); set_capture_mixer(spec); - store_pin_configs(codec); return 1; } @@ -13819,6 +14203,12 @@ static int patch_alc861(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861_AUTO) setup_preset(spec, &alc861_presets[board_config]); @@ -13830,6 +14220,8 @@ static int patch_alc861(struct hda_codec *codec) spec->stream_digital_playback = &alc861_pcm_digital_playback; spec->stream_digital_capture = &alc861_pcm_digital_capture; + set_beep_amp(spec, 0x23, 0, HDA_OUTPUT); + spec->vmaster_nid = 0x03; codec->patch_ops = alc_patch_ops; @@ -13986,9 +14378,6 @@ static struct snd_kcontrol_new alc861vd_6st_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ }; @@ -14012,9 +14401,6 @@ static struct snd_kcontrol_new alc861vd_3st_mixer[] = { HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - { } /* end */ }; @@ -14053,8 +14439,6 @@ static struct snd_kcontrol_new alc861vd_dallas_mixer[] = { HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Beep Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Beep Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -14365,9 +14749,7 @@ static struct snd_pci_quirk alc861vd_cfg_tbl[] = { SND_PCI_QUIRK(0x1179, 0xff03, "Toshiba P205", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1179, 0xff31, "Toshiba L30-149", ALC861VD_DALLAS), SND_PCI_QUIRK(0x1565, 0x820d, "Biostar NF61S SE", ALC861VD_6ST_DIG), - SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x17aa, 0x3802, "Lenovo 3000 C200", ALC861VD_LENOVO), - SND_PCI_QUIRK(0x17aa, 0x384e, "Lenovo 3000 N200", ALC861VD_LENOVO), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", ALC861VD_LENOVO), SND_PCI_QUIRK(0x1849, 0x0862, "ASRock K8NF6G-VSTA", ALC861VD_6ST_DIG), {} }; @@ -14529,11 +14911,9 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; if (alc861vd_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN); - if (nid != ALC861VD_PIN_CD_NID) + alc_set_input_pin(codec, nid, i); + if (nid != ALC861VD_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -14699,7 +15079,7 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC861VD_DIGOUT_NID; if (spec->kctls.list) @@ -14708,13 +15088,12 @@ static int alc861vd_parse_auto_config(struct hda_codec *codec) add_verb(spec, alc861vd_volume_init_verbs); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -14765,6 +15144,12 @@ static int patch_alc861vd(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x23); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC861VD_AUTO) setup_preset(spec, &alc861vd_presets[board_config]); @@ -14787,9 +15172,10 @@ static int patch_alc861vd(struct hda_codec *codec) spec->adc_nids = alc861vd_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc861vd_adc_nids); spec->capsrc_nids = alc861vd_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; set_capture_mixer(spec); + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); spec->vmaster_nid = 0x02; @@ -14978,8 +15364,6 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -15001,8 +15385,6 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), { } /* end */ }; @@ -15978,56 +16360,55 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { }; static struct snd_pci_quirk alc662_cfg_tbl[] = { - SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), - SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), - SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), - SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), SND_PCI_QUIRK(0x1043, 0x1000, "ASUS N50Vm", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x11d3, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1203, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), - SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1339, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1813, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x11f3, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5), + SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x1843, "ASUS NB", ALC662_ASUS_MODE2), SND_PCI_QUIRK(0x1043, 0x1864, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1783, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1753, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x16c3, "ASUS NB", ALC662_ASUS_MODE2), - SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1876, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M51VA", ALC663_ASUS_M51VA), + /*SND_PCI_QUIRK(0x1043, 0x1878, "ASUS M50Vr", ALC663_ASUS_MODE1),*/ SND_PCI_QUIRK(0x1043, 0x1893, "ASUS M50Vm", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x11c3, "ASUS M70V", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3), SND_PCI_QUIRK(0x1043, 0x1894, "ASUS X55", ALC663_ASUS_MODE3), - SND_PCI_QUIRK(0x1043, 0x1092, "ASUS NB", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1903, "ASUS F5GL", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1913, "ASUS NB", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1933, "ASUS F80Q", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x1953, "ASUS NB", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71C", ALC663_ASUS_MODE3), + SND_PCI_QUIRK(0x1043, 0x1993, "ASUS N20", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS G50V", ALC663_ASUS_G50V), + /*SND_PCI_QUIRK(0x1043, 0x19a3, "ASUS NB", ALC663_ASUS_MODE1),*/ + SND_PCI_QUIRK(0x1043, 0x19b3, "ASUS F7Z", ALC663_ASUS_MODE1), + SND_PCI_QUIRK(0x1043, 0x19c3, "ASUS F5Z/F6x", ALC662_ASUS_MODE2), + SND_PCI_QUIRK(0x1043, 0x19e3, "ASUS NB", ALC663_ASUS_MODE1), SND_PCI_QUIRK(0x1043, 0x19f3, "ASUS NB", ALC663_ASUS_MODE4), - SND_PCI_QUIRK(0x1043, 0x1823, "ASUS NB", ALC663_ASUS_MODE5), - SND_PCI_QUIRK(0x1043, 0x1833, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1763, "ASUS NB", ALC663_ASUS_MODE6), - SND_PCI_QUIRK(0x1043, 0x1765, "ASUS NB", ALC663_ASUS_MODE6), + SND_PCI_QUIRK(0x1043, 0x8290, "ASUS P5GC-MX", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x1043, 0x82a1, "ASUS Eeepc", ALC662_ASUS_EEEPC_P701), + SND_PCI_QUIRK(0x1043, 0x82d1, "ASUS Eeepc EP20", ALC662_ASUS_EEEPC_EP20), + SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), - SND_PCI_QUIRK(0x1019, 0x9087, "ECS", ALC662_ECS), - SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), + SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo", ALC662_LENOVO_101E), SND_PCI_QUIRK(0x1849, 0x3662, "ASROCK K10N78FullHD-hSLI R3.0", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1854, 0x2000, "ASUS H13-2000", ALC663_ASUS_H13), - SND_PCI_QUIRK(0x1854, 0x2001, "ASUS H13-2001", ALC663_ASUS_H13), - SND_PCI_QUIRK(0x1854, 0x2002, "ASUS H13-2002", ALC663_ASUS_H13), + SND_PCI_QUIRK_MASK(0x1854, 0xf000, 0x2000, "ASUS H13-200x", + ALC663_ASUS_H13), {} }; @@ -16347,7 +16728,7 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, if (alc880_is_fixed_pin(pin)) { nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin)); - /* printk("DAC nid=%x\n",nid); */ + /* printk(KERN_DEBUG "DAC nid=%x\n",nid); */ /* specify the DAC as the extra output */ if (!spec->multiout.hp_nid) spec->multiout.hp_nid = nid; @@ -16377,26 +16758,58 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, return 0; } +/* return the index of the src widget from the connection list of the nid. + * return -1 if not found + */ +static int alc662_input_pin_idx(struct hda_codec *codec, hda_nid_t nid, + hda_nid_t src) +{ + hda_nid_t conn_list[HDA_MAX_CONNECTIONS]; + int i, conns; + + conns = snd_hda_get_connections(codec, nid, conn_list, + ARRAY_SIZE(conn_list)); + if (conns < 0) + return -1; + for (i = 0; i < conns; i++) + if (conn_list[i] == src) + return i; + return -1; +} + +static int alc662_is_input_pin(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned int pincap = snd_hda_query_pin_caps(codec, nid); + return (pincap & AC_PINCAP_IN) != 0; +} + /* create playback/capture controls for input pins */ -static int alc662_auto_create_analog_input_ctls(struct alc_spec *spec, +static int alc662_auto_create_analog_input_ctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { - struct hda_input_mux *imux = &spec->private_imux; + struct alc_spec *spec = codec->spec; + struct hda_input_mux *imux = &spec->private_imux[0]; int i, err, idx; for (i = 0; i < AUTO_PIN_LAST; i++) { - if (alc880_is_input_pin(cfg->input_pins[i])) { - idx = alc880_input_pin_idx(cfg->input_pins[i]); - err = new_analog_input(spec, cfg->input_pins[i], - auto_pin_cfg_labels[i], - idx, 0x0b); - if (err < 0) - return err; - imux->items[imux->num_items].label = - auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = - alc880_input_pin_idx(cfg->input_pins[i]); - imux->num_items++; + if (alc662_is_input_pin(codec, cfg->input_pins[i])) { + idx = alc662_input_pin_idx(codec, 0x0b, + cfg->input_pins[i]); + if (idx >= 0) { + err = new_analog_input(spec, cfg->input_pins[i], + auto_pin_cfg_labels[i], + idx, 0x0b); + if (err < 0) + return err; + } + idx = alc662_input_pin_idx(codec, 0x22, + cfg->input_pins[i]); + if (idx >= 0) { + imux->items[imux->num_items].label = + auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = idx; + imux->num_items++; + } } } return 0; @@ -16446,7 +16859,6 @@ static void alc662_auto_init_hp_out(struct hda_codec *codec) alc662_auto_set_output_and_unmute(codec, pin, PIN_OUT, 0); } -#define alc662_is_input_pin(nid) alc880_is_input_pin(nid) #define ALC662_PIN_CD_NID ALC880_PIN_CD_NID static void alc662_auto_init_analog_input(struct hda_codec *codec) @@ -16456,12 +16868,10 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) for (i = 0; i < AUTO_PIN_LAST; i++) { hda_nid_t nid = spec->autocfg.input_pins[i]; - if (alc662_is_input_pin(nid)) { - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - (i <= AUTO_PIN_FRONT_MIC ? - PIN_VREF80 : PIN_IN)); - if (nid != ALC662_PIN_CD_NID) + if (alc662_is_input_pin(codec, nid)) { + alc_set_input_pin(codec, nid, i); + if (nid != ALC662_PIN_CD_NID && + (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP)) snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); @@ -16499,20 +16909,20 @@ static int alc662_parse_auto_config(struct hda_codec *codec) "Headphone"); if (err < 0) return err; - err = alc662_auto_create_analog_input_ctls(spec, &spec->autocfg); + err = alc662_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = ALC880_DIGOUT_NID; if (spec->kctls.list) add_mixer(spec, spec->kctls.list); spec->num_mux_defs = 1; - spec->input_mux = &spec->private_imux; + spec->input_mux = &spec->private_imux[0]; add_verb(spec, alc662_auto_init_verbs); if (codec->vendor_id == 0x10ec0663) @@ -16522,7 +16932,6 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (err < 0) return err; - store_pin_configs(codec); return 1; } @@ -16574,6 +16983,12 @@ static int patch_alc662(struct hda_codec *codec) } } + err = snd_hda_attach_beep_device(codec, 0x1); + if (err < 0) { + alc_free(codec); + return err; + } + if (board_config != ALC662_AUTO) setup_preset(spec, &alc662_presets[board_config]); @@ -16597,10 +17012,14 @@ static int patch_alc662(struct hda_codec *codec) spec->adc_nids = alc662_adc_nids; spec->num_adc_nids = ARRAY_SIZE(alc662_adc_nids); spec->capsrc_nids = alc662_capsrc_nids; - spec->is_mix_capture = 1; + spec->capture_style = CAPT_MIX; if (!spec->cap_mixer) set_capture_mixer(spec); + if (codec->vendor_id == 0x10ec0662) + set_beep_amp(spec, 0x0b, 0x05, HDA_INPUT); + else + set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); spec->vmaster_nid = 0x02; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 35b83dc..61996a2 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -43,6 +43,7 @@ enum { }; enum { + STAC_AUTO, STAC_REF, STAC_9200_OQO, STAC_9200_DELL_D21, @@ -55,20 +56,24 @@ enum { STAC_9200_DELL_M25, STAC_9200_DELL_M26, STAC_9200_DELL_M27, - STAC_9200_GATEWAY, + STAC_9200_M4, + STAC_9200_M4_2, STAC_9200_PANASONIC, STAC_9200_MODELS }; enum { + STAC_9205_AUTO, STAC_9205_REF, STAC_9205_DELL_M42, STAC_9205_DELL_M43, STAC_9205_DELL_M44, + STAC_9205_EAPD, STAC_9205_MODELS }; enum { + STAC_92HD73XX_AUTO, STAC_92HD73XX_NO_JD, /* no jack-detection */ STAC_92HD73XX_REF, STAC_DELL_M6_AMIC, @@ -79,28 +84,40 @@ enum { }; enum { + STAC_92HD83XXX_AUTO, STAC_92HD83XXX_REF, + STAC_92HD83XXX_PWR_REF, + STAC_DELL_S14, STAC_92HD83XXX_MODELS }; enum { + STAC_92HD71BXX_AUTO, STAC_92HD71BXX_REF, STAC_DELL_M4_1, STAC_DELL_M4_2, STAC_DELL_M4_3, STAC_HP_M4, + STAC_HP_DV5, + STAC_HP_HDX, STAC_92HD71BXX_MODELS }; enum { + STAC_925x_AUTO, STAC_925x_REF, + STAC_M1, + STAC_M1_2, + STAC_M2, STAC_M2_2, - STAC_MA6, - STAC_PA6, + STAC_M3, + STAC_M5, + STAC_M6, STAC_925x_MODELS }; enum { + STAC_922X_AUTO, STAC_D945_REF, STAC_D945GTP3, STAC_D945GTP5, @@ -128,6 +145,7 @@ enum { }; enum { + STAC_927X_AUTO, STAC_D965_REF_NO_JD, /* no jack-detection */ STAC_D965_REF, STAC_D965_3ST, @@ -137,6 +155,12 @@ enum { STAC_927X_MODELS }; +enum { + STAC_9872_AUTO, + STAC_9872_VAIO, + STAC_9872_MODELS +}; + struct sigmatel_event { hda_nid_t nid; unsigned char type; @@ -160,6 +184,7 @@ struct sigmatel_spec { unsigned int alt_switch: 1; unsigned int hp_detect: 1; unsigned int spdif_mute: 1; + unsigned int check_volume_offset:1; /* gpio lines */ unsigned int eapd_mask; @@ -172,6 +197,7 @@ struct sigmatel_spec { unsigned int stream_delay; /* analog loopback */ + struct snd_kcontrol_new *aloopback_ctl; unsigned char aloopback_mask; unsigned char aloopback_shift; @@ -196,6 +222,8 @@ struct sigmatel_spec { hda_nid_t hp_dacs[5]; hda_nid_t speaker_dacs[5]; + int volume_offset; + /* capture */ hda_nid_t *adc_nids; unsigned int num_adcs; @@ -217,7 +245,6 @@ struct sigmatel_spec { /* pin widgets */ hda_nid_t *pin_nids; unsigned int num_pins; - unsigned int *pin_configs; /* codec specific stuff */ struct hda_verb *init; @@ -328,7 +355,11 @@ static hda_nid_t stac92hd83xxx_slave_dig_outs[2] = { }; static unsigned int stac92hd83xxx_pwr_mapping[4] = { - 0x03, 0x0c, 0x10, 0x40, + 0x03, 0x0c, 0x20, 0x40, +}; + +static hda_nid_t stac92hd83xxx_amp_nids[1] = { + 0xc, }; static hda_nid_t stac92hd71bxx_pwr_nids[3] = { @@ -389,6 +420,10 @@ static hda_nid_t stac922x_mux_nids[2] = { 0x12, 0x13, }; +static hda_nid_t stac927x_slave_dig_outs[2] = { + 0x1f, 0, +}; + static hda_nid_t stac927x_adc_nids[3] = { 0x07, 0x08, 0x09 }; @@ -461,15 +496,21 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { 0x14, 0x22, 0x23 }; -static hda_nid_t stac92hd83xxx_pin_nids[14] = { +static hda_nid_t stac92hd83xxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x10, 0x11, 0x12, 0x13, - 0x1d, 0x1e, 0x1f, 0x20 + 0x0f, 0x10, 0x11, 0x1f, 0x20, }; -static hda_nid_t stac92hd71bxx_pin_nids[11] = { + +#define STAC92HD71BXX_NUM_PINS 13 +static hda_nid_t stac92hd71bxx_pin_nids_4port[STAC92HD71BXX_NUM_PINS] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x00, + 0x00, 0x14, 0x18, 0x19, 0x1e, + 0x1f, 0x20, 0x27 +}; +static hda_nid_t stac92hd71bxx_pin_nids_6port[STAC92HD71BXX_NUM_PINS] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x18, 0x19, 0x1e, - 0x1f, + 0x1f, 0x20, 0x27 }; static hda_nid_t stac927x_pin_nids[14] = { @@ -831,13 +872,9 @@ static struct hda_verb stac92hd73xx_10ch_core_init[] = { }; static struct hda_verb stac92hd83xxx_core_init[] = { - /* start of config #1 */ - { 0xe, AC_VERB_SET_CONNECT_SEL, 0x3}, - - /* start of config #2 */ - { 0xa, AC_VERB_SET_CONNECT_SEL, 0x0}, - { 0xb, AC_VERB_SET_CONNECT_SEL, 0x0}, - { 0xd, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xa, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xb, AC_VERB_SET_CONNECT_SEL, 0x1}, + { 0xd, AC_VERB_SET_CONNECT_SEL, 0x0}, /* power state controls amps */ { 0x01, AC_VERB_SET_EAPD, 1 << 2}, @@ -847,26 +884,25 @@ static struct hda_verb stac92hd83xxx_core_init[] = { static struct hda_verb stac92hd71bxx_core_init[] = { /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ - { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; -#define HD_DISABLE_PORTF 2 +#define HD_DISABLE_PORTF 1 static struct hda_verb stac92hd71bxx_analog_core_init[] = { /* start of config #1 */ /* connect port 0f to audio mixer */ { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - /* unmute right and left channels for node 0x0f */ - { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* start of config #2 */ /* set master volume and direct control */ { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, - /* unmute right and left channels for nodes 0x0a, 0xd */ + {} +}; + +static struct hda_verb stac92hd71bxx_unmute_core_init[] = { + /* unmute right and left channels for nodes 0x0f, 0xa, 0x0d */ + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} @@ -875,6 +911,8 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { static struct hda_verb stac925x_core_init[] = { /* set dac0mux for dac converter */ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* mute the master volume */ + { 0x0e, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, {} }; @@ -945,16 +983,6 @@ static struct hda_verb stac9205_core_init[] = { .private_value = HDA_COMPOSE_AMP_VAL(nid, chs, idx, dir) \ } -#define STAC_INPUT_SOURCE(cnt) \ - { \ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ - .name = "Input Source", \ - .count = cnt, \ - .info = stac92xx_mux_enum_info, \ - .get = stac92xx_mux_enum_get, \ - .put = stac92xx_mux_enum_put, \ - } - #define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -969,7 +997,6 @@ static struct hda_verb stac9205_core_init[] = { static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), - STAC_INPUT_SOURCE(1), HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), { } /* end */ @@ -994,8 +1021,6 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1005,9 +1030,22 @@ static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { +static struct snd_kcontrol_new stac92hd73xx_6ch_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + {} +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_loopback[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), + {} +}; +static struct snd_kcontrol_new stac92hd73xx_10ch_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), + {} +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1032,8 +1070,6 @@ static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { }; static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -1085,9 +1121,6 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = { }; static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { - STAC_INPUT_SOURCE(2), - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1113,10 +1146,11 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { { } /* end */ }; -static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { - STAC_INPUT_SOURCE(2), - STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), +static struct snd_kcontrol_new stac92hd71bxx_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2) +}; +static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -1126,16 +1160,14 @@ static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { }; static struct snd_kcontrol_new stac925x_mixer[] = { - STAC_INPUT_SOURCE(1), + HDA_CODEC_VOLUME("Master Playback Volume", 0x0e, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x0e, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x14, 0, HDA_OUTPUT), { } /* end */ }; static struct snd_kcontrol_new stac9205_mixer[] = { - STAC_INPUT_SOURCE(2), - STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), @@ -1144,9 +1176,13 @@ static struct snd_kcontrol_new stac9205_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac9205_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), + {} +}; + /* This needs to be generated dynamically based on sequence */ static struct snd_kcontrol_new stac922x_mixer[] = { - STAC_INPUT_SOURCE(2), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), @@ -1157,9 +1193,6 @@ static struct snd_kcontrol_new stac922x_mixer[] = { static struct snd_kcontrol_new stac927x_mixer[] = { - STAC_INPUT_SOURCE(3), - STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), @@ -1171,6 +1204,11 @@ static struct snd_kcontrol_new stac927x_mixer[] = { { } /* end */ }; +static struct snd_kcontrol_new stac927x_loopback[] = { + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), + {} +}; + static struct snd_kcontrol_new stac_dmux_mixer = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Digital Input Source", @@ -1196,10 +1234,7 @@ static const char *slave_vols[] = { "LFE Playback Volume", "Side Playback Volume", "Headphone Playback Volume", - "Headphone Playback Volume", "Speaker Playback Volume", - "External Speaker Playback Volume", - "Speaker2 Playback Volume", NULL }; @@ -1210,10 +1245,7 @@ static const char *slave_sws[] = { "LFE Playback Switch", "Side Playback Switch", "Headphone Playback Switch", - "Headphone Playback Switch", "Speaker Playback Switch", - "External Speaker Playback Switch", - "Speaker2 Playback Switch", "IEC958 Playback Switch", NULL }; @@ -1283,6 +1315,8 @@ static int stac92xx_build_controls(struct hda_codec *codec) unsigned int vmaster_tlv[4]; snd_hda_set_vmaster_tlv(codec, spec->multiout.dac_nids[0], HDA_OUTPUT, vmaster_tlv); + /* correct volume offset */ + vmaster_tlv[2] += vmaster_tlv[3] * spec->volume_offset; err = snd_hda_add_vmaster(codec, "Master Playback Volume", vmaster_tlv, slave_vols); if (err < 0) @@ -1295,6 +1329,13 @@ static int stac92xx_build_controls(struct hda_codec *codec) return err; } + if (spec->aloopback_ctl && + snd_hda_get_bool_hint(codec, "loopback") == 1) { + err = snd_hda_add_new_ctls(codec, spec->aloopback_ctl); + if (err < 0) + return err; + } + stac92xx_free_kctls(codec); /* no longer needed */ /* create jack input elements */ @@ -1334,7 +1375,16 @@ static unsigned int ref9200_pin_configs[8] = { 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; -/* +static unsigned int gateway9200_m4_pin_configs[8] = { + 0x400000fe, 0x404500f4, 0x400100f0, 0x90110010, + 0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3, +}; +static unsigned int gateway9200_m4_2_pin_configs[8] = { + 0x400000fe, 0x404500f4, 0x400100f0, 0x90110010, + 0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3, +}; + +/* STAC 9200 pin configs for 102801A8 102801DE @@ -1464,10 +1514,13 @@ static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_9200_DELL_M25] = dell9200_m25_pin_configs, [STAC_9200_DELL_M26] = dell9200_m26_pin_configs, [STAC_9200_DELL_M27] = dell9200_m27_pin_configs, + [STAC_9200_M4] = gateway9200_m4_pin_configs, + [STAC_9200_M4_2] = gateway9200_m4_2_pin_configs, [STAC_9200_PANASONIC] = ref9200_pin_configs, }; static const char *stac9200_models[STAC_9200_MODELS] = { + [STAC_AUTO] = "auto", [STAC_REF] = "ref", [STAC_9200_OQO] = "oqo", [STAC_9200_DELL_D21] = "dell-d21", @@ -1480,7 +1533,8 @@ static const char *stac9200_models[STAC_9200_MODELS] = { [STAC_9200_DELL_M25] = "dell-m25", [STAC_9200_DELL_M26] = "dell-m26", [STAC_9200_DELL_M27] = "dell-m27", - [STAC_9200_GATEWAY] = "gateway", + [STAC_9200_M4] = "gateway-m4", + [STAC_9200_M4_2] = "gateway-m4-2", [STAC_9200_PANASONIC] = "panasonic", }; @@ -1488,6 +1542,8 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_REF), /* Dell laptops have BIOS problem */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01a8, "unknown Dell", STAC_9200_DELL_D21), @@ -1550,11 +1606,9 @@ static struct snd_pci_quirk stac9200_cfg_tbl[] = { /* Panasonic */ SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_9200_PANASONIC), /* Gateway machines needs EAPD to be set on resume */ - SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_9200_GATEWAY), - SND_PCI_QUIRK(0x107b, 0x0317, "Gateway MT3423, MX341*", - STAC_9200_GATEWAY), - SND_PCI_QUIRK(0x107b, 0x0318, "Gateway ML3019, MT3707", - STAC_9200_GATEWAY), + SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_9200_M4), + SND_PCI_QUIRK(0x107b, 0x0317, "Gateway MT3423, MX341*", STAC_9200_M4_2), + SND_PCI_QUIRK(0x107b, 0x0318, "Gateway ML3019, MT3707", STAC_9200_M4_2), /* OQO Mobile */ SND_PCI_QUIRK(0x1106, 0x3288, "OQO Model 2", STAC_9200_OQO), {} /* terminator */ @@ -1565,44 +1619,87 @@ static unsigned int ref925x_pin_configs[8] = { 0x90a70320, 0x02214210, 0x01019020, 0x9033032e, }; -static unsigned int stac925x_MA6_pin_configs[8] = { - 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021, - 0x90a70320, 0x90100211, 0x400003f1, 0x9033032e, +static unsigned int stac925xM1_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, }; -static unsigned int stac925x_PA6_pin_configs[8] = { - 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021, - 0x50a103f0, 0x90100211, 0x400003f1, 0x9033032e, +static unsigned int stac925xM1_2_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, +}; + +static unsigned int stac925xM2_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, }; static unsigned int stac925xM2_2_pin_configs[8] = { - 0x40c003f3, 0x424503f2, 0x04180011, 0x02a19020, - 0x50a103f0, 0x90100212, 0x400003f1, 0x9033032e, + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, +}; + +static unsigned int stac925xM3_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x503303f3, +}; + +static unsigned int stac925xM5_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x9033032e, +}; + +static unsigned int stac925xM6_pin_configs[8] = { + 0x40c003f4, 0x424503f2, 0x400000f3, 0x02a19020, + 0x40a000f0, 0x90100210, 0x400003f1, 0x90330320, }; static unsigned int *stac925x_brd_tbl[STAC_925x_MODELS] = { [STAC_REF] = ref925x_pin_configs, + [STAC_M1] = stac925xM1_pin_configs, + [STAC_M1_2] = stac925xM1_2_pin_configs, + [STAC_M2] = stac925xM2_pin_configs, [STAC_M2_2] = stac925xM2_2_pin_configs, - [STAC_MA6] = stac925x_MA6_pin_configs, - [STAC_PA6] = stac925x_PA6_pin_configs, + [STAC_M3] = stac925xM3_pin_configs, + [STAC_M5] = stac925xM5_pin_configs, + [STAC_M6] = stac925xM6_pin_configs, }; static const char *stac925x_models[STAC_925x_MODELS] = { + [STAC_925x_AUTO] = "auto", [STAC_REF] = "ref", + [STAC_M1] = "m1", + [STAC_M1_2] = "m1-2", + [STAC_M2] = "m2", [STAC_M2_2] = "m2-2", - [STAC_MA6] = "m6", - [STAC_PA6] = "pa6", + [STAC_M3] = "m3", + [STAC_M5] = "m5", + [STAC_M6] = "m6", +}; + +static struct snd_pci_quirk stac925x_codec_id_cfg_tbl[] = { + SND_PCI_QUIRK(0x107b, 0x0316, "Gateway M255", STAC_M2), + SND_PCI_QUIRK(0x107b, 0x0366, "Gateway MP6954", STAC_M5), + SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_M1), + SND_PCI_QUIRK(0x107b, 0x0681, "Gateway NX860", STAC_M2), + SND_PCI_QUIRK(0x107b, 0x0367, "Gateway MX6453", STAC_M1_2), + /* Not sure about the brand name for those */ + SND_PCI_QUIRK(0x107b, 0x0281, "Gateway mobile", STAC_M1), + SND_PCI_QUIRK(0x107b, 0x0507, "Gateway mobile", STAC_M3), + SND_PCI_QUIRK(0x107b, 0x0281, "Gateway mobile", STAC_M6), + SND_PCI_QUIRK(0x107b, 0x0685, "Gateway mobile", STAC_M2_2), + {} /* terminator */ }; static struct snd_pci_quirk stac925x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_REF), SND_PCI_QUIRK(0x8384, 0x7632, "Stac9202 Reference Board", STAC_REF), - SND_PCI_QUIRK(0x107b, 0x0316, "Gateway M255", STAC_REF), - SND_PCI_QUIRK(0x107b, 0x0366, "Gateway MP6954", STAC_REF), - SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_MA6), - SND_PCI_QUIRK(0x107b, 0x0681, "Gateway NX860", STAC_PA6), - SND_PCI_QUIRK(0x1002, 0x437b, "Gateway MX6453", STAC_M2_2), + + /* Default table for unknown ID */ + SND_PCI_QUIRK(0x1002, 0x437b, "Gateway mobile", STAC_M2_2), + {} /* terminator */ }; @@ -1629,6 +1726,7 @@ static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { }; static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_AUTO] = "auto", [STAC_92HD73XX_NO_JD] = "no-jd", [STAC_92HD73XX_REF] = "ref", [STAC_DELL_M6_AMIC] = "dell-m6-amic", @@ -1641,6 +1739,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD73XX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, "Dell Studio 1535", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255, @@ -1664,50 +1764,68 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { {} /* terminator */ }; -static unsigned int ref92hd83xxx_pin_configs[14] = { +static unsigned int ref92hd83xxx_pin_configs[10] = { 0x02214030, 0x02211010, 0x02a19020, 0x02170130, 0x01014050, 0x01819040, 0x01014020, 0x90a3014e, - 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x01451160, 0x98560170, }; +static unsigned int dell_s14_pin_configs[10] = { + 0x02214030, 0x02211010, 0x02a19020, 0x01014050, + 0x40f000f0, 0x01819040, 0x40f000f0, 0x90a60160, + 0x40f000f0, 0x40f000f0, +}; + static unsigned int *stac92hd83xxx_brd_tbl[STAC_92HD83XXX_MODELS] = { [STAC_92HD83XXX_REF] = ref92hd83xxx_pin_configs, + [STAC_92HD83XXX_PWR_REF] = ref92hd83xxx_pin_configs, + [STAC_DELL_S14] = dell_s14_pin_configs, }; static const char *stac92hd83xxx_models[STAC_92HD83XXX_MODELS] = { + [STAC_92HD83XXX_AUTO] = "auto", [STAC_92HD83XXX_REF] = "ref", + [STAC_92HD83XXX_PWR_REF] = "mic-ref", + [STAC_DELL_S14] = "dell-s14", }; static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, - "DFI LanParty", STAC_92HD71BXX_REF), + "DFI LanParty", STAC_92HD83XXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD83XXX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ba, + "unknown Dell", STAC_DELL_S14), {} /* terminator */ }; -static unsigned int ref92hd71bxx_pin_configs[11] = { +static unsigned int ref92hd71bxx_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, 0x0181302e, 0x01014010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, 0x01452050, + 0x90a000f0, 0x01452050, 0x01452050, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_1_pin_configs[11] = { +static unsigned int dell_m4_1_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, 0x4f0000f0, + 0x40f000f0, 0x4f0000f0, 0x4f0000f0, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_2_pin_configs[11] = { +static unsigned int dell_m4_2_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000, + 0x00000000 }; -static unsigned int dell_m4_3_pin_configs[11] = { +static unsigned int dell_m4_3_pin_configs[STAC92HD71BXX_NUM_PINS] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, 0x40f000f0, 0x40f000f0, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x044413b0, 0x044413b0, + 0x40f000f0, 0x044413b0, 0x044413b0, 0x00000000, + 0x00000000 }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -1716,28 +1834,39 @@ static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { [STAC_DELL_M4_2] = dell_m4_2_pin_configs, [STAC_DELL_M4_3] = dell_m4_3_pin_configs, [STAC_HP_M4] = NULL, + [STAC_HP_DV5] = NULL, + [STAC_HP_HDX] = NULL, }; static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { + [STAC_92HD71BXX_AUTO] = "auto", [STAC_92HD71BXX_REF] = "ref", [STAC_DELL_M4_1] = "dell-m4-1", [STAC_DELL_M4_2] = "dell-m4-2", [STAC_DELL_M4_3] = "dell-m4-3", [STAC_HP_M4] = "hp-m4", + [STAC_HP_DV5] = "hp-dv5", + [STAC_HP_HDX] = "hp-hdx", }; static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_92HD71BXX_REF), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f2, - "HP dv5", STAC_HP_M4), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30f4, - "HP dv7", STAC_HP_M4), - SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x30fc, - "HP dv7", STAC_HP_M4), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_92HD71BXX_REF), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3080, + "HP", STAC_HP_DV5), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x30f0, + "HP dv4-7", STAC_HP_DV5), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3600, + "HP dv4-7", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3610, + "HP HDX", STAC_HP_HDX), /* HDX18 */ SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361a, - "unknown HP", STAC_HP_M4), + "HP mini 1000", STAC_HP_M4), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b, + "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, "unknown Dell", STAC_DELL_M4_1), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0234, @@ -1889,6 +2018,7 @@ static unsigned int *stac922x_brd_tbl[STAC_922X_MODELS] = { }; static const char *stac922x_models[STAC_922X_MODELS] = { + [STAC_922X_AUTO] = "auto", [STAC_D945_REF] = "ref", [STAC_D945GTP5] = "5stack", [STAC_D945GTP3] = "3stack", @@ -1916,6 +2046,8 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D945_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_D945_REF), /* Intel 945G based systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0101, "Intel D945G", STAC_D945GTP3), @@ -1969,6 +2101,9 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { "Intel D945P", STAC_D945GTP3), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0707, "Intel D945P", STAC_D945GTP5), + /* other intel */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x0204, + "Intel D945", STAC_D945_REF), /* other systems */ /* Apple Intel Mac (Mac Mini, MacBook, MacBook Pro...) */ SND_PCI_QUIRK(0x8384, 0x7680, @@ -1993,31 +2128,7 @@ static struct snd_pci_quirk stac922x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d7, "Dell XPS M1210", STAC_922X_DELL_M82), /* ECS/PC Chips boards */ - SND_PCI_QUIRK(0x1019, 0x2144, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2608, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2633, - "ECS/PC chips P17G/1333", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2811, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2812, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2813, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2814, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2815, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2816, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2817, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2818, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2819, - "ECS/PC chips", STAC_ECS_202), - SND_PCI_QUIRK(0x1019, 0x2820, + SND_PCI_QUIRK_MASK(0x1019, 0xf000, 0x2000, "ECS/PC chips", STAC_ECS_202), {} /* terminator */ }; @@ -2060,6 +2171,7 @@ static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { }; static const char *stac927x_models[STAC_927X_MODELS] = { + [STAC_927X_AUTO] = "auto", [STAC_D965_REF_NO_JD] = "ref-no-jd", [STAC_D965_REF] = "ref", [STAC_D965_3ST] = "3stack", @@ -2072,26 +2184,16 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_D965_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_D965_REF), /* Intel 946 based systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x3d01, "Intel D946", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0xa301, "Intel D946", STAC_D965_3ST), /* 965 based 3 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2116, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2115, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2114, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2113, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2112, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2111, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2110, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2009, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2008, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2007, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2006, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2005, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2004, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2100, + "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2000, + "Intel D965", STAC_D965_3ST), /* Dell 3 stack systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), @@ -2107,15 +2209,10 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_DELL_BIOS), /* 965 based 5 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2304, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2305, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2501, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2502, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2503, "Intel D965", STAC_D965_5ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2504, "Intel D965", STAC_D965_5ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2300, + "Intel D965", STAC_D965_5ST), + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_INTEL, 0xff00, 0x2500, + "Intel D965", STAC_D965_5ST), {} /* terminator */ }; @@ -2168,19 +2265,25 @@ static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { [STAC_9205_DELL_M42] = dell_9205_m42_pin_configs, [STAC_9205_DELL_M43] = dell_9205_m43_pin_configs, [STAC_9205_DELL_M44] = dell_9205_m44_pin_configs, + [STAC_9205_EAPD] = NULL, }; static const char *stac9205_models[STAC_9205_MODELS] = { + [STAC_9205_AUTO] = "auto", [STAC_9205_REF] = "ref", [STAC_9205_DELL_M42] = "dell-m42", [STAC_9205_DELL_M43] = "dell-m43", [STAC_9205_DELL_M44] = "dell-m44", + [STAC_9205_EAPD] = "eapd", }; static struct snd_pci_quirk stac9205_cfg_tbl[] = { /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, + "DFI LanParty", STAC_9205_REF), + /* Dell */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f1, "unknown Dell", STAC_9205_DELL_M42), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f2, @@ -2211,101 +2314,24 @@ static struct snd_pci_quirk stac9205_cfg_tbl[] = { "Dell Inspiron", STAC_9205_DELL_M44), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0228, "Dell Vostro 1500", STAC_9205_DELL_M42), + /* Gateway */ + SND_PCI_QUIRK(0x107b, 0x0565, "Gateway T1616", STAC_9205_EAPD), {} /* terminator */ }; -static int stac92xx_save_bios_config_regs(struct hda_codec *codec) -{ - int i; - struct sigmatel_spec *spec = codec->spec; - - kfree(spec->pin_configs); - spec->pin_configs = kcalloc(spec->num_pins, sizeof(*spec->pin_configs), - GFP_KERNEL); - if (!spec->pin_configs) - return -ENOMEM; - - for (i = 0; i < spec->num_pins; i++) { - hda_nid_t nid = spec->pin_nids[i]; - unsigned int pin_cfg; - - pin_cfg = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x bios pin config %8.8x\n", - nid, pin_cfg); - spec->pin_configs[i] = pin_cfg; - } - - return 0; -} - -static void stac92xx_set_config_reg(struct hda_codec *codec, - hda_nid_t pin_nid, unsigned int pin_config) -{ - int i; - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - pin_config & 0x000000ff); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (pin_config & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (pin_config & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - pin_config >> 24); - i = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", - pin_nid, i); -} - -static void stac92xx_set_config_regs(struct hda_codec *codec) +static void stac92xx_set_config_regs(struct hda_codec *codec, + unsigned int *pincfgs) { int i; struct sigmatel_spec *spec = codec->spec; - if (!spec->pin_configs) - return; + if (!pincfgs) + return; for (i = 0; i < spec->num_pins; i++) - stac92xx_set_config_reg(codec, spec->pin_nids[i], - spec->pin_configs[i]); -} - -static int stac_save_pin_cfgs(struct hda_codec *codec, unsigned int *pins) -{ - struct sigmatel_spec *spec = codec->spec; - - if (!pins) - return stac92xx_save_bios_config_regs(codec); - - kfree(spec->pin_configs); - spec->pin_configs = kmemdup(pins, - spec->num_pins * sizeof(*pins), - GFP_KERNEL); - if (!spec->pin_configs) - return -ENOMEM; - - stac92xx_set_config_regs(codec); - return 0; -} - -static void stac_change_pin_config(struct hda_codec *codec, hda_nid_t nid, - unsigned int cfg) -{ - struct sigmatel_spec *spec = codec->spec; - int i; - - for (i = 0; i < spec->num_pins; i++) { - if (spec->pin_nids[i] == nid) { - spec->pin_configs[i] = cfg; - stac92xx_set_config_reg(codec, nid, cfg); - break; - } - } + if (spec->pin_nids[i] && pincfgs[i]) + snd_hda_codec_set_pincfg(codec, spec->pin_nids[i], + pincfgs[i]); } /* @@ -2370,6 +2396,14 @@ static int stac92xx_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, stream_tag, format, substream); } +static int stac92xx_dig_playback_pcm_cleanup(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + struct snd_pcm_substream *substream) +{ + struct sigmatel_spec *spec = codec->spec; + return snd_hda_multi_out_dig_cleanup(codec, &spec->multiout); +} + /* * Analog capture callbacks @@ -2414,7 +2448,8 @@ static struct hda_pcm_stream stac92xx_pcm_digital_playback = { .ops = { .open = stac92xx_dig_playback_pcm_open, .close = stac92xx_dig_playback_pcm_close, - .prepare = stac92xx_dig_playback_pcm_prepare + .prepare = stac92xx_dig_playback_pcm_prepare, + .cleanup = stac92xx_dig_playback_pcm_cleanup }, }; @@ -2469,6 +2504,8 @@ static int stac92xx_build_pcms(struct hda_codec *codec) info->name = "STAC92xx Analog"; info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_analog_playback; + info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = + spec->multiout.dac_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE] = stac92xx_pcm_analog_capture; info->stream[SNDRV_PCM_STREAM_CAPTURE].nid = spec->adc_nids[0]; info->stream[SNDRV_PCM_STREAM_CAPTURE].substreams = spec->num_adcs; @@ -2484,7 +2521,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) codec->num_pcms++; info++; info->name = "STAC92xx Digital"; - info->pcm_type = HDA_PCM_TYPE_SPDIF; + info->pcm_type = spec->autocfg.dig_out_type[0]; if (spec->multiout.dig_out_nid) { info->stream[SNDRV_PCM_STREAM_PLAYBACK] = stac92xx_pcm_digital_playback; info->stream[SNDRV_PCM_STREAM_PLAYBACK].nid = spec->multiout.dig_out_nid; @@ -2500,8 +2537,7 @@ static int stac92xx_build_pcms(struct hda_codec *codec) static unsigned int stac92xx_get_vref(struct hda_codec *codec, hda_nid_t nid) { - unsigned int pincap = snd_hda_param_read(codec, nid, - AC_PAR_PIN_CAP); + unsigned int pincap = snd_hda_query_pin_caps(codec, nid); pincap = (pincap & AC_PINCAP_VREF) >> AC_PINCAP_VREF_SHIFT; if (pincap & AC_PINCAP_VREF_100) return AC_PINCTL_VREF_100; @@ -2676,22 +2712,37 @@ static struct snd_kcontrol_new stac92xx_control_templates[] = { }; /* add dynamic controls */ -static int stac92xx_add_control_temp(struct sigmatel_spec *spec, - struct snd_kcontrol_new *ktemp, - int idx, const char *name, - unsigned long val) +static struct snd_kcontrol_new * +stac_control_new(struct sigmatel_spec *spec, + struct snd_kcontrol_new *ktemp, + const char *name) { struct snd_kcontrol_new *knew; snd_array_init(&spec->kctls, sizeof(*knew), 32); knew = snd_array_new(&spec->kctls); if (!knew) - return -ENOMEM; + return NULL; *knew = *ktemp; - knew->index = idx; knew->name = kstrdup(name, GFP_KERNEL); - if (!knew->name) + if (!knew->name) { + /* roolback */ + memset(knew, 0, sizeof(*knew)); + spec->kctls.alloced--; + return NULL; + } + return knew; +} + +static int stac92xx_add_control_temp(struct sigmatel_spec *spec, + struct snd_kcontrol_new *ktemp, + int idx, const char *name, + unsigned long val) +{ + struct snd_kcontrol_new *knew = stac_control_new(spec, ktemp, name); + if (!knew) return -ENOMEM; + knew->index = idx; knew->private_value = val; return 0; } @@ -2713,6 +2764,29 @@ static inline int stac92xx_add_control(struct sigmatel_spec *spec, int type, return stac92xx_add_control_idx(spec, type, 0, name, val); } +static struct snd_kcontrol_new stac_input_src_temp = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Input Source", + .info = stac92xx_mux_enum_info, + .get = stac92xx_mux_enum_get, + .put = stac92xx_mux_enum_put, +}; + +static int stac92xx_add_input_source(struct sigmatel_spec *spec) +{ + struct snd_kcontrol_new *knew; + struct hda_input_mux *imux = &spec->private_imux; + + if (!spec->num_adcs || imux->num_items <= 1) + return 0; /* no need for input source control */ + knew = stac_control_new(spec, &stac_input_src_temp, + stac_input_src_temp.name); + if (!knew) + return -ENOMEM; + knew->count = spec->num_adcs; + return 0; +} + /* check whether the line-input can be used as line-out */ static hda_nid_t check_line_out_switch(struct hda_codec *codec) { @@ -2724,7 +2798,7 @@ static hda_nid_t check_line_out_switch(struct hda_codec *codec) if (cfg->line_out_type != AUTO_PIN_LINE_OUT) return 0; nid = cfg->input_pins[AUTO_PIN_LINE]; - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_OUT) return nid; return 0; @@ -2743,12 +2817,11 @@ static hda_nid_t check_mic_out_switch(struct hda_codec *codec) mic_pin = AUTO_PIN_MIC; for (;;) { hda_nid_t nid = cfg->input_pins[mic_pin]; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); /* some laptops have an internal analog microphone * which can't be used as a output */ if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) { - pincap = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); + pincap = snd_hda_query_pin_caps(codec, nid); if (pincap & AC_PINCAP_OUT) return nid; } @@ -2796,8 +2869,7 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) conn_len = snd_hda_get_connections(codec, nid, conn, HDA_MAX_CONNECTIONS); for (j = 0; j < conn_len; j++) { - wcaps = snd_hda_param_read(codec, conn[j], - AC_PAR_AUDIO_WIDGET_CAP); + wcaps = get_wcaps(codec, conn[j]); wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; /* we check only analog outputs */ if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) @@ -2812,6 +2884,16 @@ static hda_nid_t get_unassigned_dac(struct hda_codec *codec, hda_nid_t nid) return conn[j]; } } + /* if all DACs are already assigned, connect to the primary DAC */ + if (conn_len > 1) { + for (j = 0; j < conn_len; j++) { + if (conn[j] == spec->multiout.dac_nids[0]) { + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, j); + break; + } + } + } return 0; } @@ -2852,6 +2934,26 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) add_spec_dacs(spec, dac); } + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + dac = get_unassigned_dac(codec, nid); + if (dac) { + if (!spec->multiout.hp_nid) + spec->multiout.hp_nid = dac; + else + add_spec_extra_dacs(spec, dac); + } + spec->hp_dacs[i] = dac; + } + + for (i = 0; i < cfg->speaker_outs; i++) { + nid = cfg->speaker_pins[i]; + dac = get_unassigned_dac(codec, nid); + if (dac) + add_spec_extra_dacs(spec, dac); + spec->speaker_dacs[i] = dac; + } + /* add line-in as output */ nid = check_line_out_switch(codec); if (nid) { @@ -2879,26 +2981,6 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } } - for (i = 0; i < cfg->hp_outs; i++) { - nid = cfg->hp_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) { - if (!spec->multiout.hp_nid) - spec->multiout.hp_nid = dac; - else - add_spec_extra_dacs(spec, dac); - } - spec->hp_dacs[i] = dac; - } - - for (i = 0; i < cfg->speaker_outs; i++) { - nid = cfg->speaker_pins[i]; - dac = get_unassigned_dac(codec, nid); - if (dac) - add_spec_extra_dacs(spec, dac); - spec->speaker_dacs[i] = dac; - } - snd_printd("stac92xx: dac_nids=%d (0x%x/0x%x/0x%x/0x%x/0x%x)\n", spec->multiout.num_dacs, spec->multiout.dac_nids[0], @@ -2911,24 +2993,47 @@ static int stac92xx_auto_fill_dac_nids(struct hda_codec *codec) } /* create volume control/switch for the given prefx type */ -static int create_controls(struct sigmatel_spec *spec, const char *pfx, hda_nid_t nid, int chs) +static int create_controls_idx(struct hda_codec *codec, const char *pfx, + int idx, hda_nid_t nid, int chs) { + struct sigmatel_spec *spec = codec->spec; char name[32]; int err; + if (!spec->check_volume_offset) { + unsigned int caps, step, nums, db_scale; + caps = query_amp_caps(codec, nid, HDA_OUTPUT); + step = (caps & AC_AMPCAP_STEP_SIZE) >> + AC_AMPCAP_STEP_SIZE_SHIFT; + step = (step + 1) * 25; /* in .01dB unit */ + nums = (caps & AC_AMPCAP_NUM_STEPS) >> + AC_AMPCAP_NUM_STEPS_SHIFT; + db_scale = nums * step; + /* if dB scale is over -64dB, and finer enough, + * let's reduce it to half + */ + if (db_scale > 6400 && nums >= 0x1f) + spec->volume_offset = nums / 2; + spec->check_volume_offset = 1; + } + sprintf(name, "%s Playback Volume", pfx); - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, name, - HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); + err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_VOL, idx, name, + HDA_COMPOSE_AMP_VAL_OFS(nid, chs, 0, HDA_OUTPUT, + spec->volume_offset)); if (err < 0) return err; sprintf(name, "%s Playback Switch", pfx); - err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, name, + err = stac92xx_add_control_idx(spec, STAC_CTL_WIDGET_MUTE, idx, name, HDA_COMPOSE_AMP_VAL(nid, chs, 0, HDA_OUTPUT)); if (err < 0) return err; return 0; } +#define create_controls(codec, pfx, nid, chs) \ + create_controls_idx(codec, pfx, 0, nid, chs) + static int add_spec_dacs(struct sigmatel_spec *spec, hda_nid_t nid) { if (spec->multiout.num_dacs > 4) { @@ -2954,40 +3059,32 @@ static int add_spec_extra_dacs(struct sigmatel_spec *spec, hda_nid_t nid) return 1; } -static int is_unique_dac(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - - if (spec->autocfg.line_outs != 1) - return 0; - if (spec->multiout.hp_nid == nid) - return 0; - for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) - if (spec->multiout.extra_out_nid[i] == nid) - return 0; - return 1; -} - -/* add playback controls from the parsed DAC table */ -static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, - const struct auto_pin_cfg *cfg) +/* Create output controls + * The mixer elements are named depending on the given type (AUTO_PIN_XXX_OUT) + */ +static int create_multi_out_ctls(struct hda_codec *codec, int num_outs, + const hda_nid_t *pins, + const hda_nid_t *dac_nids, + int type) { struct sigmatel_spec *spec = codec->spec; static const char *chname[4] = { "Front", "Surround", NULL /*CLFE*/, "Side" }; - hda_nid_t nid = 0; + hda_nid_t nid; int i, err; unsigned int wid_caps; - for (i = 0; i < cfg->line_outs && spec->multiout.dac_nids[i]; i++) { - nid = spec->multiout.dac_nids[i]; - if (i == 2) { + for (i = 0; i < num_outs && i < ARRAY_SIZE(chname); i++) { + nid = dac_nids[i]; + if (!nid) + continue; + if (type != AUTO_PIN_HP_OUT && i == 2) { /* Center/LFE */ - err = create_controls(spec, "Center", nid, 1); + err = create_controls(codec, "Center", nid, 1); if (err < 0) return err; - err = create_controls(spec, "LFE", nid, 2); + err = create_controls(codec, "LFE", nid, 2); if (err < 0) return err; @@ -3003,23 +3100,47 @@ static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, } } else { - const char *name = chname[i]; - /* if it's a single DAC, assign a better name */ - if (!i && is_unique_dac(spec, nid)) { - switch (cfg->line_out_type) { - case AUTO_PIN_HP_OUT: - name = "Headphone"; - break; - case AUTO_PIN_SPEAKER_OUT: - name = "Speaker"; - break; - } + const char *name; + int idx; + switch (type) { + case AUTO_PIN_HP_OUT: + name = "Headphone"; + idx = i; + break; + case AUTO_PIN_SPEAKER_OUT: + name = "Speaker"; + idx = i; + break; + default: + name = chname[i]; + idx = 0; + break; } - err = create_controls(spec, name, nid, 3); + err = create_controls_idx(codec, name, idx, nid, 3); if (err < 0) return err; + if (type == AUTO_PIN_HP_OUT && !spec->hp_detect) { + wid_caps = get_wcaps(codec, pins[i]); + if (wid_caps & AC_WCAP_UNSOL_CAP) + spec->hp_detect = 1; + } } } + return 0; +} + +/* add playback controls from the parsed DAC table */ +static int stac92xx_auto_create_multi_out_ctls(struct hda_codec *codec, + const struct auto_pin_cfg *cfg) +{ + struct sigmatel_spec *spec = codec->spec; + int err; + + err = create_multi_out_ctls(codec, cfg->line_outs, cfg->line_out_pins, + spec->multiout.dac_nids, + cfg->line_out_type); + if (err < 0) + return err; if (cfg->hp_outs > 1 && cfg->line_out_type == AUTO_PIN_LINE_OUT) { err = stac92xx_add_control(spec, @@ -3054,40 +3175,18 @@ static int stac92xx_auto_create_hp_ctls(struct hda_codec *codec, struct auto_pin_cfg *cfg) { struct sigmatel_spec *spec = codec->spec; - hda_nid_t nid; - int i, err, nums; + int err; + + err = create_multi_out_ctls(codec, cfg->hp_outs, cfg->hp_pins, + spec->hp_dacs, AUTO_PIN_HP_OUT); + if (err < 0) + return err; + + err = create_multi_out_ctls(codec, cfg->speaker_outs, cfg->speaker_pins, + spec->speaker_dacs, AUTO_PIN_SPEAKER_OUT); + if (err < 0) + return err; - nums = 0; - for (i = 0; i < cfg->hp_outs; i++) { - static const char *pfxs[] = { - "Headphone", "Headphone2", "Headphone3", - }; - unsigned int wid_caps = get_wcaps(codec, cfg->hp_pins[i]); - if (wid_caps & AC_WCAP_UNSOL_CAP) - spec->hp_detect = 1; - if (nums >= ARRAY_SIZE(pfxs)) - continue; - nid = spec->hp_dacs[i]; - if (!nid) - continue; - err = create_controls(spec, pfxs[nums++], nid, 3); - if (err < 0) - return err; - } - nums = 0; - for (i = 0; i < cfg->speaker_outs; i++) { - static const char *pfxs[] = { - "Speaker", "External Speaker", "Speaker2", - }; - if (nums >= ARRAY_SIZE(pfxs)) - continue; - nid = spec->speaker_dacs[i]; - if (!nid) - continue; - err = create_controls(spec, pfxs[nums++], nid, 3); - if (err < 0) - return err; - } return 0; } @@ -3296,11 +3395,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, unsigned int wcaps; unsigned int def_conf; - def_conf = snd_hda_codec_read(codec, - spec->dmic_nids[i], - 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0); + def_conf = snd_hda_codec_get_pincfg(codec, spec->dmic_nids[i]); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; @@ -3424,6 +3519,7 @@ static void stac92xx_auto_init_hp_out(struct hda_codec *codec) static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out, hda_nid_t dig_in) { struct sigmatel_spec *spec = codec->spec; + int hp_swap = 0; int err; if ((err = snd_hda_parse_pin_def_config(codec, @@ -3451,6 +3547,7 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out spec->autocfg.line_outs = spec->autocfg.hp_outs; spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; spec->autocfg.hp_outs = 0; + hp_swap = 1; } if (spec->autocfg.mono_out_pin) { int dir = get_wcaps(codec, spec->autocfg.mono_out_pin) & @@ -3506,13 +3603,12 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out err = stac92xx_auto_fill_dac_nids(codec); if (err < 0) return err; + err = stac92xx_auto_create_multi_out_ctls(codec, + &spec->autocfg); + if (err < 0) + return err; } - err = stac92xx_auto_create_multi_out_ctls(codec, &spec->autocfg); - - if (err < 0) - return err; - /* setup analog beep controls */ if (spec->anabeep_nid > 0) { err = stac92xx_auto_create_beep_ctls(codec, @@ -3545,12 +3641,19 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out #endif err = stac92xx_auto_create_hp_ctls(codec, &spec->autocfg); - if (err < 0) return err; - err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); + /* All output parsing done, now restore the swapped hp pins */ + if (hp_swap) { + memcpy(spec->autocfg.hp_pins, spec->autocfg.line_out_pins, + sizeof(spec->autocfg.hp_pins)); + spec->autocfg.hp_outs = spec->autocfg.line_outs; + spec->autocfg.line_out_type = AUTO_PIN_HP_OUT; + spec->autocfg.line_outs = 0; + } + err = stac92xx_auto_create_analog_input_ctls(codec, &spec->autocfg); if (err < 0) return err; @@ -3579,11 +3682,15 @@ static int stac92xx_parse_auto_config(struct hda_codec *codec, hda_nid_t dig_out return err; } + err = stac92xx_add_input_source(spec); + if (err < 0) + return err; + spec->multiout.max_channels = spec->multiout.num_dacs * 2; if (spec->multiout.max_channels > 2) spec->surr_switch = 1; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = dig_out; if (dig_in && spec->autocfg.dig_in_pin) spec->dig_in_nid = dig_in; @@ -3646,9 +3753,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, for (i = 0; i < spec->autocfg.line_outs && lfe_pin == 0x0; i++) { hda_nid_t pin = spec->autocfg.line_out_pins[i]; unsigned int defcfg; - defcfg = snd_hda_codec_read(codec, pin, 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); + defcfg = snd_hda_codec_get_pincfg(codec, pin); if (get_defcfg_device(defcfg) == AC_JACK_SPEAKER) { unsigned int wcaps = get_wcaps(codec, pin); wcaps &= (AC_WCAP_STEREO | AC_WCAP_OUT_AMP); @@ -3661,7 +3766,7 @@ static int stac9200_auto_create_lfe_ctls(struct hda_codec *codec, } if (lfe_pin) { - err = create_controls(spec, "LFE", lfe_pin, 1); + err = create_controls(codec, "LFE", lfe_pin, 1); if (err < 0) return err; } @@ -3692,7 +3797,11 @@ static int stac9200_parse_auto_config(struct hda_codec *codec) return err; } - if (spec->autocfg.dig_out_pin) + err = stac92xx_add_input_source(spec); + if (err < 0) + return err; + + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = 0x05; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = 0x04; @@ -3748,8 +3857,7 @@ static int stac92xx_add_jack(struct hda_codec *codec, #ifdef CONFIG_SND_JACK struct sigmatel_spec *spec = codec->spec; struct sigmatel_jack *jack; - int def_conf = snd_hda_codec_read(codec, nid, - 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + int def_conf = snd_hda_codec_get_pincfg(codec, nid); int connectivity = get_defcfg_connect(def_conf); char name[32]; @@ -3864,6 +3972,36 @@ static void stac92xx_power_down(struct hda_codec *codec) static void stac_toggle_power_map(struct hda_codec *codec, hda_nid_t nid, int enable); +/* override some hints from the hwdep entry */ +static void stac_store_hints(struct hda_codec *codec) +{ + struct sigmatel_spec *spec = codec->spec; + const char *p; + int val; + + val = snd_hda_get_bool_hint(codec, "hp_detect"); + if (val >= 0) + spec->hp_detect = val; + p = snd_hda_get_hint(codec, "gpio_mask"); + if (p) { + spec->gpio_mask = simple_strtoul(p, NULL, 0); + spec->eapd_mask = spec->gpio_dir = spec->gpio_data = + spec->gpio_mask; + } + p = snd_hda_get_hint(codec, "gpio_dir"); + if (p) + spec->gpio_dir = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "gpio_data"); + if (p) + spec->gpio_data = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + p = snd_hda_get_hint(codec, "eapd_mask"); + if (p) + spec->eapd_mask = simple_strtoul(p, NULL, 0) & spec->gpio_mask; + val = snd_hda_get_bool_hint(codec, "eapd_switch"); + if (val >= 0) + spec->eapd_switch = val; +} + static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -3880,6 +4018,9 @@ static int stac92xx_init(struct hda_codec *codec) spec->adc_nids[i], 0, AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + /* override some hints */ + stac_store_hints(codec); + /* set up GPIO */ gpio = spec->gpio_data; /* turn on EAPD statically when spec->eapd_switch isn't set. @@ -3929,8 +4070,7 @@ static int stac92xx_init(struct hda_codec *codec) pinctl); } } - conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + conf = snd_hda_codec_get_pincfg(codec, nid); if (get_defcfg_connect(conf) != AC_JACK_PORT_FIXED) { enable_pin_detect(codec, nid, STAC_INSERT_EVENT); @@ -3942,8 +4082,8 @@ static int stac92xx_init(struct hda_codec *codec) for (i = 0; i < spec->num_dmics; i++) stac92xx_auto_set_pinctl(codec, spec->dmic_nids[i], AC_PINCTL_IN_EN); - if (cfg->dig_out_pin) - stac92xx_auto_set_pinctl(codec, cfg->dig_out_pin, + if (cfg->dig_out_pins[0]) + stac92xx_auto_set_pinctl(codec, cfg->dig_out_pins[0], AC_PINCTL_OUT_EN); if (cfg->dig_in_pin) stac92xx_auto_set_pinctl(codec, cfg->dig_in_pin, @@ -3971,8 +4111,7 @@ static int stac92xx_init(struct hda_codec *codec) stac_toggle_power_map(codec, nid, 1); continue; } - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); def_conf = get_defcfg_connect(def_conf); /* skip any ports that don't have jacks since presence * detection is useless */ @@ -4026,7 +4165,6 @@ static void stac92xx_free(struct hda_codec *codec) if (! spec) return; - kfree(spec->pin_configs); stac92xx_free_jacks(codec); snd_array_free(&spec->events); @@ -4037,7 +4175,9 @@ static void stac92xx_free(struct hda_codec *codec) static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, unsigned int flag) { - unsigned int pin_ctl = snd_hda_codec_read(codec, nid, + unsigned int old_ctl, pin_ctl; + + pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); if (pin_ctl & AC_PINCTL_IN_EN) { @@ -4051,14 +4191,17 @@ static void stac92xx_set_pinctl(struct hda_codec *codec, hda_nid_t nid, return; } + old_ctl = pin_ctl; /* if setting pin direction bits, clear the current direction bits first */ if (flag & (AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN)) pin_ctl &= ~(AC_PINCTL_IN_EN | AC_PINCTL_OUT_EN); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl | flag); + pin_ctl |= flag; + if (old_ctl != pin_ctl) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_ctl); } static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, @@ -4066,9 +4209,10 @@ static void stac92xx_reset_pinctl(struct hda_codec *codec, hda_nid_t nid, { unsigned int pin_ctl = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0x00); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pin_ctl & ~flag); + if (pin_ctl & flag) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + pin_ctl & ~flag); } static int get_pin_presence(struct hda_codec *codec, hda_nid_t nid) @@ -4163,8 +4307,19 @@ static void stac92xx_hp_detect(struct hda_codec *codec) continue; if (presence) stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); +#if 0 /* FIXME */ +/* Resetting the pinctl like below may lead to (a sort of) regressions + * on some devices since they use the HP pin actually for line/speaker + * outs although the default pin config shows a different pin (that is + * wrong and useless). + * + * So, it's basically a problem of default pin configs, likely a BIOS issue. + * But, disabling the code below just works around it, and I'm too tired of + * bug reports with such devices... + */ else stac92xx_reset_pinctl(codec, cfg->hp_pins[i], val); +#endif /* FIXME */ } } @@ -4258,6 +4413,24 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) if (spec->num_pwrs > 0) stac92xx_pin_sense(codec, event->nid); stac92xx_report_jack(codec, event->nid); + + switch (codec->subsystem_id) { + case 0x103c308f: + if (event->nid == 0xb) { + int pin = AC_PINCTL_IN_EN; + + if (get_pin_presence(codec, 0xa) + && get_pin_presence(codec, 0xb)) + pin |= AC_PINCTL_VREF_80; + if (!get_pin_presence(codec, 0xb)) + pin |= AC_PINCTL_VREF_80; + + /* toggle VREF state based on mic + hp pin + * status + */ + stac92xx_auto_set_pinctl(codec, 0x0a, pin); + } + } break; case STAC_VREF_EVENT: data = snd_hda_codec_read(codec, codec->afg, 0, @@ -4320,7 +4493,6 @@ static int stac92xx_resume(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - stac92xx_set_config_regs(codec); stac92xx_init(codec); snd_hda_codec_resume_amp(codec); snd_hda_codec_resume_cache(codec); @@ -4331,6 +4503,37 @@ static int stac92xx_resume(struct hda_codec *codec) return 0; } + +/* + * using power check for controlling mute led of HP HDX notebooks + * check for mute state only on Speakers (nid = 0x10) + * + * For this feature CONFIG_SND_HDA_POWER_SAVE is needed, otherwise + * the LED is NOT working properly ! + */ + +#ifdef CONFIG_SND_HDA_POWER_SAVE +static int stac92xx_hp_hdx_check_power_status(struct hda_codec *codec, + hda_nid_t nid) +{ + struct sigmatel_spec *spec = codec->spec; + + if (nid == 0x10) { + if (snd_hda_codec_amp_read(codec, nid, 0, HDA_OUTPUT, 0) & + HDA_AMP_MUTE) + spec->gpio_data &= ~0x08; /* orange */ + else + spec->gpio_data |= 0x08; /* white */ + + stac_gpio_set(codec, spec->gpio_mask, + spec->gpio_dir, + spec->gpio_data); + } + + return 0; +} +#endif + static int stac92xx_suspend(struct hda_codec *codec, pm_message_t state) { struct sigmatel_spec *spec = codec->spec; @@ -4369,16 +4572,11 @@ static int patch_stac9200(struct hda_codec *codec) spec->board_config = snd_hda_check_board_config(codec, STAC_9200_MODELS, stac9200_models, stac9200_cfg_tbl); - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9200, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac9200_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; @@ -4390,7 +4588,8 @@ static int patch_stac9200(struct hda_codec *codec) spec->num_adcs = 1; spec->num_pwrs = 0; - if (spec->board_config == STAC_9200_GATEWAY || + if (spec->board_config == STAC_9200_M4 || + spec->board_config == STAC_9200_M4_2 || spec->board_config == STAC_9200_OQO) spec->init = stac9200_eapd_init; else @@ -4408,6 +4607,12 @@ static int patch_stac9200(struct hda_codec *codec) return err; } + /* CF-74 has no headphone detection, and the driver should *NOT* + * do detection and HP/speaker toggle because the hardware does it. + */ + if (spec->board_config == STAC_9200_PANASONIC) + spec->hp_detect = 0; + codec->patch_ops = stac92xx_patch_ops; return 0; @@ -4425,21 +4630,26 @@ static int patch_stac925x(struct hda_codec *codec) codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac925x_pin_nids); spec->pin_nids = stac925x_pin_nids; - spec->board_config = snd_hda_check_board_config(codec, STAC_925x_MODELS, + + /* Check first for codec ID */ + spec->board_config = snd_hda_check_board_codec_sid_config(codec, + STAC_925x_MODELS, + stac925x_models, + stac925x_codec_id_cfg_tbl); + + /* Now checks for PCI ID, if codec ID is not found */ + if (spec->board_config < 0) + spec->board_config = snd_hda_check_board_config(codec, + STAC_925x_MODELS, stac925x_models, stac925x_cfg_tbl); again: - if (spec->board_config < 0) { - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC925x," "using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac925x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->multiout.max_channels = 2; spec->multiout.num_dacs = 1; @@ -4517,17 +4727,12 @@ static int patch_stac92hd73xx(struct hda_codec *codec) stac92hd73xx_models, stac92hd73xx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD73XX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd73xx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } num_dacs = snd_hda_get_connections(codec, 0x0a, conn, STAC92HD73_DAC_COUNT + 2) - 1; @@ -4541,14 +4746,18 @@ again: case 0x3: /* 6 Channel */ spec->mixer = stac92hd73xx_6ch_mixer; spec->init = stac92hd73xx_6ch_core_init; + spec->aloopback_ctl = stac92hd73xx_6ch_loopback; break; case 0x4: /* 8 Channel */ spec->mixer = stac92hd73xx_8ch_mixer; spec->init = stac92hd73xx_8ch_core_init; + spec->aloopback_ctl = stac92hd73xx_8ch_loopback; break; case 0x5: /* 10 Channel */ spec->mixer = stac92hd73xx_10ch_mixer; spec->init = stac92hd73xx_10ch_core_init; + spec->aloopback_ctl = stac92hd73xx_10ch_loopback; + break; } spec->multiout.dac_nids = spec->dac_nids; @@ -4587,18 +4796,18 @@ again: spec->init = dell_m6_core_init; switch (spec->board_config) { case STAC_DELL_M6_AMIC: /* Analog Mics */ - stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); + snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); spec->num_dmics = 0; spec->private_dimux.num_items = 1; break; case STAC_DELL_M6_DMIC: /* Digital Mics */ - stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; spec->private_dimux.num_items = 2; break; case STAC_DELL_M6_BOTH: /* Both */ - stac92xx_set_config_reg(codec, 0x0b, 0x90A70170); - stac92xx_set_config_reg(codec, 0x13, 0x90A60160); + snd_hda_codec_set_pincfg(codec, 0x0b, 0x90A70170); + snd_hda_codec_set_pincfg(codec, 0x13, 0x90A60160); spec->num_dmics = 1; spec->private_dimux.num_items = 2; break; @@ -4658,7 +4867,10 @@ static struct hda_input_mux stac92hd83xxx_dmux = { static int patch_stac92hd83xxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + hda_nid_t conn[STAC92HD83_DAC_COUNT + 1]; int err; + int num_dacs; + hda_nid_t nid; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4672,23 +4884,17 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) spec->dmux_nids = stac92hd83xxx_dmux_nids; spec->adc_nids = stac92hd83xxx_adc_nids; spec->pwr_nids = stac92hd83xxx_pwr_nids; + spec->amp_nids = stac92hd83xxx_amp_nids; spec->pwr_mapping = stac92hd83xxx_pwr_mapping; spec->num_pwrs = ARRAY_SIZE(stac92hd83xxx_pwr_nids); spec->multiout.dac_nids = spec->dac_nids; spec->init = stac92hd83xxx_core_init; - switch (codec->vendor_id) { - case 0x111d7605: - break; - default: - spec->num_pwrs--; - spec->init++; /* switch to config #2 */ - } - spec->mixer = stac92hd83xxx_mixer; spec->num_pins = ARRAY_SIZE(stac92hd83xxx_pin_nids); spec->num_dmuxes = ARRAY_SIZE(stac92hd83xxx_dmux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd83xxx_adc_nids); + spec->num_amps = ARRAY_SIZE(stac92hd83xxx_amp_nids); spec->num_dmics = STAC92HD83XXX_NUM_DMICS; spec->dinput_mux = &stac92hd83xxx_dmux; spec->pin_nids = stac92hd83xxx_pin_nids; @@ -4697,16 +4903,21 @@ static int patch_stac92hd83xxx(struct hda_codec *codec) stac92hd83xxx_models, stac92hd83xxx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD83XXX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd83xxx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; + + switch (codec->vendor_id) { + case 0x111d7604: + case 0x111d7605: + case 0x111d76d5: + if (spec->board_config == STAC_92HD83XXX_PWR_REF) + break; + spec->num_pwrs = 0; + break; } err = stac92xx_parse_auto_config(codec, 0x1d, 0); @@ -4725,6 +4936,23 @@ again: return err; } + switch (spec->board_config) { + case STAC_DELL_S14: + nid = 0xf; + break; + default: + nid = 0xe; + break; + } + + num_dacs = snd_hda_get_connections(codec, nid, + conn, STAC92HD83_DAC_COUNT + 1) - 1; + + /* set port X to select the last DAC + */ + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_CONNECT_SEL, num_dacs); + codec->patch_ops = stac92xx_patch_ops; codec->proc_widget_hook = stac92hd_proc_hook; @@ -4732,7 +4960,16 @@ again: return 0; } -static struct hda_input_mux stac92hd71bxx_dmux = { +static struct hda_input_mux stac92hd71bxx_dmux_nomixer = { + .num_items = 3, + .items = { + { "Analog Inputs", 0x00 }, + { "Digital Mic 1", 0x02 }, + { "Digital Mic 2", 0x03 }, + } +}; + +static struct hda_input_mux stac92hd71bxx_dmux_amixer = { .num_items = 4, .items = { { "Analog Inputs", 0x00 }, @@ -4742,10 +4979,67 @@ static struct hda_input_mux stac92hd71bxx_dmux = { } }; +/* get the pin connection (fixed, none, etc) */ +static unsigned int stac_get_defcfg_connect(struct hda_codec *codec, int idx) +{ + struct sigmatel_spec *spec = codec->spec; + unsigned int cfg; + + cfg = snd_hda_codec_get_pincfg(codec, spec->pin_nids[idx]); + return get_defcfg_connect(cfg); +} + +static int stac92hd71bxx_connected_ports(struct hda_codec *codec, + hda_nid_t *nids, int num_nids) +{ + struct sigmatel_spec *spec = codec->spec; + int idx, num; + unsigned int def_conf; + + for (num = 0; num < num_nids; num++) { + for (idx = 0; idx < spec->num_pins; idx++) + if (spec->pin_nids[idx] == nids[num]) + break; + if (idx >= spec->num_pins) + break; + def_conf = stac_get_defcfg_connect(codec, idx); + if (def_conf == AC_JACK_PORT_NONE) + break; + } + return num; +} + +static int stac92hd71bxx_connected_smuxes(struct hda_codec *codec, + hda_nid_t dig0pin) +{ + struct sigmatel_spec *spec = codec->spec; + int idx; + + for (idx = 0; idx < spec->num_pins; idx++) + if (spec->pin_nids[idx] == dig0pin) + break; + if ((idx + 2) >= spec->num_pins) + return 0; + + /* dig1pin case */ + if (stac_get_defcfg_connect(codec, idx + 1) != AC_JACK_PORT_NONE) + return 2; + + /* dig0pin + dig2pin case */ + if (stac_get_defcfg_connect(codec, idx + 2) != AC_JACK_PORT_NONE) + return 2; + if (stac_get_defcfg_connect(codec, idx) != AC_JACK_PORT_NONE) + return 1; + else + return 0; +} + static int patch_stac92hd71bxx(struct hda_codec *codec) { struct sigmatel_spec *spec; + struct hda_verb *unmute_init = stac92hd71bxx_unmute_core_init; int err = 0; + unsigned int ndmic_nids = 0; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -4753,27 +5047,32 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; codec->patch_ops = stac92xx_patch_ops; - spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pins = STAC92HD71BXX_NUM_PINS; + switch (codec->vendor_id) { + case 0x111d76b6: + case 0x111d76b7: + spec->pin_nids = stac92hd71bxx_pin_nids_4port; + break; + case 0x111d7603: + case 0x111d7608: + /* On 92HD75Bx 0x27 isn't a pin nid */ + spec->num_pins--; + /* fallthrough */ + default: + spec->pin_nids = stac92hd71bxx_pin_nids_6port; + } spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); - spec->pin_nids = stac92hd71bxx_pin_nids; - memcpy(&spec->private_dimux, &stac92hd71bxx_dmux, - sizeof(stac92hd71bxx_dmux)); spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, stac92hd71bxx_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD71BXX, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac92hd71bxx_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } if (spec->board_config > STAC_92HD71BXX_REF) { /* GPIO0 = EAPD */ @@ -4782,16 +5081,34 @@ again: spec->gpio_data = 0x01; } + spec->dmic_nids = stac92hd71bxx_dmic_nids; + spec->dmux_nids = stac92hd71bxx_dmux_nids; + switch (codec->vendor_id) { case 0x111d76b6: /* 4 Port without Analog Mixer */ case 0x111d76b7: + unmute_init++; + /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_nomixer, + sizeof(stac92hd71bxx_dmux_nomixer)); spec->mixer = stac92hd71bxx_mixer; spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; + spec->num_dmics = stac92hd71bxx_connected_ports(codec, + stac92hd71bxx_dmic_nids, + STAC92HD71BXX_NUM_DMICS); + if (spec->num_dmics) { + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + spec->dinput_mux = &spec->private_dimux; + ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1; + } break; case 0x111d7608: /* 5 Port with Analog Mixer */ + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, + sizeof(stac92hd71bxx_dmux_amixer)); + spec->private_dimux.num_items--; switch (spec->board_config) { case STAC_HP_M4: /* Enable VREF power saving on GPIO1 detect */ @@ -4818,7 +5135,15 @@ again: /* disable VSW */ spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF]; - stac_change_pin_config(codec, 0xf, 0x40f000f0); + unmute_init++; + snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); + snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); + stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS - 1] = 0; + spec->num_dmics = stac92hd71bxx_connected_ports(codec, + stac92hd71bxx_dmic_nids, + STAC92HD71BXX_NUM_DMICS - 1); + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 2; break; case 0x111d7603: /* 6 Port with Analog Mixer */ if ((codec->revision_id & 0xf) == 1) @@ -4828,12 +5153,23 @@ again: spec->num_pwrs = 0; /* fallthru */ default: + memcpy(&spec->private_dimux, &stac92hd71bxx_dmux_amixer, + sizeof(stac92hd71bxx_dmux_amixer)); spec->dinput_mux = &spec->private_dimux; spec->mixer = stac92hd71bxx_analog_mixer; spec->init = stac92hd71bxx_analog_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; + spec->num_dmics = stac92hd71bxx_connected_ports(codec, + stac92hd71bxx_dmic_nids, + STAC92HD71BXX_NUM_DMICS); + spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); + ndmic_nids = ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1; } + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) + snd_hda_sequence_write_cache(codec, unmute_init); + + spec->aloopback_ctl = stac92hd71bxx_loopback; spec->aloopback_mask = 0x50; spec->aloopback_shift = 0; @@ -4841,18 +5177,17 @@ again: spec->digbeep_nid = 0x26; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; - spec->dmic_nids = stac92hd71bxx_dmic_nids; - spec->dmux_nids = stac92hd71bxx_dmux_nids; spec->smux_nids = stac92hd71bxx_smux_nids; spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); + spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); switch (spec->board_config) { case STAC_HP_M4: /* enable internal microphone */ - stac_change_pin_config(codec, 0x0e, 0x01813040); + snd_hda_codec_set_pincfg(codec, 0x0e, 0x01813040); stac92xx_auto_set_pinctl(codec, 0x0e, AC_PINCTL_IN_EN | AC_PINCTL_VREF_80); /* fallthru */ @@ -4865,21 +5200,38 @@ again: case STAC_DELL_M4_3: spec->num_dmics = 1; spec->num_smuxes = 0; - spec->num_dmuxes = 0; + spec->num_dmuxes = 1; + break; + case STAC_HP_DV5: + snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); + stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); + break; + case STAC_HP_HDX: + spec->num_dmics = 1; + spec->num_dmuxes = 1; + spec->num_smuxes = 1; + /* + * For controlling MUTE LED on HP HDX16/HDX18 notebooks, + * the CONFIG_SND_HDA_POWER_SAVE is needed to be set. + */ +#ifdef CONFIG_SND_HDA_POWER_SAVE + /* orange/white mute led on GPIO3, orange=0, white=1 */ + spec->gpio_mask |= 0x08; + spec->gpio_dir |= 0x08; + spec->gpio_data |= 0x08; /* set to white */ + + /* register check_power_status callback. */ + codec->patch_ops.check_power_status = + stac92xx_hp_hdx_check_power_status; +#endif break; - default: - spec->num_dmics = STAC92HD71BXX_NUM_DMICS; - spec->num_smuxes = ARRAY_SIZE(stac92hd71bxx_smux_nids); - spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); }; spec->multiout.dac_nids = spec->dac_nids; if (spec->dinput_mux) - spec->private_dimux.num_items += - spec->num_dmics - - (ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1); + spec->private_dimux.num_items += spec->num_dmics - ndmic_nids; - err = stac92xx_parse_auto_config(codec, 0x21, 0x23); + err = stac92xx_parse_auto_config(codec, 0x21, 0); if (!err) { if (spec->board_config < 0) { printk(KERN_WARNING "hda_codec: No auto-config is " @@ -4954,17 +5306,12 @@ static int patch_stac922x(struct hda_codec *codec) } again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC922x, " "using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac922x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->adc_nids = stac922x_adc_nids; spec->mux_nids = stac922x_mux_nids; @@ -5015,24 +5362,19 @@ static int patch_stac927x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + codec->slave_dig_outs = stac927x_slave_dig_outs; spec->num_pins = ARRAY_SIZE(stac927x_pin_nids); spec->pin_nids = stac927x_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_927X_MODELS, stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0 || !stac927x_brd_tbl[spec->board_config]) { - if (spec->board_config < 0) - snd_printdd(KERN_INFO "hda_codec: Unknown model for" - "STAC927x, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + "STAC927x, using BIOS defaults\n"); + else + stac92xx_set_config_regs(codec, stac927x_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->digbeep_nid = 0x23; spec->adc_nids = stac927x_adc_nids; @@ -5061,15 +5403,15 @@ static int patch_stac927x(struct hda_codec *codec) case 0x10280209: case 0x1028022e: /* correct the device field to SPDIF out */ - stac_change_pin_config(codec, 0x21, 0x01442070); + snd_hda_codec_set_pincfg(codec, 0x21, 0x01442070); break; }; /* configure the analog microphone on some laptops */ - stac_change_pin_config(codec, 0x0c, 0x90a79130); + snd_hda_codec_set_pincfg(codec, 0x0c, 0x90a79130); /* correct the front output jack as a hp out */ - stac_change_pin_config(codec, 0x0f, 0x0227011f); + snd_hda_codec_set_pincfg(codec, 0x0f, 0x0227011f); /* correct the front input jack as a mic */ - stac_change_pin_config(codec, 0x0e, 0x02a79130); + snd_hda_codec_set_pincfg(codec, 0x0e, 0x02a79130); /* fallthru */ case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ @@ -5096,6 +5438,7 @@ static int patch_stac927x(struct hda_codec *codec) } spec->num_pwrs = 0; + spec->aloopback_ctl = stac927x_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; spec->eapd_switch = 1; @@ -5154,16 +5497,11 @@ static int patch_stac9205(struct hda_codec *codec) stac9205_models, stac9205_cfg_tbl); again: - if (spec->board_config < 0) { + if (spec->board_config < 0) snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9205, using BIOS defaults\n"); - err = stac92xx_save_bios_config_regs(codec); - } else - err = stac_save_pin_cfgs(codec, + else + stac92xx_set_config_regs(codec, stac9205_brd_tbl[spec->board_config]); - if (err < 0) { - stac92xx_free(codec); - return err; - } spec->digbeep_nid = 0x23; spec->adc_nids = stac9205_adc_nids; @@ -5180,17 +5518,20 @@ static int patch_stac9205(struct hda_codec *codec) spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; + spec->aloopback_ctl = stac9205_loopback; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; - spec->eapd_switch = 1; + /* Turn on/off EAPD per HP plugging */ + if (spec->board_config != STAC_9205_EAPD) + spec->eapd_switch = 1; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config){ case STAC_9205_DELL_M43: /* Enable SPDIF in/out */ - stac_change_pin_config(codec, 0x1f, 0x01441030); - stac_change_pin_config(codec, 0x20, 0x1c410030); + snd_hda_codec_set_pincfg(codec, 0x1f, 0x01441030); + snd_hda_codec_set_pincfg(codec, 0x20, 0x1c410030); /* Enable unsol response for GPIO4/Dock HP connection */ err = stac_add_event(spec, codec->afg, STAC_VREF_EVENT, 0x01); @@ -5247,223 +5588,87 @@ static int patch_stac9205(struct hda_codec *codec) * STAC9872 hack */ -/* static config for Sony VAIO FE550G and Sony VAIO AR */ -static hda_nid_t vaio_dacs[] = { 0x2 }; -#define VAIO_HP_DAC 0x5 -static hda_nid_t vaio_adcs[] = { 0x8 /*,0x6*/ }; -static hda_nid_t vaio_mux_nids[] = { 0x15 }; - -static struct hda_input_mux vaio_mux = { - .num_items = 3, - .items = { - /* { "HP", 0x0 }, */ - { "Mic Jack", 0x1 }, - { "Internal Mic", 0x2 }, - { "PCM", 0x3 }, - } -}; - -static struct hda_verb vaio_init[] = { - {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ - {0x0a, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | STAC_HP_EVENT}, - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ - {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ - {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ +static struct hda_verb stac9872_core_init[] = { {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ {} }; -static struct hda_verb vaio_ar_init[] = { - {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ - {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ - {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ - {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ -/* {0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT },*/ /* Optical Out */ - {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ - {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ - {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ - {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ -/* {0x10, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE},*/ /* Optical Out */ - {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ - {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ - {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ - {} -}; - -static struct snd_kcontrol_new vaio_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ +static struct snd_kcontrol_new stac9872_mixer[] = { HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = stac92xx_mux_enum_info, - .get = stac92xx_mux_enum_get, - .put = stac92xx_mux_enum_put, - }, - {} + { } /* end */ }; -static struct snd_kcontrol_new vaio_ar_mixer[] = { - HDA_CODEC_VOLUME("Headphone Playback Volume", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Headphone Playback Switch", 0x02, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Speaker Playback Volume", 0x05, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Speaker Playback Switch", 0x05, 0, HDA_OUTPUT), - /* HDA_CODEC_VOLUME("CD Capture Volume", 0x07, 0, HDA_INPUT), */ - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0, HDA_INPUT), - /*HDA_CODEC_MUTE("Optical Out Switch", 0x10, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Optical Out Volume", 0x10, 0, HDA_OUTPUT),*/ - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .count = 1, - .info = stac92xx_mux_enum_info, - .get = stac92xx_mux_enum_get, - .put = stac92xx_mux_enum_put, - }, - {} +static hda_nid_t stac9872_pin_nids[] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, + 0x11, 0x13, 0x14, }; -static struct hda_codec_ops stac9872_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac92xx_init, - .free = stac92xx_free, -#ifdef SND_HDA_NEEDS_RESUME - .resume = stac92xx_resume, -#endif +static hda_nid_t stac9872_adc_nids[] = { + 0x8 /*,0x6*/ }; -static int stac9872_vaio_init(struct hda_codec *codec) -{ - int err; - - err = stac92xx_init(codec); - if (err < 0) - return err; - if (codec->patch_ops.unsol_event) - codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); - return 0; -} - -static void stac9872_vaio_hp_detect(struct hda_codec *codec, unsigned int res) -{ - if (get_pin_presence(codec, 0x0a)) { - stac92xx_reset_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); - stac92xx_set_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); - } else { - stac92xx_reset_pinctl(codec, 0x0a, AC_PINCTL_OUT_EN); - stac92xx_set_pinctl(codec, 0x0f, AC_PINCTL_OUT_EN); - } -} - -static void stac9872_vaio_unsol_event(struct hda_codec *codec, unsigned int res) -{ - switch (res >> 26) { - case STAC_HP_EVENT: - stac9872_vaio_hp_detect(codec, res); - break; - } -} - -static struct hda_codec_ops stac9872_vaio_patch_ops = { - .build_controls = stac92xx_build_controls, - .build_pcms = stac92xx_build_pcms, - .init = stac9872_vaio_init, - .free = stac92xx_free, - .unsol_event = stac9872_vaio_unsol_event, -#ifdef CONFIG_PM - .resume = stac92xx_resume, -#endif +static hda_nid_t stac9872_mux_nids[] = { + 0x15 }; -enum { /* FE and SZ series. id=0x83847661 and subsys=0x104D0700 or 104D1000. */ - CXD9872RD_VAIO, - /* Unknown. id=0x83847662 and subsys=0x104D1200 or 104D1000. */ - STAC9872AK_VAIO, - /* Unknown. id=0x83847661 and subsys=0x104D1200. */ - STAC9872K_VAIO, - /* AR Series. id=0x83847664 and subsys=104D1300 */ - CXD9872AKD_VAIO, - STAC_9872_MODELS, +static unsigned int stac9872_vaio_pin_configs[9] = { + 0x03211020, 0x411111f0, 0x411111f0, 0x03a15030, + 0x411111f0, 0x90170110, 0x411111f0, 0x411111f0, + 0x90a7013e }; static const char *stac9872_models[STAC_9872_MODELS] = { - [CXD9872RD_VAIO] = "vaio", - [CXD9872AKD_VAIO] = "vaio-ar", + [STAC_9872_AUTO] = "auto", + [STAC_9872_VAIO] = "vaio", +}; + +static unsigned int *stac9872_brd_tbl[STAC_9872_MODELS] = { + [STAC_9872_VAIO] = stac9872_vaio_pin_configs, }; static struct snd_pci_quirk stac9872_cfg_tbl[] = { - SND_PCI_QUIRK(0x104d, 0x81e6, "Sony VAIO F/S", CXD9872RD_VAIO), - SND_PCI_QUIRK(0x104d, 0x81ef, "Sony VAIO F/S", CXD9872RD_VAIO), - SND_PCI_QUIRK(0x104d, 0x81fd, "Sony VAIO AR", CXD9872AKD_VAIO), - SND_PCI_QUIRK(0x104d, 0x8205, "Sony VAIO AR", CXD9872AKD_VAIO), - {} + {} /* terminator */ }; static int patch_stac9872(struct hda_codec *codec) { struct sigmatel_spec *spec; - int board_config; + int err; - board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, - stac9872_models, - stac9872_cfg_tbl); - if (board_config < 0) - /* unknown config, let generic-parser do its job... */ - return snd_hda_parse_generic_codec(codec); - spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) return -ENOMEM; - codec->spec = spec; - switch (board_config) { - case CXD9872RD_VAIO: - case STAC9872AK_VAIO: - case STAC9872K_VAIO: - spec->mixer = vaio_mixer; - spec->init = vaio_init; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); - spec->multiout.dac_nids = vaio_dacs; - spec->multiout.hp_nid = VAIO_HP_DAC; - spec->num_adcs = ARRAY_SIZE(vaio_adcs); - spec->adc_nids = vaio_adcs; - spec->num_pwrs = 0; - spec->input_mux = &vaio_mux; - spec->mux_nids = vaio_mux_nids; - codec->patch_ops = stac9872_vaio_patch_ops; - break; - - case CXD9872AKD_VAIO: - spec->mixer = vaio_ar_mixer; - spec->init = vaio_ar_init; - spec->multiout.max_channels = 2; - spec->multiout.num_dacs = ARRAY_SIZE(vaio_dacs); - spec->multiout.dac_nids = vaio_dacs; - spec->multiout.hp_nid = VAIO_HP_DAC; - spec->num_adcs = ARRAY_SIZE(vaio_adcs); - spec->num_pwrs = 0; - spec->adc_nids = vaio_adcs; - spec->input_mux = &vaio_mux; - spec->mux_nids = vaio_mux_nids; - codec->patch_ops = stac9872_patch_ops; - break; - } + spec->board_config = snd_hda_check_board_config(codec, STAC_9872_MODELS, + stac9872_models, + stac9872_cfg_tbl); + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC9872, " + "using BIOS defaults\n"); + else + stac92xx_set_config_regs(codec, + stac9872_brd_tbl[spec->board_config]); + + spec->num_pins = ARRAY_SIZE(stac9872_pin_nids); + spec->pin_nids = stac9872_pin_nids; + spec->multiout.dac_nids = spec->dac_nids; + spec->num_adcs = ARRAY_SIZE(stac9872_adc_nids); + spec->adc_nids = stac9872_adc_nids; + spec->num_muxes = ARRAY_SIZE(stac9872_mux_nids); + spec->mux_nids = stac9872_mux_nids; + spec->mixer = stac9872_mixer; + spec->init = stac9872_core_init; + + err = stac92xx_parse_auto_config(codec, 0x10, 0x12); + if (err < 0) { + stac92xx_free(codec); + return -EINVAL; + } + spec->input_mux = &spec->private_imux; + codec->patch_ops = stac92xx_patch_ops; return 0; } @@ -5521,6 +5726,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx}, + { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx}, { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c761394..b25a5cc 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1308,16 +1308,13 @@ static void vt1708_set_pinconfig_connect(struct hda_codec *codec, hda_nid_t nid) unsigned int def_conf; unsigned char seqassoc; - def_conf = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = snd_hda_codec_get_pincfg(codec, nid); seqassoc = (unsigned char) get_defcfg_association(def_conf); seqassoc = (seqassoc << 4) | get_defcfg_sequence(def_conf); if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) { if (seqassoc == 0xff) { def_conf = def_conf & (~(AC_JACK_PORT_BOTH << 30)); - snd_hda_codec_write(codec, nid, 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - def_conf >> 24); + snd_hda_codec_set_pincfg(codec, nid, def_conf); } } @@ -1354,7 +1351,7 @@ static int vt1708_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708_DIGIN_NID; @@ -1827,7 +1824,7 @@ static int vt1709_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1709_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1709_DIGIN_NID; @@ -2371,7 +2368,7 @@ static int vt1708B_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708B_DIGOUT_NID; if (spec->autocfg.dig_in_pin) spec->dig_in_nid = VT1708B_DIGIN_NID; @@ -2836,7 +2833,7 @@ static int vt1708S_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID; spec->extra_dig_out_nid = 0x15; @@ -3155,7 +3152,7 @@ static int vt1702_parse_auto_config(struct hda_codec *codec) spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + if (spec->autocfg.dig_outs) spec->multiout.dig_out_nid = VT1702_DIGOUT_NID; spec->extra_dig_out_nid = 0x1B; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index bab1c70..0d0cdbd 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -458,7 +458,7 @@ static irqreturn_t snd_ice1712_interrupt(int irq, void *dev_id) u16 pbkstatus; struct snd_pcm_substream *substream; pbkstatus = inw(ICEDS(ice, INTSTAT)); - /* printk("pbkstatus = 0x%x\n", pbkstatus); */ + /* printk(KERN_DEBUG "pbkstatus = 0x%x\n", pbkstatus); */ for (idx = 0; idx < 6; idx++) { if ((pbkstatus & (3 << (idx * 2))) == 0) continue; @@ -2533,8 +2533,8 @@ static int __devinit snd_ice1712_create(struct snd_card *card, if (err < 0) return err; /* check, if we can restrict PCI DMA transfers to 28 bits */ - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index 7ff36d3..128510e7 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -241,6 +241,8 @@ get_rawmidi_substream(struct snd_ice1712 *ice, unsigned int stream) struct snd_rawmidi_substream, list); } +static void enable_midi_irq(struct snd_ice1712 *ice, u8 flag, int enable); + static void vt1724_midi_write(struct snd_ice1712 *ice) { struct snd_rawmidi_substream *s; @@ -254,6 +256,11 @@ static void vt1724_midi_write(struct snd_ice1712 *ice) for (i = 0; i < count; ++i) outb(buffer[i], ICEREG1724(ice, MPU_DATA)); } + /* mask irq when all bytes have been transmitted. + * enabled again in output_trigger when the new data comes in. + */ + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, + !snd_rawmidi_transmit_empty(s)); } static void vt1724_midi_read(struct snd_ice1712 *ice) @@ -272,31 +279,34 @@ static void vt1724_midi_read(struct snd_ice1712 *ice) } } -static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream, - u8 flag, int enable) +/* call with ice->reg_lock */ +static void enable_midi_irq(struct snd_ice1712 *ice, u8 flag, int enable) { - struct snd_ice1712 *ice = substream->rmidi->private_data; - u8 mask; - - spin_lock_irq(&ice->reg_lock); - mask = inb(ICEREG1724(ice, IRQMASK)); + u8 mask = inb(ICEREG1724(ice, IRQMASK)); if (enable) mask &= ~flag; else mask |= flag; outb(mask, ICEREG1724(ice, IRQMASK)); +} + +static void vt1724_enable_midi_irq(struct snd_rawmidi_substream *substream, + u8 flag, int enable) +{ + struct snd_ice1712 *ice = substream->rmidi->private_data; + + spin_lock_irq(&ice->reg_lock); + enable_midi_irq(ice, flag, enable); spin_unlock_irq(&ice->reg_lock); } static int vt1724_midi_output_open(struct snd_rawmidi_substream *s) { - vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 1); return 0; } static int vt1724_midi_output_close(struct snd_rawmidi_substream *s) { - vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0); return 0; } @@ -311,6 +321,7 @@ static void vt1724_midi_output_trigger(struct snd_rawmidi_substream *s, int up) vt1724_midi_write(ice); } else { ice->midi_output = 0; + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0); } spin_unlock_irqrestore(&ice->reg_lock, flags); } @@ -320,6 +331,7 @@ static void vt1724_midi_output_drain(struct snd_rawmidi_substream *s) struct snd_ice1712 *ice = s->rmidi->private_data; unsigned long timeout; + vt1724_enable_midi_irq(s, VT1724_IRQ_MPU_TX, 0); /* 32 bytes should be transmitted in less than about 12 ms */ timeout = jiffies + msecs_to_jiffies(15); do { @@ -389,24 +401,24 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) status &= status_mask; if (status == 0) break; + spin_lock(&ice->reg_lock); if (++timeout > 10) { status = inb(ICEREG1724(ice, IRQSTAT)); printk(KERN_ERR "ice1724: Too long irq loop, " "status = 0x%x\n", status); if (status & VT1724_IRQ_MPU_TX) { printk(KERN_ERR "ice1724: Disabling MPU_TX\n"); - outb(inb(ICEREG1724(ice, IRQMASK)) | - VT1724_IRQ_MPU_TX, - ICEREG1724(ice, IRQMASK)); + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0); } + spin_unlock(&ice->reg_lock); break; } handled = 1; if (status & VT1724_IRQ_MPU_TX) { - spin_lock(&ice->reg_lock); if (ice->midi_output) vt1724_midi_write(ice); - spin_unlock(&ice->reg_lock); + else + enable_midi_irq(ice, VT1724_IRQ_MPU_TX, 0); /* Due to mysterical reasons, MPU_TX is always * generated (and can't be cleared) when a PCM * playback is going. So let's ignore at the @@ -415,15 +427,14 @@ static irqreturn_t snd_vt1724_interrupt(int irq, void *dev_id) status_mask &= ~VT1724_IRQ_MPU_TX; } if (status & VT1724_IRQ_MPU_RX) { - spin_lock(&ice->reg_lock); if (ice->midi_input) vt1724_midi_read(ice); else vt1724_midi_clear_rx(ice); - spin_unlock(&ice->reg_lock); } /* ack MPU irq */ outb(status, ICEREG1724(ice, IRQSTAT)); + spin_unlock(&ice->reg_lock); if (status & VT1724_IRQ_MTPCM) { /* * Multi-track PCM @@ -745,7 +756,14 @@ static int snd_vt1724_playback_pro_prepare(struct snd_pcm_substream *substream) spin_unlock_irq(&ice->reg_lock); - /* printk("pro prepare: ch = %d, addr = 0x%x, buffer = 0x%x, period = 0x%x\n", substream->runtime->channels, (unsigned int)substream->runtime->dma_addr, snd_pcm_lib_buffer_bytes(substream), snd_pcm_lib_period_bytes(substream)); */ + /* + printk(KERN_DEBUG "pro prepare: ch = %d, addr = 0x%x, " + "buffer = 0x%x, period = 0x%x\n", + substream->runtime->channels, + (unsigned int)substream->runtime->dma_addr, + snd_pcm_lib_buffer_bytes(substream), + snd_pcm_lib_period_bytes(substream)); + */ return 0; } @@ -2122,7 +2140,9 @@ unsigned char snd_vt1724_read_i2c(struct snd_ice1712 *ice, wait_i2c_busy(ice); val = inb(ICEREG1724(ice, I2C_DATA)); mutex_unlock(&ice->i2c_mutex); - /* printk("i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); */ + /* + printk(KERN_DEBUG "i2c_read: [0x%x,0x%x] = 0x%x\n", dev, addr, val); + */ return val; } @@ -2131,7 +2151,9 @@ void snd_vt1724_write_i2c(struct snd_ice1712 *ice, { mutex_lock(&ice->i2c_mutex); wait_i2c_busy(ice); - /* printk("i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); */ + /* + printk(KERN_DEBUG "i2c_write: [0x%x,0x%x] = 0x%x\n", dev, addr, data); + */ outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR)); outb(data, ICEREG1724(ice, I2C_DATA)); outb(dev | VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR)); diff --git a/sound/pci/ice1712/juli.c b/sound/pci/ice1712/juli.c index c51659b..fd948bf 100644 --- a/sound/pci/ice1712/juli.c +++ b/sound/pci/ice1712/juli.c @@ -345,8 +345,9 @@ static int juli_mute_put(struct snd_kcontrol *kcontrol, new_gpio = old_gpio & ~((unsigned int) kcontrol->private_value); } - /* printk("JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, \ - new_gpio 0x%x\n", + /* printk(KERN_DEBUG + "JULI - mute/unmute: control_value: 0x%x, old_gpio: 0x%x, " + "new_gpio 0x%x\n", (unsigned int)ucontrol->value.integer.value[0], old_gpio, new_gpio); */ if (old_gpio != new_gpio) { diff --git a/sound/pci/ice1712/prodigy192.c b/sound/pci/ice1712/prodigy192.c index 48d3679..2a8e5cd 100644 --- a/sound/pci/ice1712/prodigy192.c +++ b/sound/pci/ice1712/prodigy192.c @@ -133,8 +133,10 @@ static int stac9460_dac_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id) + STAC946X_LF_VOLUME; /* due to possible conflicts with stac9460_set_rate_val, mutexing */ mutex_lock(&spec->mute_mutex); - /*printk("Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx, - ucontrol->value.integer.value[0]);*/ + /* + printk(KERN_DEBUG "Mute put: reg 0x%02x, ctrl value: 0x%02x\n", idx, + ucontrol->value.integer.value[0]); + */ change = stac9460_dac_mute(ice, idx, ucontrol->value.integer.value[0]); mutex_unlock(&spec->mute_mutex); return change; @@ -185,7 +187,10 @@ static int stac9460_dac_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_el change = (ovol != nvol); if (change) { ovol = (0x7f - nvol) | (tmp & 0x80); - /*printk("DAC Volume: reg 0x%02x: 0x%02x\n", idx, ovol);*/ + /* + printk(KERN_DEBUG "DAC Volume: reg 0x%02x: 0x%02x\n", + idx, ovol); + */ stac9460_put(ice, idx, (0x7f - nvol) | (tmp & 0x80)); } return change; @@ -344,7 +349,7 @@ static void stac9460_set_rate_val(struct snd_ice1712 *ice, unsigned int rate) for (idx = 0; idx < 7 ; ++idx) changed[idx] = stac9460_dac_mute(ice, STAC946X_MASTER_VOLUME + idx, 0); - /*printk("Rate change: %d, new MC: 0x%02x\n", rate, new);*/ + /*printk(KERN_DEBUG "Rate change: %d, new MC: 0x%02x\n", rate, new);*/ stac9460_put(ice, STAC946X_MASTER_CLOCKING, new); udelay(10); /* unmuting - only originally unmuted dacs - diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 671ff65..5764881 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -617,7 +617,7 @@ static int snd_intel8x0_ali_codec_semaphore(struct intel8x0 *chip) int time = 100; if (chip->buggy_semaphore) return 0; /* just ignore ... */ - while (time-- && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY)) + while (--time && (igetdword(chip, ICHREG(ALI_CAS)) & ALI_CAS_SEM_BUSY)) udelay(1); if (! time && ! chip->in_ac97_init) snd_printk(KERN_WARNING "ali_codec_semaphore timeout\n"); @@ -689,7 +689,7 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */ ichdev->fragsize >> ichdev->pos_shift); #if 0 - printk("bdbar[%i] = 0x%x [0x%x]\n", + printk(KERN_DEBUG "bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]); #endif } @@ -701,8 +701,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0 *chip, struct ichdev *ich ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags; ichdev->position = 0; #if 0 - printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n", - ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1); + printk(KERN_DEBUG "lvi_frag = %i, frags = %i, period_size = 0x%x, " + "period_size1 = 0x%x\n", + ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, + ichdev->fragsize1); #endif /* clear interrupts */ iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI); @@ -768,7 +770,8 @@ static inline void snd_intel8x0_update(struct intel8x0 *chip, struct ichdev *ich ichdev->lvi_frag %= ichdev->frags; ichdev->bdbar[ichdev->lvi * 2] = cpu_to_le32(ichdev->physbuf + ichdev->lvi_frag * ichdev->fragsize1); #if 0 - printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n", + printk(KERN_DEBUG "new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, " + "all = 0x%x, 0x%x\n", ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2], ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port), inl(port + 4), inb(port + ICH_REG_OFF_CR)); @@ -2287,23 +2290,23 @@ static void do_ali_reset(struct intel8x0 *chip) iputdword(chip, ICHREG(ALI_INTERRUPTSR), 0x00000000); } -static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) -{ - unsigned long end_time; - unsigned int cnt, status, nstatus; - - /* put logic to right state */ - /* first clear status bits */ - status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT; - if (chip->device_type == DEVICE_NFORCE) - status |= ICH_NVSPINT; - cnt = igetdword(chip, ICHREG(GLOB_STA)); - iputdword(chip, ICHREG(GLOB_STA), cnt & status); +#ifdef CONFIG_SND_AC97_POWER_SAVE +static struct snd_pci_quirk ich_chip_reset_mode[] = { + SND_PCI_QUIRK(0x1014, 0x051f, "Thinkpad R32", 1), + { } /* end */ +}; +static int snd_intel8x0_ich_chip_cold_reset(struct intel8x0 *chip) +{ + unsigned int cnt; /* ACLink on, 2 channels */ + + if (snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode)) + return -EIO; + cnt = igetdword(chip, ICHREG(GLOB_CNT)); cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK); -#ifdef CONFIG_SND_AC97_POWER_SAVE + /* do cold reset - the full ac97 powerdown may leave the controller * in a warm state but actually it cannot communicate with the codec. */ @@ -2312,22 +2315,58 @@ static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) udelay(10); iputdword(chip, ICHREG(GLOB_CNT), cnt | ICH_AC97COLD); msleep(1); + return 0; +} +#define snd_intel8x0_ich_chip_can_cold_reset(chip) \ + (!snd_pci_quirk_lookup(chip->pci, ich_chip_reset_mode)) #else +#define snd_intel8x0_ich_chip_cold_reset(chip) 0 +#define snd_intel8x0_ich_chip_can_cold_reset(chip) (0) +#endif + +static int snd_intel8x0_ich_chip_reset(struct intel8x0 *chip) +{ + unsigned long end_time; + unsigned int cnt; + /* ACLink on, 2 channels */ + cnt = igetdword(chip, ICHREG(GLOB_CNT)); + cnt &= ~(ICH_ACLINK | ICH_PCM_246_MASK); /* finish cold or do warm reset */ cnt |= (cnt & ICH_AC97COLD) == 0 ? ICH_AC97COLD : ICH_AC97WARM; iputdword(chip, ICHREG(GLOB_CNT), cnt); end_time = (jiffies + (HZ / 4)) + 1; do { if ((igetdword(chip, ICHREG(GLOB_CNT)) & ICH_AC97WARM) == 0) - goto __ok; + return 0; schedule_timeout_uninterruptible(1); } while (time_after_eq(end_time, jiffies)); snd_printk(KERN_ERR "AC'97 warm reset still in progress? [0x%x]\n", igetdword(chip, ICHREG(GLOB_CNT))); return -EIO; +} + +static int snd_intel8x0_ich_chip_init(struct intel8x0 *chip, int probing) +{ + unsigned long end_time; + unsigned int status, nstatus; + unsigned int cnt; + int err; + + /* put logic to right state */ + /* first clear status bits */ + status = ICH_RCS | ICH_MCINT | ICH_POINT | ICH_PIINT; + if (chip->device_type == DEVICE_NFORCE) + status |= ICH_NVSPINT; + cnt = igetdword(chip, ICHREG(GLOB_STA)); + iputdword(chip, ICHREG(GLOB_STA), cnt & status); + + if (snd_intel8x0_ich_chip_can_cold_reset(chip)) + err = snd_intel8x0_ich_chip_cold_reset(chip); + else + err = snd_intel8x0_ich_chip_reset(chip); + if (err < 0) + return err; - __ok: -#endif if (probing) { /* wait for any codec ready status. * Once it becomes ready it should remain ready diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 33a843c..6ec0fc5 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -411,7 +411,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic bdbar[idx + 0] = cpu_to_le32(ichdev->physbuf + (((idx >> 1) * ichdev->fragsize) % ichdev->size)); bdbar[idx + 1] = cpu_to_le32(0x80000000 | /* interrupt on completion */ ichdev->fragsize >> chip->pcm_pos_shift); - // printk("bdbar[%i] = 0x%x [0x%x]\n", idx + 0, bdbar[idx + 0], bdbar[idx + 1]); + /* + printk(KERN_DEBUG "bdbar[%i] = 0x%x [0x%x]\n", + idx + 0, bdbar[idx + 0], bdbar[idx + 1]); + */ } ichdev->frags = ichdev->size / ichdev->fragsize; } @@ -421,8 +424,10 @@ static void snd_intel8x0_setup_periods(struct intel8x0m *chip, struct ichdev *ic ichdev->lvi_frag = ICH_REG_LVI_MASK % ichdev->frags; ichdev->position = 0; #if 0 - printk("lvi_frag = %i, frags = %i, period_size = 0x%x, period_size1 = 0x%x\n", - ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, ichdev->fragsize1); + printk(KERN_DEBUG "lvi_frag = %i, frags = %i, period_size = 0x%x, " + "period_size1 = 0x%x\n", + ichdev->lvi_frag, ichdev->frags, ichdev->fragsize, + ichdev->fragsize1); #endif /* clear interrupts */ iputbyte(chip, port + ichdev->roff_sr, ICH_FIFOE | ICH_BCIS | ICH_LVBCI); @@ -465,7 +470,8 @@ static inline void snd_intel8x0_update(struct intel8x0m *chip, struct ichdev *ic ichdev->lvi_frag * ichdev->fragsize1); #if 0 - printk("new: bdbar[%i] = 0x%x [0x%x], prefetch = %i, all = 0x%x, 0x%x\n", + printk(KERN_DEBUG "new: bdbar[%i] = 0x%x [0x%x], " + "prefetch = %i, all = 0x%x, 0x%x\n", ichdev->lvi * 2, ichdev->bdbar[ichdev->lvi * 2], ichdev->bdbar[ichdev->lvi * 2 + 1], inb(ICH_REG_OFF_PIV + port), inl(port + 4), inb(port + ICH_REG_OFF_CR)); diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c index 7014154..75283fbb 100644 --- a/sound/pci/maestro3.c +++ b/sound/pci/maestro3.c @@ -2530,8 +2530,8 @@ snd_m3_create(struct snd_card *card, struct pci_dev *pci, return -EIO; /* check, if we can restrict PCI DMA transfers to 28 bits */ - if (pci_set_dma_mask(pci, DMA_28BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_28BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(28)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(28)) < 0) { snd_printk(KERN_ERR "architecture does not support 28bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index bfc19e3..82bc5b9e 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -607,6 +607,7 @@ static int snd_mixart_hw_params(struct snd_pcm_substream *subs, /* set the format to the board */ err = mixart_set_format(stream, format); if(err < 0) { + mutex_unlock(&mgr->setup_mutex); return err; } @@ -1290,7 +1291,7 @@ static int __devinit snd_mixart_probe(struct pci_dev *pci, pci_set_master(pci); /* check if we can restrict PCI DMA transfers to 32 bits */ - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { snd_printk(KERN_ERR "architecture does not support 32bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 3782b52..4cf4cd8 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -345,8 +345,8 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw status_daught = readl_be( MIXART_MEM( mgr,MIXART_PSEUDOREG_DXLX_STATUS_OFFSET )); /* motherboard xilinx status 5 will say that the board is performing a reset */ - if( status_xilinx == 5 ) { - snd_printk( KERN_ERR "miXart is resetting !\n"); + if (status_xilinx == 5) { + snd_printk(KERN_ERR "miXart is resetting !\n"); return -EAGAIN; /* try again later */ } @@ -354,13 +354,14 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw case MIXART_MOTHERBOARD_XLX_INDEX: /* xilinx already loaded ? */ - if( status_xilinx == 4 ) { - snd_printk( KERN_DEBUG "xilinx is already loaded !\n"); + if (status_xilinx == 4) { + snd_printk(KERN_DEBUG "xilinx is already loaded !\n"); return 0; } /* the status should be 0 == "idle" */ - if( status_xilinx != 0 ) { - snd_printk( KERN_ERR "xilinx load error ! status = %d\n", status_xilinx); + if (status_xilinx != 0) { + snd_printk(KERN_ERR "xilinx load error ! status = %d\n", + status_xilinx); return -EIO; /* modprob -r may help ? */ } @@ -389,21 +390,23 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw case MIXART_MOTHERBOARD_ELF_INDEX: - if( status_elf == 4 ) { - snd_printk( KERN_DEBUG "elf file already loaded !\n"); + if (status_elf == 4) { + snd_printk(KERN_DEBUG "elf file already loaded !\n"); return 0; } /* the status should be 0 == "idle" */ - if( status_elf != 0 ) { - snd_printk( KERN_ERR "elf load error ! status = %d\n", status_elf); + if (status_elf != 0) { + snd_printk(KERN_ERR "elf load error ! status = %d\n", + status_elf); return -EIO; /* modprob -r may help ? */ } /* wait for xilinx status == 4 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_MXLX_STATUS_OFFSET, 1, 4, 500); /* 5sec */ if (err < 0) { - snd_printk( KERN_ERR "xilinx was not loaded or could not be started\n"); + snd_printk(KERN_ERR "xilinx was not loaded or " + "could not be started\n"); return err; } @@ -424,7 +427,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for elf status == 4 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_ELF_STATUS_OFFSET, 1, 4, 300); /* 3sec */ if (err < 0) { - snd_printk( KERN_ERR "elf could not be started\n"); + snd_printk(KERN_ERR "elf could not be started\n"); return err; } @@ -437,15 +440,16 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw default: /* elf and xilinx should be loaded */ - if( (status_elf != 4) || (status_xilinx != 4) ) { - printk( KERN_ERR "xilinx or elf not successfully loaded\n"); + if (status_elf != 4 || status_xilinx != 4) { + printk(KERN_ERR "xilinx or elf not " + "successfully loaded\n"); return -EIO; /* modprob -r may help ? */ } /* wait for daughter detection != 0 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DBRD_PRESENCE_OFFSET, 0, 0, 30); /* 300msec */ if (err < 0) { - snd_printk( KERN_ERR "error starting elf file\n"); + snd_printk(KERN_ERR "error starting elf file\n"); return err; } @@ -460,8 +464,9 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw return -EINVAL; /* daughter should be idle */ - if( status_daught != 0 ) { - printk( KERN_ERR "daughter load error ! status = %d\n", status_daught); + if (status_daught != 0) { + printk(KERN_ERR "daughter load error ! status = %d\n", + status_daught); return -EIO; /* modprob -r may help ? */ } @@ -480,7 +485,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for status == 2 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 2, 30); /* 300msec */ if (err < 0) { - snd_printk( KERN_ERR "daughter board load error\n"); + snd_printk(KERN_ERR "daughter board load error\n"); return err; } @@ -502,7 +507,8 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* wait for daughter status == 3 */ err = mixart_wait_nice_for_register_value( mgr, MIXART_PSEUDOREG_DXLX_STATUS_OFFSET, 1, 3, 300); /* 3sec */ if (err < 0) { - snd_printk( KERN_ERR "daughter board could not be initialised\n"); + snd_printk(KERN_ERR + "daughter board could not be initialised\n"); return err; } @@ -512,7 +518,7 @@ static int mixart_dsp_load(struct mixart_mgr* mgr, int index, const struct firmw /* first communication with embedded */ err = mixart_first_init(mgr); if (err < 0) { - snd_printk( KERN_ERR "miXart could not be set up\n"); + snd_printk(KERN_ERR "miXart could not be set up\n"); return err; } @@ -581,16 +587,6 @@ MODULE_FIRMWARE("mixart/miXart8AES.xlx"); /* miXart hwdep interface id string */ #define SND_MIXART_HWDEP_ID "miXart Loader" -static int mixart_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int mixart_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - static int mixart_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { @@ -643,8 +639,6 @@ int snd_mixart_setup_firmware(struct mixart_mgr *mgr) hw->iface = SNDRV_HWDEP_IFACE_MIXART; hw->private_data = mgr; - hw->ops.open = mixart_hwdep_open; - hw->ops.release = mixart_hwdep_release; hw->ops.dsp_status = mixart_hwdep_dsp_status; hw->ops.dsp_load = mixart_hwdep_dsp_load; hw->exclusive = 1; diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 1ab833f..84ef131 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -45,6 +45,7 @@ MODULE_PARM_DESC(enable, "enable card"); static struct pci_device_id hifier_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x14c3, 0x1710) }, { OXYGEN_PCI_SUBID(0x14c3, 0x1711) }, + { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; MODULE_DEVICE_TABLE(pci, hifier_ids); @@ -151,7 +152,6 @@ static const struct oxygen_model model_hifier = { .shortname = "C-Media CMI8787", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", - .owner = THIS_MODULE, .init = hifier_init, .control_filter = hifier_control_filter, .cleanup = hifier_cleanup, @@ -173,6 +173,13 @@ static const struct oxygen_model model_hifier = { .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; +static int __devinit get_hifier_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + chip->model = model_hifier; + return 0; +} + static int __devinit hifier_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -185,7 +192,8 @@ static int __devinit hifier_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - err = oxygen_pci_probe(pci, index[dev], id[dev], &model_hifier, 0); + err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, + hifier_ids, get_hifier_model); if (err >= 0) ++dev; return err; diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index de999c6..72db4c3 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -1,5 +1,5 @@ /* - * C-Media CMI8788 driver for C-Media's reference design and for the X-Meridian + * C-Media CMI8788 driver for C-Media's reference design and similar models * * Copyright (c) Clemens Ladisch <clemens@ladisch.de> * @@ -26,6 +26,7 @@ * * GPIO 0 -> DFS0 of AK5385 * GPIO 1 -> DFS1 of AK5385 + * GPIO 8 -> enable headphone amplifier on HT-Omega models */ #include <linux/delay.h> @@ -61,7 +62,8 @@ MODULE_PARM_DESC(enable, "enable card"); enum { MODEL_CMEDIA_REF, /* C-Media's reference design */ MODEL_MERIDIAN, /* AuzenTech X-Meridian */ - MODEL_HALO, /* HT-Omega Claro halo */ + MODEL_CLARO, /* HT-Omega Claro */ + MODEL_CLARO_HALO, /* HT-Omega Claro halo */ }; static struct pci_device_id oxygen_ids[] __devinitdata = { @@ -74,8 +76,8 @@ static struct pci_device_id oxygen_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x147a, 0xa017), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x1a58, 0x0910), .driver_data = MODEL_CMEDIA_REF }, { OXYGEN_PCI_SUBID(0x415a, 0x5431), .driver_data = MODEL_MERIDIAN }, - { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CMEDIA_REF }, - { OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_HALO }, + { OXYGEN_PCI_SUBID(0x7284, 0x9761), .driver_data = MODEL_CLARO }, + { OXYGEN_PCI_SUBID(0x7284, 0x9781), .driver_data = MODEL_CLARO_HALO }, { } }; MODULE_DEVICE_TABLE(pci, oxygen_ids); @@ -86,6 +88,8 @@ MODULE_DEVICE_TABLE(pci, oxygen_ids); #define GPIO_AK5385_DFS_DOUBLE 0x0001 #define GPIO_AK5385_DFS_QUAD 0x0002 +#define GPIO_CLARO_HP 0x0100 + struct generic_data { u8 ak4396_ctl2; u16 saved_wm8785_registers[2]; @@ -196,10 +200,46 @@ static void meridian_init(struct oxygen *chip) ak5385_init(chip); } +static void claro_enable_hp(struct oxygen *chip) +{ + msleep(300); + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_CLARO_HP); + oxygen_set_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_HP); +} + +static void claro_init(struct oxygen *chip) +{ + ak4396_init(chip); + wm8785_init(chip); + claro_enable_hp(chip); +} + +static void claro_halo_init(struct oxygen *chip) +{ + ak4396_init(chip); + ak5385_init(chip); + claro_enable_hp(chip); +} + static void generic_cleanup(struct oxygen *chip) { } +static void claro_disable_hp(struct oxygen *chip) +{ + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, GPIO_CLARO_HP); +} + +static void claro_cleanup(struct oxygen *chip) +{ + claro_disable_hp(chip); +} + +static void claro_suspend(struct oxygen *chip) +{ + claro_disable_hp(chip); +} + static void generic_resume(struct oxygen *chip) { ak4396_registers_init(chip); @@ -211,6 +251,12 @@ static void meridian_resume(struct oxygen *chip) ak4396_registers_init(chip); } +static void claro_resume(struct oxygen *chip) +{ + ak4396_registers_init(chip); + claro_enable_hp(chip); +} + static void set_ak4396_params(struct oxygen *chip, struct snd_pcm_hw_params *params) { @@ -293,30 +339,10 @@ static void set_ak5385_params(struct oxygen *chip, static const DECLARE_TLV_DB_LINEAR(ak4396_db_scale, TLV_DB_GAIN_MUTE, 0); -static int generic_probe(struct oxygen *chip, unsigned long driver_data) -{ - if (driver_data == MODEL_MERIDIAN) { - chip->model.init = meridian_init; - chip->model.resume = meridian_resume; - chip->model.set_adc_params = set_ak5385_params; - chip->model.device_config = PLAYBACK_0_TO_I2S | - PLAYBACK_1_TO_SPDIF | - CAPTURE_0_FROM_I2S_2 | - CAPTURE_1_FROM_SPDIF; - } - if (driver_data == MODEL_MERIDIAN || driver_data == MODEL_HALO) { - chip->model.misc_flags = OXYGEN_MISC_MIDI; - chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT; - } - return 0; -} - static const struct oxygen_model model_generic = { .shortname = "C-Media CMI8788", .longname = "C-Media Oxygen HD Audio", .chip = "CMI8788", - .owner = THIS_MODULE, - .probe = generic_probe, .init = generic_init, .cleanup = generic_cleanup, .resume = generic_resume, @@ -341,6 +367,42 @@ static const struct oxygen_model model_generic = { .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; +static int __devinit get_oxygen_model(struct oxygen *chip, + const struct pci_device_id *id) +{ + chip->model = model_generic; + switch (id->driver_data) { + case MODEL_MERIDIAN: + chip->model.init = meridian_init; + chip->model.resume = meridian_resume; + chip->model.set_adc_params = set_ak5385_params; + chip->model.device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2 | + CAPTURE_1_FROM_SPDIF; + break; + case MODEL_CLARO: + chip->model.init = claro_init; + chip->model.cleanup = claro_cleanup; + chip->model.suspend = claro_suspend; + chip->model.resume = claro_resume; + break; + case MODEL_CLARO_HALO: + chip->model.init = claro_halo_init; + chip->model.cleanup = claro_cleanup; + chip->model.suspend = claro_suspend; + chip->model.resume = claro_resume; + chip->model.set_adc_params = set_ak5385_params; + break; + } + if (id->driver_data == MODEL_MERIDIAN || + id->driver_data == MODEL_CLARO_HALO) { + chip->model.misc_flags = OXYGEN_MISC_MIDI; + chip->model.device_config |= MIDI_OUTPUT | MIDI_INPUT; + } + return 0; +} + static int __devinit generic_oxygen_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -353,8 +415,8 @@ static int __devinit generic_oxygen_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - err = oxygen_pci_probe(pci, index[dev], id[dev], - &model_generic, pci_id->driver_data); + err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, + oxygen_ids, get_oxygen_model); if (err >= 0) ++dev; return err; diff --git a/sound/pci/oxygen/oxygen.h b/sound/pci/oxygen/oxygen.h index 19107c6..bd615db 100644 --- a/sound/pci/oxygen/oxygen.h +++ b/sound/pci/oxygen/oxygen.h @@ -18,6 +18,8 @@ #define OXYGEN_IO_SIZE 0x100 +#define OXYGEN_EEPROM_ID 0x434d /* "CM" */ + /* model-specific configuration of outputs/inputs */ #define PLAYBACK_0_TO_I2S 0x0001 /* PLAYBACK_0_TO_AC97_0 not implemented */ @@ -49,7 +51,13 @@ enum { .subvendor = sv, \ .subdevice = sd +#define BROKEN_EEPROM_DRIVER_DATA ((unsigned long)-1) +#define OXYGEN_PCI_SUBID_BROKEN_EEPROM \ + OXYGEN_PCI_SUBID(PCI_VENDOR_ID_CMEDIA, 0x8788), \ + .driver_data = BROKEN_EEPROM_DRIVER_DATA + struct pci_dev; +struct pci_device_id; struct snd_card; struct snd_pcm_substream; struct snd_pcm_hardware; @@ -62,8 +70,6 @@ struct oxygen_model { const char *shortname; const char *longname; const char *chip; - struct module *owner; - int (*probe)(struct oxygen *chip, unsigned long driver_data); void (*init)(struct oxygen *chip); int (*control_filter)(struct snd_kcontrol_new *template); int (*mixer_init)(struct oxygen *chip); @@ -83,6 +89,7 @@ struct oxygen_model { void (*ac97_switch)(struct oxygen *chip, unsigned int reg, unsigned int mute); const unsigned int *dac_tlv; + unsigned long private_data; size_t model_data_size; unsigned int device_config; u8 dac_channels; @@ -134,8 +141,12 @@ struct oxygen { /* oxygen_lib.c */ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, - const struct oxygen_model *model, - unsigned long driver_data); + struct module *owner, + const struct pci_device_id *ids, + int (*get_model)(struct oxygen *chip, + const struct pci_device_id *id + ) + ); void oxygen_pci_remove(struct pci_dev *pci); #ifdef CONFIG_PM int oxygen_pci_suspend(struct pci_dev *pci, pm_message_t state); @@ -180,6 +191,9 @@ void oxygen_write_i2c(struct oxygen *chip, u8 device, u8 map, u8 data); void oxygen_reset_uart(struct oxygen *chip); void oxygen_write_uart(struct oxygen *chip, u8 data); +u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index); +void oxygen_write_eeprom(struct oxygen *chip, unsigned int index, u16 value); + static inline void oxygen_set_bits8(struct oxygen *chip, unsigned int reg, u8 value) { diff --git a/sound/pci/oxygen/oxygen_io.c b/sound/pci/oxygen/oxygen_io.c index 3126c4b..c1eb923 100644 --- a/sound/pci/oxygen/oxygen_io.c +++ b/sound/pci/oxygen/oxygen_io.c @@ -254,3 +254,34 @@ void oxygen_write_uart(struct oxygen *chip, u8 data) _write_uart(chip, 0, data); } EXPORT_SYMBOL(oxygen_write_uart); + +u16 oxygen_read_eeprom(struct oxygen *chip, unsigned int index) +{ + unsigned int timeout; + + oxygen_write8(chip, OXYGEN_EEPROM_CONTROL, + index | OXYGEN_EEPROM_DIR_READ); + for (timeout = 0; timeout < 100; ++timeout) { + udelay(1); + if (!(oxygen_read8(chip, OXYGEN_EEPROM_STATUS) + & OXYGEN_EEPROM_BUSY)) + break; + } + return oxygen_read16(chip, OXYGEN_EEPROM_DATA); +} + +void oxygen_write_eeprom(struct oxygen *chip, unsigned int index, u16 value) +{ + unsigned int timeout; + + oxygen_write16(chip, OXYGEN_EEPROM_DATA, value); + oxygen_write8(chip, OXYGEN_EEPROM_CONTROL, + index | OXYGEN_EEPROM_DIR_WRITE); + for (timeout = 0; timeout < 10; ++timeout) { + msleep(1); + if (!(oxygen_read8(chip, OXYGEN_EEPROM_STATUS) + & OXYGEN_EEPROM_BUSY)) + return; + } + snd_printk(KERN_ERR "EEPROM write timeout\n"); +} diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 9c81e0b..312251d 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -34,6 +34,7 @@ MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>"); MODULE_DESCRIPTION("C-Media CMI8788 helper library"); MODULE_LICENSE("GPL v2"); +#define DRIVER "oxygen" static inline int oxygen_uart_input_ready(struct oxygen *chip) { @@ -243,6 +244,62 @@ static void oxygen_proc_init(struct oxygen *chip) #define oxygen_proc_init(chip) #endif +static const struct pci_device_id * +oxygen_search_pci_id(struct oxygen *chip, const struct pci_device_id ids[]) +{ + u16 subdevice; + + /* + * Make sure the EEPROM pins are available, i.e., not used for SPI. + * (This function is called before we initialize or use SPI.) + */ + oxygen_clear_bits8(chip, OXYGEN_FUNCTION, + OXYGEN_FUNCTION_ENABLE_SPI_4_5); + /* + * Read the subsystem device ID directly from the EEPROM, because the + * chip didn't if the first EEPROM word was overwritten. + */ + subdevice = oxygen_read_eeprom(chip, 2); + /* + * We use only the subsystem device ID for searching because it is + * unique even without the subsystem vendor ID, which may have been + * overwritten in the EEPROM. + */ + for (; ids->vendor; ++ids) + if (ids->subdevice == subdevice && + ids->driver_data != BROKEN_EEPROM_DRIVER_DATA) + return ids; + return NULL; +} + +static void oxygen_restore_eeprom(struct oxygen *chip, + const struct pci_device_id *id) +{ + if (oxygen_read_eeprom(chip, 0) != OXYGEN_EEPROM_ID) { + /* + * This function gets called only when a known card model has + * been detected, i.e., we know there is a valid subsystem + * product ID at index 2 in the EEPROM. Therefore, we have + * been able to deduce the correct subsystem vendor ID, and + * this is enough information to restore the original EEPROM + * contents. + */ + oxygen_write_eeprom(chip, 1, id->subvendor); + oxygen_write_eeprom(chip, 0, OXYGEN_EEPROM_ID); + + oxygen_set_bits8(chip, OXYGEN_MISC, + OXYGEN_MISC_WRITE_PCI_SUBID); + pci_write_config_word(chip->pci, PCI_SUBSYSTEM_VENDOR_ID, + id->subvendor); + pci_write_config_word(chip->pci, PCI_SUBSYSTEM_ID, + id->subdevice); + oxygen_clear_bits8(chip, OXYGEN_MISC, + OXYGEN_MISC_WRITE_PCI_SUBID); + + snd_printk(KERN_INFO "EEPROM ID restored\n"); + } +} + static void oxygen_init(struct oxygen *chip) { unsigned int i; @@ -446,21 +503,26 @@ static void oxygen_card_free(struct snd_card *card) free_irq(chip->irq, chip); flush_scheduled_work(); chip->model.cleanup(chip); + kfree(chip->model_data); mutex_destroy(&chip->mutex); pci_release_regions(chip->pci); pci_disable_device(chip->pci); } int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, - const struct oxygen_model *model, - unsigned long driver_data) + struct module *owner, + const struct pci_device_id *ids, + int (*get_model)(struct oxygen *chip, + const struct pci_device_id *id + ) + ) { struct snd_card *card; struct oxygen *chip; + const struct pci_device_id *pci_id; int err; - err = snd_card_create(index, id, model->owner, - sizeof(*chip) + model->model_data_size, &card); + err = snd_card_create(index, id, owner, sizeof(*chip), &card); if (err < 0) return err; @@ -468,8 +530,6 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, chip->card = card; chip->pci = pci; chip->irq = -1; - chip->model = *model; - chip->model_data = chip + 1; spin_lock_init(&chip->reg_lock); mutex_init(&chip->mutex); INIT_WORK(&chip->spdif_input_bits_work, @@ -481,7 +541,7 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, if (err < 0) goto err_card; - err = pci_request_regions(pci, model->chip); + err = pci_request_regions(pci, DRIVER); if (err < 0) { snd_printk(KERN_ERR "cannot reserve PCI resources\n"); goto err_pci_enable; @@ -495,20 +555,34 @@ int oxygen_pci_probe(struct pci_dev *pci, int index, char *id, } chip->addr = pci_resource_start(pci, 0); + pci_id = oxygen_search_pci_id(chip, ids); + if (!pci_id) { + err = -ENODEV; + goto err_pci_regions; + } + oxygen_restore_eeprom(chip, pci_id); + err = get_model(chip, pci_id); + if (err < 0) + goto err_pci_regions; + + if (chip->model.model_data_size) { + chip->model_data = kzalloc(chip->model.model_data_size, + GFP_KERNEL); + if (!chip->model_data) { + err = -ENOMEM; + goto err_pci_regions; + } + } + pci_set_master(pci); snd_card_set_dev(card, &pci->dev); card->private_free = oxygen_card_free; - if (chip->model.probe) { - err = chip->model.probe(chip, driver_data); - if (err < 0) - goto err_card; - } oxygen_init(chip); chip->model.init(chip); err = request_irq(pci->irq, oxygen_interrupt, IRQF_SHARED, - chip->model.chip, chip); + DRIVER, chip); if (err < 0) { snd_printk(KERN_ERR "cannot grab interrupt %d\n", pci->irq); goto err_card; diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 98c6a8c..bc5ce11 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -112,6 +112,34 @@ * CS4362A: AD0 <- 0 */ +/* + * Xonar Essence STX + * ----------------- + * + * CMI8788: + * + * I²C <-> PCM1792A + * + * GPI 0 <- external power present + * + * GPIO 0 -> enable output to speakers + * GPIO 1 -> route HP to front panel (0) or rear jack (1) + * GPIO 2 -> M0 of CS5381 + * GPIO 3 -> M1 of CS5381 + * GPIO 7 -> route output to speaker jacks (0) or HP (1) + * GPIO 8 -> route input jack to line-in (0) or mic-in (1) + * + * PCM1792A: + * + * AD0 <- 0 + * + * H6 daughterboard + * ---------------- + * + * GPIO 4 <- 0 + * GPIO 5 <- 0 + */ + #include <linux/pci.h> #include <linux/delay.h> #include <linux/mutex.h> @@ -152,6 +180,7 @@ enum { MODEL_DX, MODEL_HDAV, /* without daughterboard */ MODEL_HDAV_H6, /* with H6 daughterboard */ + MODEL_STX, }; static struct pci_device_id xonar_ids[] __devinitdata = { @@ -160,6 +189,8 @@ static struct pci_device_id xonar_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X }, { OXYGEN_PCI_SUBID(0x1043, 0x8314), .driver_data = MODEL_HDAV }, { OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 }, + { OXYGEN_PCI_SUBID(0x1043, 0x835c), .driver_data = MODEL_STX }, + { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; MODULE_DEVICE_TABLE(pci, xonar_ids); @@ -183,12 +214,14 @@ MODULE_DEVICE_TABLE(pci, xonar_ids); #define GPIO_HDAV_DB_H6 0x0000 #define GPIO_HDAV_DB_XX 0x0020 +#define GPIO_ST_HP_REAR 0x0002 +#define GPIO_ST_HP 0x0080 + #define I2C_DEVICE_PCM1796(i) (0x98 + ((i) << 1)) /* 10011, ADx=i, /W=0 */ #define I2C_DEVICE_CS4398 0x9e /* 10011, AD1=1, AD0=1, /W=0 */ #define I2C_DEVICE_CS4362A 0x30 /* 001100, AD0=0, /W=0 */ struct xonar_data { - unsigned int model; unsigned int anti_pop_delay; unsigned int dacs; u16 output_enable_bit; @@ -334,15 +367,9 @@ static void xonar_d2_init(struct oxygen *chip) struct xonar_data *data = chip->model_data; data->anti_pop_delay = 300; + data->dacs = 4; data->output_enable_bit = GPIO_D2_OUTPUT_ENABLE; data->pcm1796_oversampling = PCM1796_OS_64; - if (data->model == MODEL_D2X) { - data->ext_power_reg = OXYGEN_GPIO_DATA; - data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK; - data->ext_power_bit = GPIO_D2X_EXT_POWER; - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_D2X_EXT_POWER); - } pcm1796_init(chip); @@ -355,6 +382,18 @@ static void xonar_d2_init(struct oxygen *chip) snd_component_add(chip->card, "CS5381"); } +static void xonar_d2x_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->ext_power_reg = OXYGEN_GPIO_DATA; + data->ext_power_int_reg = OXYGEN_GPIO_INTERRUPT_MASK; + data->ext_power_bit = GPIO_D2X_EXT_POWER; + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, GPIO_D2X_EXT_POWER); + + xonar_d2_init(chip); +} + static void update_cs4362a_volumes(struct oxygen *chip) { u8 mute; @@ -422,11 +461,6 @@ static void xonar_d1_init(struct oxygen *chip) data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; data->cs4362a_fm = CS4362A_FM_SINGLE | CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; - if (data->model == MODEL_DX) { - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; - } oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, OXYGEN_2WIRE_LENGTH_8 | @@ -447,6 +481,17 @@ static void xonar_d1_init(struct oxygen *chip) snd_component_add(chip->card, "CS5361"); } +static void xonar_dx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->ext_power_reg = OXYGEN_GPI_DATA; + data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->ext_power_bit = GPI_DX_EXT_POWER; + + xonar_d1_init(chip); +} + static void xonar_hdav_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -458,6 +503,7 @@ static void xonar_hdav_init(struct oxygen *chip) OXYGEN_2WIRE_SPEED_FAST); data->anti_pop_delay = 100; + data->dacs = chip->model.private_data == MODEL_HDAV_H6 ? 4 : 1; data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; data->ext_power_reg = OXYGEN_GPI_DATA; data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; @@ -484,6 +530,36 @@ static void xonar_hdav_init(struct oxygen *chip) snd_component_add(chip->card, "CS5381"); } +static void xonar_stx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + oxygen_write16(chip, OXYGEN_2WIRE_BUS_STATUS, + OXYGEN_2WIRE_LENGTH_8 | + OXYGEN_2WIRE_INTERRUPT_MASK | + OXYGEN_2WIRE_SPEED_FAST); + + data->anti_pop_delay = 100; + data->dacs = 1; + data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; + data->ext_power_reg = OXYGEN_GPI_DATA; + data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->ext_power_bit = GPI_DX_EXT_POWER; + data->pcm1796_oversampling = PCM1796_OS_64; + + pcm1796_init(chip); + + oxygen_set_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); + oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, + GPIO_DX_INPUT_ROUTE | GPIO_ST_HP_REAR | GPIO_ST_HP); + + xonar_common_init(chip); + + snd_component_add(chip->card, "PCM1792A"); + snd_component_add(chip->card, "CS5381"); +} + static void xonar_disable_output(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -511,6 +587,11 @@ static void xonar_hdav_cleanup(struct oxygen *chip) xonar_disable_output(chip); } +static void xonar_st_cleanup(struct oxygen *chip) +{ + xonar_disable_output(chip); +} + static void xonar_d2_suspend(struct oxygen *chip) { xonar_d2_cleanup(chip); @@ -527,6 +608,11 @@ static void xonar_hdav_suspend(struct oxygen *chip) msleep(2); } +static void xonar_st_suspend(struct oxygen *chip) +{ + xonar_st_cleanup(chip); +} + static void xonar_d2_resume(struct oxygen *chip) { pcm1796_init(chip); @@ -554,6 +640,12 @@ static void xonar_hdav_resume(struct oxygen *chip) xonar_enable_output(chip); } +static void xonar_st_resume(struct oxygen *chip) +{ + pcm1796_init(chip); + xonar_enable_output(chip); +} + static void xonar_hdav_pcm_hardware_filter(unsigned int channel, struct snd_pcm_hardware *hardware) { @@ -676,7 +768,7 @@ static void xonar_hdav_uart_input(struct oxygen *chip) if (chip->uart_input_count >= 2 && chip->uart_input[chip->uart_input_count - 2] == 'O' && chip->uart_input[chip->uart_input_count - 1] == 'K') { - printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:"); + printk(KERN_DEBUG "message from Xonar HDAV HDMI chip received:\n"); print_hex_dump_bytes("", DUMP_PREFIX_OFFSET, chip->uart_input, chip->uart_input_count); chip->uart_input_count = 0; @@ -733,6 +825,72 @@ static const struct snd_kcontrol_new front_panel_switch = { .private_value = GPIO_DX_FRONT_PANEL, }; +static int st_output_switch_info(struct snd_kcontrol *ctl, + struct snd_ctl_elem_info *info) +{ + static const char *const names[3] = { + "Speakers", "Headphones", "FP Headphones" + }; + + info->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + info->count = 1; + info->value.enumerated.items = 3; + if (info->value.enumerated.item >= 3) + info->value.enumerated.item = 2; + strcpy(info->value.enumerated.name, names[info->value.enumerated.item]); + return 0; +} + +static int st_output_switch_get(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 gpio; + + gpio = oxygen_read16(chip, OXYGEN_GPIO_DATA); + if (!(gpio & GPIO_ST_HP)) + value->value.enumerated.item[0] = 0; + else if (gpio & GPIO_ST_HP_REAR) + value->value.enumerated.item[0] = 1; + else + value->value.enumerated.item[0] = 2; + return 0; +} + + +static int st_output_switch_put(struct snd_kcontrol *ctl, + struct snd_ctl_elem_value *value) +{ + struct oxygen *chip = ctl->private_data; + u16 gpio_old, gpio; + + mutex_lock(&chip->mutex); + gpio_old = oxygen_read16(chip, OXYGEN_GPIO_DATA); + gpio = gpio_old; + switch (value->value.enumerated.item[0]) { + case 0: + gpio &= ~(GPIO_ST_HP | GPIO_ST_HP_REAR); + break; + case 1: + gpio |= GPIO_ST_HP | GPIO_ST_HP_REAR; + break; + case 2: + gpio = (gpio | GPIO_ST_HP) & ~GPIO_ST_HP_REAR; + break; + } + oxygen_write16(chip, OXYGEN_GPIO_DATA, gpio); + mutex_unlock(&chip->mutex); + return gpio != gpio_old; +} + +static const struct snd_kcontrol_new st_output_switch = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Analog Output", + .info = st_output_switch_info, + .get = st_output_switch_get, + .put = st_output_switch_put, +}; + static void xonar_line_mic_ac97_switch(struct oxygen *chip, unsigned int reg, unsigned int mute) { @@ -745,8 +903,8 @@ static void xonar_line_mic_ac97_switch(struct oxygen *chip, } } -static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -12000, 50, 0); -static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -12700, 100, 0); +static const DECLARE_TLV_DB_SCALE(pcm1796_db_scale, -6000, 50, 0); +static const DECLARE_TLV_DB_SCALE(cs4362a_db_scale, -6000, 100, 0); static int xonar_d2_control_filter(struct snd_kcontrol_new *template) { @@ -763,6 +921,15 @@ static int xonar_d1_control_filter(struct snd_kcontrol_new *template) return 0; } +static int xonar_st_control_filter(struct snd_kcontrol_new *template) +{ + if (!strncmp(template->name, "CD Capture ", 11)) + return 1; /* no CD input */ + if (!strcmp(template->name, "Stereo Upmixing")) + return 1; /* stereo only - we don't need upmixing */ + return 0; +} + static int xonar_d2_mixer_init(struct oxygen *chip) { return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); @@ -773,51 +940,14 @@ static int xonar_d1_mixer_init(struct oxygen *chip) return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); } -static int xonar_model_probe(struct oxygen *chip, unsigned long driver_data) +static int xonar_st_mixer_init(struct oxygen *chip) { - static const char *const names[] = { - [MODEL_D1] = "Xonar D1", - [MODEL_DX] = "Xonar DX", - [MODEL_D2] = "Xonar D2", - [MODEL_D2X] = "Xonar D2X", - [MODEL_HDAV] = "Xonar HDAV1.3", - [MODEL_HDAV_H6] = "Xonar HDAV1.3+H6", - }; - static const u8 dacs[] = { - [MODEL_D1] = 2, - [MODEL_DX] = 2, - [MODEL_D2] = 4, - [MODEL_D2X] = 4, - [MODEL_HDAV] = 1, - [MODEL_HDAV_H6] = 4, - }; - struct xonar_data *data = chip->model_data; - - data->model = driver_data; - if (data->model == MODEL_HDAV) { - oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, - GPIO_HDAV_DB_MASK); - switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & - GPIO_HDAV_DB_MASK) { - case GPIO_HDAV_DB_H6: - data->model = MODEL_HDAV_H6; - break; - case GPIO_HDAV_DB_XX: - snd_printk(KERN_ERR "unknown daughterboard\n"); - return -ENODEV; - } - } - - data->dacs = dacs[data->model]; - chip->model.shortname = names[data->model]; - return 0; + return snd_ctl_add(chip->card, snd_ctl_new1(&st_output_switch, chip)); } static const struct oxygen_model model_xonar_d2 = { .longname = "Asus Virtuoso 200", .chip = "AV200", - .owner = THIS_MODULE, - .probe = xonar_model_probe, .init = xonar_d2_init, .control_filter = xonar_d2_control_filter, .mixer_init = xonar_d2_mixer_init, @@ -837,8 +967,8 @@ static const struct oxygen_model model_xonar_d2 = { MIDI_OUTPUT | MIDI_INPUT, .dac_channels = 8, - .dac_volume_min = 0x0f, - .dac_volume_max = 0xff, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_SPI | OXYGEN_FUNCTION_ENABLE_SPI_4_5, @@ -849,8 +979,6 @@ static const struct oxygen_model model_xonar_d2 = { static const struct oxygen_model model_xonar_d1 = { .longname = "Asus Virtuoso 100", .chip = "AV200", - .owner = THIS_MODULE, - .probe = xonar_model_probe, .init = xonar_d1_init, .control_filter = xonar_d1_control_filter, .mixer_init = xonar_d1_mixer_init, @@ -868,7 +996,7 @@ static const struct oxygen_model model_xonar_d1 = { PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2, .dac_channels = 8, - .dac_volume_min = 0, + .dac_volume_min = 127 - 60, .dac_volume_max = 127, .function_flags = OXYGEN_FUNCTION_2WIRE, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, @@ -878,8 +1006,6 @@ static const struct oxygen_model model_xonar_d1 = { static const struct oxygen_model model_xonar_hdav = { .longname = "Asus Virtuoso 200", .chip = "AV200", - .owner = THIS_MODULE, - .probe = xonar_model_probe, .init = xonar_hdav_init, .cleanup = xonar_hdav_cleanup, .suspend = xonar_hdav_suspend, @@ -897,15 +1023,43 @@ static const struct oxygen_model model_xonar_hdav = { PLAYBACK_1_TO_SPDIF | CAPTURE_0_FROM_I2S_2, .dac_channels = 8, - .dac_volume_min = 0x0f, - .dac_volume_max = 0xff, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .misc_flags = OXYGEN_MISC_MIDI, .function_flags = OXYGEN_FUNCTION_2WIRE, .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }; -static int __devinit xonar_probe(struct pci_dev *pci, - const struct pci_device_id *pci_id) +static const struct oxygen_model model_xonar_st = { + .longname = "Asus Virtuoso 100", + .chip = "AV200", + .init = xonar_stx_init, + .control_filter = xonar_st_control_filter, + .mixer_init = xonar_st_mixer_init, + .cleanup = xonar_st_cleanup, + .suspend = xonar_st_suspend, + .resume = xonar_st_resume, + .set_dac_params = set_pcm1796_params, + .set_adc_params = set_cs53x1_params, + .update_dac_volume = update_pcm1796_volume, + .update_dac_mute = update_pcm1796_mute, + .ac97_switch = xonar_line_mic_ac97_switch, + .dac_tlv = pcm1796_db_scale, + .model_data_size = sizeof(struct xonar_data), + .device_config = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2, + .dac_channels = 2, + .dac_volume_min = 255 - 2*60, + .dac_volume_max = 255, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, +}; + +static int __devinit get_xonar_model(struct oxygen *chip, + const struct pci_device_id *id) { static const struct oxygen_model *const models[] = { [MODEL_D1] = &model_xonar_d1, @@ -913,7 +1067,57 @@ static int __devinit xonar_probe(struct pci_dev *pci, [MODEL_D2] = &model_xonar_d2, [MODEL_D2X] = &model_xonar_d2, [MODEL_HDAV] = &model_xonar_hdav, + [MODEL_STX] = &model_xonar_st, }; + static const char *const names[] = { + [MODEL_D1] = "Xonar D1", + [MODEL_DX] = "Xonar DX", + [MODEL_D2] = "Xonar D2", + [MODEL_D2X] = "Xonar D2X", + [MODEL_HDAV] = "Xonar HDAV1.3", + [MODEL_HDAV_H6] = "Xonar HDAV1.3+H6", + [MODEL_STX] = "Xonar Essence STX", + }; + unsigned int model = id->driver_data; + + if (model >= ARRAY_SIZE(models) || !models[model]) + return -EINVAL; + chip->model = *models[model]; + + switch (model) { + case MODEL_D2X: + chip->model.init = xonar_d2x_init; + break; + case MODEL_DX: + chip->model.init = xonar_dx_init; + break; + case MODEL_HDAV: + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_HDAV_DB_MASK); + switch (oxygen_read16(chip, OXYGEN_GPIO_DATA) & + GPIO_HDAV_DB_MASK) { + case GPIO_HDAV_DB_H6: + model = MODEL_HDAV_H6; + break; + case GPIO_HDAV_DB_XX: + snd_printk(KERN_ERR "unknown daughterboard\n"); + return -ENODEV; + } + break; + case MODEL_STX: + oxygen_clear_bits16(chip, OXYGEN_GPIO_CONTROL, + GPIO_HDAV_DB_MASK); + break; + } + + chip->model.shortname = names[model]; + chip->model.private_data = model; + return 0; +} + +static int __devinit xonar_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ static int dev; int err; @@ -923,10 +1127,8 @@ static int __devinit xonar_probe(struct pci_dev *pci, ++dev; return -ENOENT; } - BUG_ON(pci_id->driver_data >= ARRAY_SIZE(models)); - err = oxygen_pci_probe(pci, index[dev], id[dev], - models[pci_id->driver_data], - pci_id->driver_data); + err = oxygen_pci_probe(pci, index[dev], id[dev], THIS_MODULE, + xonar_ids, get_xonar_model); if (err >= 0) ++dev; return err; diff --git a/sound/pci/pcxhr/pcxhr.c b/sound/pci/pcxhr/pcxhr.c index 7f95459..833e9c7 100644 --- a/sound/pci/pcxhr/pcxhr.c +++ b/sound/pci/pcxhr/pcxhr.c @@ -1334,6 +1334,40 @@ static void pcxhr_proc_sync(struct snd_info_entry *entry, snd_iprintf(buffer, "\n"); } +static void pcxhr_proc_gpio_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_pcxhr *chip = entry->private_data; + struct pcxhr_mgr *mgr = chip->mgr; + /* commands available when embedded DSP is running */ + if (mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX)) { + /* gpio ports on stereo boards only available */ + int value = 0; + hr222_read_gpio(mgr, 1, &value); /* GPI */ + snd_iprintf(buffer, "GPI: 0x%x\n", value); + hr222_read_gpio(mgr, 0, &value); /* GP0 */ + snd_iprintf(buffer, "GPO: 0x%x\n", value); + } else + snd_iprintf(buffer, "no firmware loaded\n"); + snd_iprintf(buffer, "\n"); +} +static void pcxhr_proc_gpo_write(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) +{ + struct snd_pcxhr *chip = entry->private_data; + struct pcxhr_mgr *mgr = chip->mgr; + char line[64]; + int value; + /* commands available when embedded DSP is running */ + if (!(mgr->dsp_loaded & (1 << PCXHR_FIRMWARE_DSP_MAIN_INDEX))) + return; + while (!snd_info_get_line(buffer, line, sizeof(line))) { + if (sscanf(line, "GPO: 0x%x", &value) != 1) + continue; + hr222_write_gpo(mgr, value); /* GP0 */ + } +} + static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) { struct snd_info_entry *entry; @@ -1342,6 +1376,13 @@ static void __devinit pcxhr_proc_init(struct snd_pcxhr *chip) snd_info_set_text_ops(entry, chip, pcxhr_proc_info); if (! snd_card_proc_new(chip->card, "sync", &entry)) snd_info_set_text_ops(entry, chip, pcxhr_proc_sync); + /* gpio available on stereo sound cards only */ + if (chip->mgr->is_hr_stereo && + !snd_card_proc_new(chip->card, "gpio", &entry)) { + snd_info_set_text_ops(entry, chip, pcxhr_proc_gpio_read); + entry->c.text.write = pcxhr_proc_gpo_write; + entry->mode |= S_IWUSR; + } } /* end of proc interface */ @@ -1408,7 +1449,7 @@ static int __devinit pcxhr_probe(struct pci_dev *pci, pci_set_master(pci); /* check if we can restrict PCI DMA transfers to 32 bits */ - if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(32)) < 0) { snd_printk(KERN_ERR "architecture does not support " "32bit PCI busmaster DMA\n"); pci_disable_device(pci); diff --git a/sound/pci/pcxhr/pcxhr.h b/sound/pci/pcxhr/pcxhr.h index 84131a9..bda776c 100644 --- a/sound/pci/pcxhr/pcxhr.h +++ b/sound/pci/pcxhr/pcxhr.h @@ -27,8 +27,8 @@ #include <linux/mutex.h> #include <sound/pcm.h> -#define PCXHR_DRIVER_VERSION 0x000905 /* 0.9.5 */ -#define PCXHR_DRIVER_VERSION_STRING "0.9.5" /* 0.9.5 */ +#define PCXHR_DRIVER_VERSION 0x000906 /* 0.9.6 */ +#define PCXHR_DRIVER_VERSION_STRING "0.9.6" /* 0.9.6 */ #define PCXHR_MAX_CARDS 6 @@ -97,12 +97,12 @@ struct pcxhr_mgr { int capture_chips; int fw_file_set; int firmware_num; - int is_hr_stereo:1; - int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */ - int board_has_analog:1; /* if 0 the board is digital only */ - int board_has_mic:1; /* if 1 the board has microphone input */ - int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ - int mono_capture:1; /* if 1 the board does mono capture */ + unsigned int is_hr_stereo:1; + unsigned int board_has_aes1:1; /* if 1 board has AES1 plug and SRC */ + unsigned int board_has_analog:1; /* if 0 the board is digital only */ + unsigned int board_has_mic:1; /* if 1 the board has microphone input */ + unsigned int board_aes_in_192k:1;/* if 1 the aes input plugs do support 192kHz */ + unsigned int mono_capture:1; /* if 1 the board does mono capture */ struct snd_dma_buffer hostport; @@ -124,6 +124,7 @@ struct pcxhr_mgr { unsigned char xlx_cfg; /* copy of PCXHR_XLX_CFG register */ unsigned char xlx_selmic; /* copy of PCXHR_XLX_SELMIC register */ + unsigned char dsp_reset; /* copy of PCXHR_DSP_RESET register */ }; diff --git a/sound/pci/pcxhr/pcxhr_core.h b/sound/pci/pcxhr/pcxhr_core.h index bbbd66d..be01737 100644 --- a/sound/pci/pcxhr/pcxhr_core.h +++ b/sound/pci/pcxhr/pcxhr_core.h @@ -1,7 +1,7 @@ /* * Driver for Digigram pcxhr compatible soundcards * - * low level interface with interrupt ans message handling + * low level interface with interrupt and message handling * * Copyright (c) 2004 by Digigram <alsa@digigram.com> * diff --git a/sound/pci/pcxhr/pcxhr_hwdep.c b/sound/pci/pcxhr/pcxhr_hwdep.c index 592743a..17cb123 100644 --- a/sound/pci/pcxhr/pcxhr_hwdep.c +++ b/sound/pci/pcxhr/pcxhr_hwdep.c @@ -471,16 +471,6 @@ static int pcxhr_hwdep_dsp_load(struct snd_hwdep *hw, return 0; } -static int pcxhr_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int pcxhr_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) { int err; @@ -495,8 +485,6 @@ int pcxhr_setup_firmware(struct pcxhr_mgr *mgr) hw->iface = SNDRV_HWDEP_IFACE_PCXHR; hw->private_data = mgr; - hw->ops.open = pcxhr_hwdep_open; - hw->ops.release = pcxhr_hwdep_release; hw->ops.dsp_status = pcxhr_hwdep_dsp_status; hw->ops.dsp_load = pcxhr_hwdep_dsp_load; hw->exclusive = 1; diff --git a/sound/pci/pcxhr/pcxhr_mix22.c b/sound/pci/pcxhr/pcxhr_mix22.c index ff01912..1cb82c0 100644 --- a/sound/pci/pcxhr/pcxhr_mix22.c +++ b/sound/pci/pcxhr/pcxhr_mix22.c @@ -53,6 +53,8 @@ #define PCXHR_DSP_RESET_DSP 0x01 #define PCXHR_DSP_RESET_MUTE 0x02 #define PCXHR_DSP_RESET_CODEC 0x08 +#define PCXHR_DSP_RESET_GPO_OFFSET 5 +#define PCXHR_DSP_RESET_GPO_MASK 0x60 /* values for PCHR_XLX_CFG register */ #define PCXHR_CFG_SYNCDSP_MASK 0x80 @@ -81,6 +83,8 @@ /* values for PCHR_XLX_STATUS register - READ */ #define PCXHR_STAT_SRC_LOCK 0x01 #define PCXHR_STAT_LEVEL_IN 0x02 +#define PCXHR_STAT_GPI_OFFSET 2 +#define PCXHR_STAT_GPI_MASK 0x0C #define PCXHR_STAT_MIC_CAPS 0x10 /* values for PCHR_XLX_STATUS register - WRITE */ #define PCXHR_STAT_FREQ_SYNC_MASK 0x01 @@ -291,10 +295,11 @@ int hr222_sub_init(struct pcxhr_mgr *mgr) PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, PCXHR_DSP_RESET_DSP); msleep(5); - PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, - PCXHR_DSP_RESET_DSP | - PCXHR_DSP_RESET_MUTE | - PCXHR_DSP_RESET_CODEC); + mgr->dsp_reset = PCXHR_DSP_RESET_DSP | + PCXHR_DSP_RESET_MUTE | + PCXHR_DSP_RESET_CODEC; + PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, mgr->dsp_reset); + /* hr222_write_gpo(mgr, 0); does the same */ msleep(5); /* config AKM */ @@ -496,6 +501,33 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr, } +int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value) +{ + if (is_gpi) { + unsigned char reg = PCXHR_INPB(mgr, PCXHR_XLX_STATUS); + *value = (int)(reg & PCXHR_STAT_GPI_MASK) >> + PCXHR_STAT_GPI_OFFSET; + } else { + *value = (int)(mgr->dsp_reset & PCXHR_DSP_RESET_GPO_MASK) >> + PCXHR_DSP_RESET_GPO_OFFSET; + } + return 0; +} + + +int hr222_write_gpo(struct pcxhr_mgr *mgr, int value) +{ + unsigned char reg = mgr->dsp_reset & ~PCXHR_DSP_RESET_GPO_MASK; + + reg |= (unsigned char)(value << PCXHR_DSP_RESET_GPO_OFFSET) & + PCXHR_DSP_RESET_GPO_MASK; + + PCXHR_OUTPB(mgr, PCXHR_DSP_RESET, reg); + mgr->dsp_reset = reg; + return 0; +} + + int hr222_update_analog_audio_level(struct snd_pcxhr *chip, int is_capture, int channel) { diff --git a/sound/pci/pcxhr/pcxhr_mix22.h b/sound/pci/pcxhr/pcxhr_mix22.h index 6b318b2..5a37a00 100644 --- a/sound/pci/pcxhr/pcxhr_mix22.h +++ b/sound/pci/pcxhr/pcxhr_mix22.h @@ -32,6 +32,9 @@ int hr222_get_external_clock(struct pcxhr_mgr *mgr, enum pcxhr_clock_type clock_type, int *sample_rate); +int hr222_read_gpio(struct pcxhr_mgr *mgr, int is_gpi, int *value); +int hr222_write_gpo(struct pcxhr_mgr *mgr, int value); + #define HR222_LINE_PLAYBACK_LEVEL_MIN 0 /* -25.5 dB */ #define HR222_LINE_PLAYBACK_ZERO_LEVEL 51 /* 0.0 dB */ #define HR222_LINE_PLAYBACK_LEVEL_MAX 99 /* +24.0 dB */ diff --git a/sound/pci/pcxhr/pcxhr_mixer.c b/sound/pci/pcxhr/pcxhr_mixer.c index 2436e37..fec0493 100644 --- a/sound/pci/pcxhr/pcxhr_mixer.c +++ b/sound/pci/pcxhr/pcxhr_mixer.c @@ -789,11 +789,15 @@ static int pcxhr_clock_type_put(struct snd_kcontrol *kcontrol, if (mgr->use_clock_type != ucontrol->value.enumerated.item[0]) { mutex_lock(&mgr->setup_mutex); mgr->use_clock_type = ucontrol->value.enumerated.item[0]; - if (mgr->use_clock_type) + rate = 0; + if (mgr->use_clock_type != PCXHR_CLOCK_TYPE_INTERNAL) { pcxhr_get_external_clock(mgr, mgr->use_clock_type, &rate); - else + } else { rate = mgr->sample_rate; + if (!rate) + rate = 48000; + } if (rate) { pcxhr_set_clock(mgr, rate); if (mgr->sample_rate) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 05b3f79..314e735 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -113,7 +113,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); /* the meters are regular i/o-mapped registers, but offset considerably from the rest. the peak registers are reset - when read; the least-significant 4 bits are full-scale counters; + when read; the least-significant 4 bits are full-scale counters; the actual peak value is in the most-significant 24 bits. */ @@ -131,7 +131,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); 26*3 values are read in ss mode 14*3 in ds mode, with no gap between values */ -#define HDSP_9652_peakBase 7164 +#define HDSP_9652_peakBase 7164 #define HDSP_9652_rmsBase 4096 /* c.f. the hdsp_9632_meters_t struct */ @@ -173,12 +173,12 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_SPDIFEmphasis (1<<10) /* 0=none, 1=on */ #define HDSP_SPDIFNonAudio (1<<11) /* 0=off, 1=on */ #define HDSP_SPDIFOpticalOut (1<<12) /* 1=use 1st ADAT connector for SPDIF, 0=do not */ -#define HDSP_SyncRef2 (1<<13) -#define HDSP_SPDIFInputSelect0 (1<<14) -#define HDSP_SPDIFInputSelect1 (1<<15) -#define HDSP_SyncRef0 (1<<16) +#define HDSP_SyncRef2 (1<<13) +#define HDSP_SPDIFInputSelect0 (1<<14) +#define HDSP_SPDIFInputSelect1 (1<<15) +#define HDSP_SyncRef0 (1<<16) #define HDSP_SyncRef1 (1<<17) -#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ +#define HDSP_AnalogExtensionBoard (1<<18) /* For H9632 cards */ #define HDSP_XLRBreakoutCable (1<<20) /* For H9632 cards */ #define HDSP_Midi0InterruptEnable (1<<22) #define HDSP_Midi1InterruptEnable (1<<23) @@ -314,7 +314,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_TimecodeSync (1<<27) #define HDSP_AEBO (1<<28) /* H9632 specific Analog Extension Boards */ #define HDSP_AEBI (1<<29) /* 0 = present, 1 = absent */ -#define HDSP_midi0IRQPending (1<<30) +#define HDSP_midi0IRQPending (1<<30) #define HDSP_midi1IRQPending (1<<31) #define HDSP_spdifFrequencyMask (HDSP_spdifFrequency0|HDSP_spdifFrequency1|HDSP_spdifFrequency2) @@ -391,7 +391,7 @@ MODULE_FIRMWARE("digiface_firmware_rev11.bin"); #define HDSP_CHANNEL_BUFFER_BYTES (4*HDSP_CHANNEL_BUFFER_SAMPLES) /* the size of the area we need to allocate for DMA transfers. the - size is the same regardless of the number of channels - the + size is the same regardless of the number of channels - the Multiface still uses the same memory area. Note that we allocate 1 more channel than is apparently needed @@ -460,7 +460,7 @@ struct hdsp { unsigned char qs_in_channels; /* quad speed mode for H9632 */ unsigned char ds_in_channels; unsigned char ss_in_channels; /* different for multiface/digiface */ - unsigned char qs_out_channels; + unsigned char qs_out_channels; unsigned char ds_out_channels; unsigned char ss_out_channels; @@ -502,9 +502,9 @@ static char channel_map_df_ss[HDSP_MAX_CHANNELS] = { static char channel_map_mf_ss[HDSP_MAX_CHANNELS] = { /* Multiface */ /* Analog */ - 0, 1, 2, 3, 4, 5, 6, 7, + 0, 1, 2, 3, 4, 5, 6, 7, /* ADAT 2 */ - 16, 17, 18, 19, 20, 21, 22, 23, + 16, 17, 18, 19, 20, 21, 22, 23, /* SPDIF */ 24, 25, -1, -1, -1, -1, -1, -1, -1, -1 @@ -525,11 +525,11 @@ static char channel_map_H9632_ss[HDSP_MAX_CHANNELS] = { /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ - -1, -1, -1, -1, -1, -1, -1, -1, + -1, -1, -1, -1, -1, -1, -1, -1, -1, -1 }; @@ -539,7 +539,7 @@ static char channel_map_H9632_ds[HDSP_MAX_CHANNELS] = { /* SPDIF */ 8, 9, /* Analog */ - 10, 11, + 10, 11, /* AO4S-192 and AI4S-192 extension boards */ 12, 13, 14, 15, /* others don't exist */ @@ -587,7 +587,7 @@ static void snd_hammerfall_free_buffer(struct snd_dma_buffer *dmab, struct pci_d static struct pci_device_id snd_hdsp_ids[] = { { .vendor = PCI_VENDOR_ID_XILINX, - .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, + .device = PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP, .subvendor = PCI_ANY_ID, .subdevice = PCI_ANY_ID, }, /* RME Hammerfall-DSP */ @@ -653,7 +653,6 @@ static unsigned int hdsp_read(struct hdsp *hdsp, int reg) static int hdsp_check_for_iobox (struct hdsp *hdsp) { - if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp_read (hdsp, HDSP_statusRegister) & HDSP_ConfigError) { snd_printk ("Hammerfall-DSP: no Digiface or Multiface connected!\n"); @@ -661,7 +660,29 @@ static int hdsp_check_for_iobox (struct hdsp *hdsp) return -EIO; } return 0; +} +static int hdsp_wait_for_iobox(struct hdsp *hdsp, unsigned int loops, + unsigned int delay) +{ + unsigned int i; + + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) + return 0; + + for (i = 0; i != loops; ++i) { + if (hdsp_read(hdsp, HDSP_statusRegister) & HDSP_ConfigError) + msleep(delay); + else { + snd_printd("Hammerfall-DSP: iobox found after %ums!\n", + i * delay); + return 0; + } + } + + snd_printk("Hammerfall-DSP: no Digiface or Multiface connected!\n"); + hdsp->state &= ~HDSP_FirmwareLoaded; + return -EIO; } static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { @@ -670,19 +691,19 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { unsigned long flags; if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + snd_printk ("Hammerfall-DSP: loading firmware\n"); hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout waiting for download preparation\n"); return -EIO; } - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_S_LOAD); - + for (i = 0; i < 24413; ++i) { hdsp_write(hdsp, HDSP_fifoData, hdsp->firmware_cache[i]); if (hdsp_fifo_wait (hdsp, 127, HDSP_LONG_WAIT)) { @@ -692,7 +713,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { } ssleep(3); - + if (hdsp_fifo_wait (hdsp, 0, HDSP_LONG_WAIT)) { snd_printk ("Hammerfall-DSP: timeout at end of firmware loading\n"); return -EIO; @@ -705,15 +726,15 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { #endif hdsp_write (hdsp, HDSP_control2Reg, hdsp->control2_register); snd_printk ("Hammerfall-DSP: finished firmware loading\n"); - + } if (hdsp->state & HDSP_InitializationComplete) { snd_printk(KERN_INFO "Hammerfall-DSP: firmware loaded from cache, restoring defaults\n"); spin_lock_irqsave(&hdsp->lock, flags); snd_hdsp_set_defaults(hdsp); - spin_unlock_irqrestore(&hdsp->lock, flags); + spin_unlock_irqrestore(&hdsp->lock, flags); } - + hdsp->state |= HDSP_FirmwareLoaded; return 0; @@ -722,7 +743,7 @@ static int snd_hdsp_load_firmware_from_cache(struct hdsp *hdsp) { static int hdsp_get_iobox_version (struct hdsp *hdsp) { if ((hdsp_read (hdsp, HDSP_statusRegister) & HDSP_DllError) != 0) { - + hdsp_write (hdsp, HDSP_control2Reg, HDSP_PROGRAM); hdsp_write (hdsp, HDSP_fifoData, 0); if (hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT) < 0) @@ -738,7 +759,7 @@ static int hdsp_get_iobox_version (struct hdsp *hdsp) hdsp_fifo_wait (hdsp, 0, HDSP_SHORT_WAIT); } else { hdsp->io_type = Digiface; - } + } } else { /* firmware was already loaded, get iobox type */ if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) @@ -786,13 +807,13 @@ static int hdsp_check_for_firmware (struct hdsp *hdsp, int load_on_demand) static int hdsp_fifo_wait(struct hdsp *hdsp, int count, int timeout) -{ +{ int i; /* the fifoStatus registers reports on how many words are available in the command FIFO. */ - + for (i = 0; i < timeout; i++) { if ((int)(hdsp_read (hdsp, HDSP_fifoStatus) & 0xff) <= count) @@ -824,11 +845,11 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short if (addr >= HDSP_MATRIX_MIXER_SIZE) return -1; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) { /* from martin bjornsen: - + "You can only write dwords to the mixer memory which contain two mixer values in the low and high @@ -847,7 +868,7 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short hdsp->mixer_matrix[addr] = data; - + /* `addr' addresses a 16-bit wide address, but the address space accessed via hdsp_write uses byte offsets. put another way, addr @@ -856,17 +877,17 @@ static int hdsp_write_gain(struct hdsp *hdsp, unsigned int addr, unsigned short to access 0 to 2703 ... */ ad = addr/2; - - hdsp_write (hdsp, 4096 + (ad*4), - (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + + + hdsp_write (hdsp, 4096 + (ad*4), + (hdsp->mixer_matrix[(addr&0x7fe)+1] << 16) + hdsp->mixer_matrix[addr&0x7fe]); - + return 0; } else { ad = (addr << 16) + data; - + if (hdsp_fifo_wait(hdsp, 127, HDSP_LONG_WAIT)) return -1; @@ -902,7 +923,7 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) if (status & HDSP_SPDIFErrorFlag) return 0; - + switch (rate_bits) { case HDSP_spdifFrequency32KHz: return 32000; case HDSP_spdifFrequency44_1KHz: return 44100; @@ -910,13 +931,13 @@ static int hdsp_spdif_sample_rate(struct hdsp *hdsp) case HDSP_spdifFrequency64KHz: return 64000; case HDSP_spdifFrequency88_2KHz: return 88200; case HDSP_spdifFrequency96KHz: return 96000; - case HDSP_spdifFrequency128KHz: + case HDSP_spdifFrequency128KHz: if (hdsp->io_type == H9632) return 128000; break; - case HDSP_spdifFrequency176_4KHz: + case HDSP_spdifFrequency176_4KHz: if (hdsp->io_type == H9632) return 176400; break; - case HDSP_spdifFrequency192KHz: + case HDSP_spdifFrequency192KHz: if (hdsp->io_type == H9632) return 192000; break; default: @@ -1027,7 +1048,7 @@ static void hdsp_set_dds_value(struct hdsp *hdsp, int rate) { u64 n; u32 r; - + if (rate >= 112000) rate /= 4; else if (rate >= 56000) @@ -1053,35 +1074,35 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) there is no need for it (e.g. during module initialization). */ - - if (!(hdsp->control_register & HDSP_ClockModeMaster)) { + + if (!(hdsp->control_register & HDSP_ClockModeMaster)) { if (called_internally) { /* request from ctl or card initialization */ snd_printk(KERN_ERR "Hammerfall-DSP: device is not running as a clock master: cannot set sample rate.\n"); return -1; - } else { + } else { /* hw_param request while in AutoSync mode */ int external_freq = hdsp_external_sample_rate(hdsp); int spdif_freq = hdsp_spdif_sample_rate(hdsp); - + if ((spdif_freq == external_freq*2) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in double speed mode\n"); else if (hdsp->io_type == H9632 && (spdif_freq == external_freq*4) && (hdsp_autosync_ref(hdsp) >= HDSP_AUTOSYNC_FROM_ADAT1)) - snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Detected ADAT in quad speed mode\n"); else if (rate != external_freq) { snd_printk(KERN_INFO "Hammerfall-DSP: No AutoSync source for requested rate\n"); return -1; - } - } + } + } } current_rate = hdsp->system_sample_rate; /* Changing from a "single speed" to a "double speed" rate is not allowed if any substreams are open. This is because - such a change causes a shift in the location of + such a change causes a shift in the location of the DMA buffers and a reduction in the number of available - buffers. + buffers. Note that a similar but essentially insoluble problem exists for externally-driven rate changes. All we can do @@ -1089,7 +1110,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) if (rate > 96000 && hdsp->io_type != H9632) return -EINVAL; - + switch (rate) { case 32000: if (current_rate > 48000) @@ -1179,7 +1200,7 @@ static int hdsp_set_rate(struct hdsp *hdsp, int rate, int called_internally) break; } } - + hdsp->system_sample_rate = rate; return 0; @@ -1245,16 +1266,16 @@ static int snd_hdsp_midi_output_write (struct hdsp_midi *hmidi) unsigned char buf[128]; /* Output is not interrupt driven */ - + spin_lock_irqsave (&hmidi->lock, flags); if (hmidi->output) { if (!snd_rawmidi_transmit_empty (hmidi->output)) { if ((n_pending = snd_hdsp_midi_output_possible (hmidi->hdsp, hmidi->id)) > 0) { if (n_pending > (int)sizeof (buf)) n_pending = sizeof (buf); - + if ((to_write = snd_rawmidi_transmit (hmidi->output, buf, n_pending)) > 0) { - for (i = 0; i < to_write; ++i) + for (i = 0; i < to_write; ++i) snd_hdsp_midi_write_byte (hmidi->hdsp, hmidi->id, buf[i]); } } @@ -1325,14 +1346,14 @@ static void snd_hdsp_midi_output_timer(unsigned long data) { struct hdsp_midi *hmidi = (struct hdsp_midi *) data; unsigned long flags; - + snd_hdsp_midi_output_write(hmidi); spin_lock_irqsave (&hmidi->lock, flags); /* this does not bump hmidi->istimer, because the kernel automatically removed the timer when it expired, and we are now adding it back, thus - leaving istimer wherever it was set before. + leaving istimer wherever it was set before. */ if (hmidi->istimer) { @@ -1501,7 +1522,7 @@ static int snd_hdsp_control_spdif_info(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_control_spdif_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif); return 0; } @@ -1511,7 +1532,7 @@ static int snd_hdsp_control_spdif_put(struct snd_kcontrol *kcontrol, struct snd_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif; @@ -1530,7 +1551,7 @@ static int snd_hdsp_control_spdif_stream_info(struct snd_kcontrol *kcontrol, str static int snd_hdsp_control_spdif_stream_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + snd_hdsp_convert_to_aes(&ucontrol->value.iec958, hdsp->creg_spdif_stream); return 0; } @@ -1540,7 +1561,7 @@ static int snd_hdsp_control_spdif_stream_put(struct snd_kcontrol *kcontrol, stru struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; u32 val; - + val = snd_hdsp_convert_from_aes(&ucontrol->value.iec958); spin_lock_irq(&hdsp->lock); change = val != hdsp->creg_spdif_stream; @@ -1602,7 +1623,7 @@ static int snd_hdsp_info_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_ static int snd_hdsp_get_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_spdif_in(hdsp); return 0; } @@ -1612,7 +1633,7 @@ static int snd_hdsp_put_spdif_in(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0] % ((hdsp->io_type == H9632) ? 4 : 3); @@ -1649,7 +1670,7 @@ static int hdsp_set_spdif_output(struct hdsp *hdsp, int out) static int snd_hdsp_get_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_out(hdsp); return 0; } @@ -1659,7 +1680,7 @@ static int snd_hdsp_put_spdif_out(struct snd_kcontrol *kcontrol, struct snd_ctl_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1693,7 +1714,7 @@ static int hdsp_set_spdif_professional(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_professional(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_professional(hdsp); return 0; } @@ -1703,7 +1724,7 @@ static int snd_hdsp_put_spdif_professional(struct snd_kcontrol *kcontrol, struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1737,7 +1758,7 @@ static int hdsp_set_spdif_emphasis(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_emphasis(hdsp); return 0; } @@ -1747,7 +1768,7 @@ static int snd_hdsp_put_spdif_emphasis(struct snd_kcontrol *kcontrol, struct snd struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1781,7 +1802,7 @@ static int hdsp_set_spdif_nonaudio(struct hdsp *hdsp, int val) static int snd_hdsp_get_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp_spdif_nonaudio(hdsp); return 0; } @@ -1791,7 +1812,7 @@ static int snd_hdsp_put_spdif_nonaudio(struct snd_kcontrol *kcontrol, struct snd struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -1828,7 +1849,7 @@ static int snd_hdsp_info_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_spdif_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1858,7 +1879,7 @@ static int snd_hdsp_get_spdif_sample_rate(struct snd_kcontrol *kcontrol, struct ucontrol->value.enumerated.item[0] = 9; break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1882,7 +1903,7 @@ static int snd_hdsp_info_system_sample_rate(struct snd_kcontrol *kcontrol, struc static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp->system_sample_rate; return 0; } @@ -1899,7 +1920,7 @@ static int snd_hdsp_get_system_sample_rate(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; + static char *texts[] = {"32000", "44100", "48000", "64000", "88200", "96000", "None", "128000", "176400", "192000"}; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = (hdsp->io_type == H9632) ? 10 : 7 ; @@ -1912,7 +1933,7 @@ static int snd_hdsp_info_autosync_sample_rate(struct snd_kcontrol *kcontrol, str static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + switch (hdsp_external_sample_rate(hdsp)) { case 32000: ucontrol->value.enumerated.item[0] = 0; @@ -1940,9 +1961,9 @@ static int snd_hdsp_get_autosync_sample_rate(struct snd_kcontrol *kcontrol, stru break; case 192000: ucontrol->value.enumerated.item[0] = 9; - break; + break; default: - ucontrol->value.enumerated.item[0] = 6; + ucontrol->value.enumerated.item[0] = 6; } return 0; } @@ -1968,7 +1989,7 @@ static int hdsp_system_clock_mode(struct hdsp *hdsp) static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Master", "Slave" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 2; @@ -1981,7 +2002,7 @@ static int snd_hdsp_info_system_clock_mode(struct snd_kcontrol *kcontrol, struct static int snd_hdsp_get_system_clock_mode(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_system_clock_mode(hdsp); return 0; } @@ -2018,7 +2039,7 @@ static int hdsp_clock_source(struct hdsp *hdsp) case 192000: return 9; default: - return 3; + return 3; } } else { return 0; @@ -2032,7 +2053,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) case HDSP_CLOCK_SOURCE_AUTOSYNC: if (hdsp_external_sample_rate(hdsp) != 0) { if (!hdsp_set_rate(hdsp, hdsp_external_sample_rate(hdsp), 1)) { - hdsp->control_register &= ~HDSP_ClockModeMaster; + hdsp->control_register &= ~HDSP_ClockModeMaster; hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); return 0; } @@ -2043,7 +2064,7 @@ static int hdsp_set_clock_source(struct hdsp *hdsp, int mode) break; case HDSP_CLOCK_SOURCE_INTERNAL_44_1KHZ: rate = 44100; - break; + break; case HDSP_CLOCK_SOURCE_INTERNAL_48KHZ: rate = 48000; break; @@ -2078,13 +2099,13 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ { static char *texts[] = {"AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz", "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz", "Internal 96.0 kHz", "Internal 128 kHz", "Internal 176.4 kHz", "Internal 192.0 KHz" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; if (hdsp->io_type == H9632) uinfo->value.enumerated.items = 10; else - uinfo->value.enumerated.items = 7; + uinfo->value.enumerated.items = 7; if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2094,7 +2115,7 @@ static int snd_hdsp_info_clock_source(struct snd_kcontrol *kcontrol, struct snd_ static int snd_hdsp_get_clock_source(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_clock_source(hdsp); return 0; } @@ -2104,7 +2125,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2130,7 +2151,7 @@ static int snd_hdsp_put_clock_source(struct snd_kcontrol *kcontrol, struct snd_c static int snd_hdsp_get_clock_source_lock(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.integer.value[0] = hdsp->clock_source_locked; return 0; } @@ -2165,7 +2186,7 @@ static int hdsp_da_gain(struct hdsp *hdsp) case HDSP_DAGainMinus10dBV: return 2; default: - return 1; + return 1; } } @@ -2180,8 +2201,8 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_DAGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_DAGainMinus10dBV; - break; + hdsp->control_register |= HDSP_DAGainMinus10dBV; + break; default: return -1; @@ -2193,7 +2214,7 @@ static int hdsp_set_da_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Hi Gain", "+4 dBu", "-10 dbV"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2206,7 +2227,7 @@ static int snd_hdsp_info_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int snd_hdsp_get_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_da_gain(hdsp); return 0; } @@ -2216,7 +2237,7 @@ static int snd_hdsp_put_da_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2250,7 +2271,7 @@ static int hdsp_ad_gain(struct hdsp *hdsp) case HDSP_ADGainLowGain: return 2; default: - return 1; + return 1; } } @@ -2262,11 +2283,11 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_ADGainMinus10dBV; break; case 1: - hdsp->control_register |= HDSP_ADGainPlus4dBu; + hdsp->control_register |= HDSP_ADGainPlus4dBu; break; case 2: - hdsp->control_register |= HDSP_ADGainLowGain; - break; + hdsp->control_register |= HDSP_ADGainLowGain; + break; default: return -1; @@ -2278,7 +2299,7 @@ static int hdsp_set_ad_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"-10 dBV", "+4 dBu", "Lo Gain"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2291,7 +2312,7 @@ static int snd_hdsp_info_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_e static int snd_hdsp_get_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_ad_gain(hdsp); return 0; } @@ -2301,7 +2322,7 @@ static int snd_hdsp_put_ad_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_el struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2335,7 +2356,7 @@ static int hdsp_phone_gain(struct hdsp *hdsp) case HDSP_PhoneGainMinus12dB: return 2; default: - return 0; + return 0; } } @@ -2347,11 +2368,11 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) hdsp->control_register |= HDSP_PhoneGain0dB; break; case 1: - hdsp->control_register |= HDSP_PhoneGainMinus6dB; + hdsp->control_register |= HDSP_PhoneGainMinus6dB; break; case 2: - hdsp->control_register |= HDSP_PhoneGainMinus12dB; - break; + hdsp->control_register |= HDSP_PhoneGainMinus12dB; + break; default: return -1; @@ -2363,7 +2384,7 @@ static int hdsp_set_phone_gain(struct hdsp *hdsp, int mode) static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"0 dB", "-6 dB", "-12 dB"}; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2376,7 +2397,7 @@ static int snd_hdsp_info_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_get_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_phone_gain(hdsp); return 0; } @@ -2386,7 +2407,7 @@ static int snd_hdsp_put_phone_gain(struct snd_kcontrol *kcontrol, struct snd_ctl struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -2432,7 +2453,7 @@ static int hdsp_set_xlr_breakout_cable(struct hdsp *hdsp, int mode) static int snd_hdsp_get_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_xlr_breakout_cable(hdsp); return 0; } @@ -2442,7 +2463,7 @@ static int snd_hdsp_put_xlr_breakout_cable(struct snd_kcontrol *kcontrol, struct struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2488,7 +2509,7 @@ static int hdsp_set_aeb(struct hdsp *hdsp, int mode) static int snd_hdsp_get_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_aeb(hdsp); return 0; } @@ -2498,7 +2519,7 @@ static int snd_hdsp_put_aeb(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_v struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2576,7 +2597,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd { static char *texts[] = {"Word", "IEC958", "ADAT1", "ADAT Sync", "ADAT2", "ADAT3" }; struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; @@ -2595,7 +2616,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd uinfo->value.enumerated.items = 0; break; } - + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) uinfo->value.enumerated.item = uinfo->value.enumerated.items - 1; strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); @@ -2605,7 +2626,7 @@ static int snd_hdsp_info_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd static int snd_hdsp_get_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_pref_sync_ref(hdsp); return 0; } @@ -2615,7 +2636,7 @@ static int snd_hdsp_put_pref_sync_ref(struct snd_kcontrol *kcontrol, struct snd_ struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change, max; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; @@ -2664,7 +2685,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) case HDSP_SelSyncRef_SPDIF: return HDSP_AUTOSYNC_FROM_SPDIF; case HDSP_SelSyncRefMask: - return HDSP_AUTOSYNC_FROM_NONE; + return HDSP_AUTOSYNC_FROM_NONE; case HDSP_SelSyncRef_ADAT1: return HDSP_AUTOSYNC_FROM_ADAT1; case HDSP_SelSyncRef_ADAT2: @@ -2680,7 +2701,7 @@ static int hdsp_autosync_ref(struct hdsp *hdsp) static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[] = {"Word", "ADAT Sync", "IEC958", "None", "ADAT1", "ADAT2", "ADAT3" }; - + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 7; @@ -2693,7 +2714,7 @@ static int snd_hdsp_info_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ static int snd_hdsp_get_autosync_ref(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_autosync_ref(hdsp); return 0; } @@ -2727,7 +2748,7 @@ static int hdsp_set_line_output(struct hdsp *hdsp, int out) static int snd_hdsp_get_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp_line_out(hdsp); spin_unlock_irq(&hdsp->lock); @@ -2739,7 +2760,7 @@ static int snd_hdsp_put_line_out(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2773,7 +2794,7 @@ static int hdsp_set_precise_pointer(struct hdsp *hdsp, int precise) static int snd_hdsp_get_precise_pointer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->precise_ptr; spin_unlock_irq(&hdsp->lock); @@ -2785,7 +2806,7 @@ static int snd_hdsp_put_precise_pointer(struct snd_kcontrol *kcontrol, struct sn struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2819,7 +2840,7 @@ static int hdsp_set_use_midi_tasklet(struct hdsp *hdsp, int use_tasklet) static int snd_hdsp_get_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[0] = hdsp->use_midi_tasklet; spin_unlock_irq(&hdsp->lock); @@ -2831,7 +2852,7 @@ static int snd_hdsp_put_use_midi_tasklet(struct snd_kcontrol *kcontrol, struct s struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; unsigned int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.integer.value[0] & 1; @@ -2873,12 +2894,12 @@ static int snd_hdsp_get_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem source = ucontrol->value.integer.value[0]; destination = ucontrol->value.integer.value[1]; - + if (source >= hdsp->max_channels) addr = hdsp_playback_to_output_key(hdsp,source-hdsp->max_channels,destination); else addr = hdsp_input_to_output_key(hdsp,source, destination); - + spin_lock_irq(&hdsp->lock); ucontrol->value.integer.value[2] = hdsp_read_gain (hdsp, addr); spin_unlock_irq(&hdsp->lock); @@ -2926,7 +2947,7 @@ static int snd_hdsp_put_mixer(struct snd_kcontrol *kcontrol, struct snd_ctl_elem static int snd_hdsp_info_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { - static char *texts[] = {"No Lock", "Lock", "Sync" }; + static char *texts[] = {"No Lock", "Lock", "Sync" }; uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 1; uinfo->value.enumerated.items = 3; @@ -2971,7 +2992,7 @@ static int hdsp_spdif_sync_check(struct hdsp *hdsp) int status = hdsp_read(hdsp, HDSP_statusRegister); if (status & HDSP_SPDIFErrorFlag) return 0; - else { + else { if (status & HDSP_SPDIFSync) return 2; else @@ -3007,7 +3028,7 @@ static int hdsp_adatsync_sync_check(struct hdsp *hdsp) return 1; } else return 0; -} +} static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3025,17 +3046,17 @@ static int snd_hdsp_get_adatsync_sync_check(struct snd_kcontrol *kcontrol, struc } static int hdsp_adat_sync_check(struct hdsp *hdsp, int idx) -{ +{ int status = hdsp_read(hdsp, HDSP_statusRegister); - + if (status & (HDSP_Lock0>>idx)) { if (status & (HDSP_Sync0>>idx)) return 2; else - return 1; + return 1; } else return 0; -} +} static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -3053,7 +3074,7 @@ static int snd_hdsp_get_adat_sync_check(struct snd_kcontrol *kcontrol, struct sn break; case Multiface: case H9632: - if (offset >= 1) + if (offset >= 1) return -EINVAL; break; default: @@ -3115,7 +3136,7 @@ static int snd_hdsp_info_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ct static int snd_hdsp_get_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); - + ucontrol->value.enumerated.item[0] = hdsp_dds_offset(hdsp); return 0; } @@ -3125,7 +3146,7 @@ static int snd_hdsp_put_dds_offset(struct snd_kcontrol *kcontrol, struct snd_ctl struct hdsp *hdsp = snd_kcontrol_chip(kcontrol); int change; int val; - + if (!snd_hdsp_use_is_exclusive(hdsp)) return -EBUSY; val = ucontrol->value.enumerated.item[0]; @@ -3170,7 +3191,7 @@ static struct snd_kcontrol_new snd_hdsp_controls[] = { .get = snd_hdsp_control_spdif_mask_get, .private_value = IEC958_AES0_NONAUDIO | IEC958_AES0_PROFESSIONAL | - IEC958_AES0_CON_EMPHASIS, + IEC958_AES0_CON_EMPHASIS, }, { .access = SNDRV_CTL_ELEM_ACCESS_READ, @@ -3188,7 +3209,7 @@ HDSP_SPDIF_OUT("IEC958 Output also on ADAT1", 0), HDSP_SPDIF_PROFESSIONAL("IEC958 Professional Bit", 0), HDSP_SPDIF_EMPHASIS("IEC958 Emphasis Bit", 0), HDSP_SPDIF_NON_AUDIO("IEC958 Non-audio Bit", 0), -/* 'Sample Clock Source' complies with the alsa control naming scheme */ +/* 'Sample Clock Source' complies with the alsa control naming scheme */ HDSP_CLOCK_SOURCE("Sample Clock Source", 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -3240,7 +3261,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) return err; } } - + /* DA, AD and Phone gain and XLR breakout cable controls for H9632 cards */ if (hdsp->io_type == H9632) { for (idx = 0; idx < ARRAY_SIZE(snd_hdsp_9632_controls); idx++) { @@ -3259,7 +3280,7 @@ static int snd_hdsp_create_controls(struct snd_card *card, struct hdsp *hdsp) } /*------------------------------------------------------------ - /proc interface + /proc interface ------------------------------------------------------------*/ static void @@ -3298,7 +3319,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } } } - + status = hdsp_read(hdsp, HDSP_statusRegister); status2 = hdsp_read(hdsp, HDSP_status2Register); @@ -3362,17 +3383,17 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; case HDSP_CLOCK_SOURCE_INTERNAL_192KHZ: clock_source = "Internal 192 kHz"; - break; + break; default: - clock_source = "Error"; + clock_source = "Error"; } snd_iprintf (buffer, "Sample Clock Source: %s\n", clock_source); - + if (hdsp_system_clock_mode(hdsp)) system_clock_mode = "Slave"; else system_clock_mode = "Master"; - + switch (hdsp_pref_sync_ref (hdsp)) { case HDSP_SYNC_FROM_WORD: pref_sync_ref = "Word Clock"; @@ -3397,7 +3418,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf (buffer, "Preferred Sync Reference: %s\n", pref_sync_ref); - + switch (hdsp_autosync_ref (hdsp)) { case HDSP_AUTOSYNC_FROM_WORD: autosync_ref = "Word Clock"; @@ -3410,7 +3431,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; case HDSP_AUTOSYNC_FROM_NONE: autosync_ref = "None"; - break; + break; case HDSP_AUTOSYNC_FROM_ADAT1: autosync_ref = "ADAT1"; break; @@ -3425,14 +3446,14 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf (buffer, "AutoSync Reference: %s\n", autosync_ref); - + snd_iprintf (buffer, "AutoSync Frequency: %d\n", hdsp_external_sample_rate(hdsp)); - + snd_iprintf (buffer, "System Clock Mode: %s\n", system_clock_mode); snd_iprintf (buffer, "System Clock Frequency: %d\n", hdsp->system_sample_rate); snd_iprintf (buffer, "System Clock Locked: %s\n", hdsp->clock_source_locked ? "Yes" : "No"); - + snd_iprintf(buffer, "\n"); switch (hdsp_spdif_in(hdsp)) { @@ -3452,7 +3473,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "IEC958 input: ???\n"); break; } - + if (hdsp->control_register & HDSP_SPDIFOpticalOut) snd_iprintf(buffer, "IEC958 output: Coaxial & ADAT1\n"); else @@ -3510,13 +3531,13 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf (buffer, "SPDIF: No Lock\n"); else snd_iprintf (buffer, "SPDIF: %s\n", x ? "Sync" : "Lock"); - + x = status2 & HDSP_wc_sync; if (status2 & HDSP_wc_lock) snd_iprintf (buffer, "Word Clock: %s\n", x ? "Sync" : "Lock"); else snd_iprintf (buffer, "Word Clock: No Lock\n"); - + x = status & HDSP_TimecodeSync; if (status & HDSP_TimecodeLock) snd_iprintf(buffer, "ADAT Sync: %s\n", x ? "Sync" : "Lock"); @@ -3524,11 +3545,11 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) snd_iprintf(buffer, "ADAT Sync: No Lock\n"); snd_iprintf(buffer, "\n"); - + /* Informations about H9632 specific controls */ if (hdsp->io_type == H9632) { char *tmp; - + switch (hdsp_ad_gain(hdsp)) { case 0: tmp = "-10 dBV"; @@ -3554,7 +3575,7 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) break; } snd_iprintf(buffer, "DA Gain : %s\n", tmp); - + switch (hdsp_phone_gain(hdsp)) { case 0: tmp = "0 dB"; @@ -3568,8 +3589,8 @@ snd_hdsp_proc_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) } snd_iprintf(buffer, "Phones Gain : %s\n", tmp); - snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); - + snd_iprintf(buffer, "XLR Breakout Cable : %s\n", hdsp_xlr_breakout_cable(hdsp) ? "yes" : "no"); + if (hdsp->control_register & HDSP_AnalogExtensionBoard) snd_iprintf(buffer, "AEB : on (ADAT1 internal)\n"); else @@ -3632,18 +3653,18 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) /* set defaults: - SPDIF Input via Coax + SPDIF Input via Coax Master clock mode maximum latency (7 => 2^7 = 8192 samples, 64Kbyte buffer, which implies 2 4096 sample, 32Kbyte periods). - Enable line out. + Enable line out. */ - hdsp->control_register = HDSP_ClockModeMaster | - HDSP_SPDIFInputCoaxial | - hdsp_encode_latency(7) | + hdsp->control_register = HDSP_ClockModeMaster | + HDSP_SPDIFInputCoaxial | + hdsp_encode_latency(7) | HDSP_LineOut; - + hdsp_write(hdsp, HDSP_controlRegister, hdsp->control_register); @@ -3661,7 +3682,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) hdsp_compute_period_size(hdsp); /* silence everything */ - + for (i = 0; i < HDSP_MATRIX_MIXER_SIZE; ++i) hdsp->mixer_matrix[i] = MINUS_INFINITY_GAIN; @@ -3669,7 +3690,7 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) if (hdsp_write_gain (hdsp, i, MINUS_INFINITY_GAIN)) return -EIO; } - + /* H9632 specific defaults */ if (hdsp->io_type == H9632) { hdsp->control_register |= (HDSP_DAGainPlus4dBu | HDSP_ADGainPlus4dBu | HDSP_PhoneGain0dB); @@ -3687,12 +3708,12 @@ static int snd_hdsp_set_defaults(struct hdsp *hdsp) static void hdsp_midi_tasklet(unsigned long arg) { struct hdsp *hdsp = (struct hdsp *)arg; - + if (hdsp->midi[0].pending) snd_hdsp_midi_input_read (&hdsp->midi[0]); if (hdsp->midi[1].pending) snd_hdsp_midi_input_read (&hdsp->midi[1]); -} +} static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) { @@ -3704,7 +3725,7 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) unsigned int midi0status; unsigned int midi1status; int schedule = 0; - + status = hdsp_read(hdsp, HDSP_statusRegister); audio = status & HDSP_audioIRQPending; @@ -3718,15 +3739,18 @@ static irqreturn_t snd_hdsp_interrupt(int irq, void *dev_id) midi0status = hdsp_read (hdsp, HDSP_midiStatusIn0) & 0xff; midi1status = hdsp_read (hdsp, HDSP_midiStatusIn1) & 0xff; - + + if (!(hdsp->state & HDSP_InitializationComplete)) + return IRQ_HANDLED; + if (audio) { if (hdsp->capture_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream); - + if (hdsp->playback_substream) snd_pcm_period_elapsed(hdsp->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream); } - + if (midi0 && midi0status) { if (hdsp->use_midi_tasklet) { /* we disable interrupts for this input until processing is done */ @@ -3769,10 +3793,10 @@ static char *hdsp_channel_buffer_location(struct hdsp *hdsp, if (snd_BUG_ON(channel < 0 || channel >= hdsp->max_channels)) return NULL; - + if ((mapped_channel = hdsp->channel_map[channel]) < 0) return NULL; - + if (stream == SNDRV_PCM_STREAM_CAPTURE) return hdsp->capture_buffer + (mapped_channel * HDSP_CHANNEL_BUFFER_BYTES); else @@ -3965,7 +3989,7 @@ static int snd_hdsp_trigger(struct snd_pcm_substream *substream, int cmd) struct hdsp *hdsp = snd_pcm_substream_chip(substream); struct snd_pcm_substream *other; int running; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; @@ -4059,10 +4083,10 @@ static struct snd_pcm_hardware snd_hdsp_playback_subinfo = .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4088,10 +4112,10 @@ static struct snd_pcm_hardware snd_hdsp_capture_subinfo = .formats = SNDRV_PCM_FMTBIT_S32_LE, #endif .rates = (SNDRV_PCM_RATE_32000 | - SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_64000 | - SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_64000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000), .rate_min = 32000, .rate_max = 96000, @@ -4170,7 +4194,7 @@ static int snd_hdsp_hw_rule_in_channels_rate(struct snd_pcm_hw_params *params, .max = hdsp->qs_in_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_in_channels, @@ -4201,7 +4225,7 @@ static int snd_hdsp_hw_rule_out_channels_rate(struct snd_pcm_hw_params *params, .max = hdsp->qs_out_channels, .integer = 1, }; - return snd_interval_refine(c, &t); + return snd_interval_refine(c, &t); } else if (r->min > 48000 && r->max <= 96000) { struct snd_interval t = { .min = hdsp->ds_out_channels, @@ -4318,8 +4342,8 @@ static int snd_hdsp_playback_open(struct snd_pcm_substream *substream) if (hdsp->io_type == H9632) { runtime->hw.channels_min = hdsp->qs_out_channels; runtime->hw.channels_max = hdsp->ss_out_channels; - } - + } + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hdsp_hw_rule_out_channels, hdsp, SNDRV_PCM_HW_PARAM_CHANNELS, -1); @@ -4413,13 +4437,6 @@ static int snd_hdsp_capture_release(struct snd_pcm_substream *substream) return 0; } -static int snd_hdsp_hwdep_dummy_op(struct snd_hwdep *hw, struct file *file) -{ - /* we have nothing to initialize but the call is required */ - return 0; -} - - /* helper functions for copying meter values */ static inline int copy_u32_le(void __user *dest, void __iomem *src) { @@ -4536,7 +4553,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm hdsp->iobase + HDSP_playbackRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_playbackRmsLevel + i * 8)) return -EFAULT; - if (copy_u64_le(&peak_rms->input_rms[i], + if (copy_u64_le(&peak_rms->input_rms[i], hdsp->iobase + HDSP_inputRmsLevel + i * 8 + 4, hdsp->iobase + HDSP_inputRmsLevel + i * 8)) return -EFAULT; @@ -4546,7 +4563,7 @@ static int hdsp_get_peak(struct hdsp *hdsp, struct hdsp_peak_rms __user *peak_rm static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigned int cmd, unsigned long arg) { - struct hdsp *hdsp = (struct hdsp *)hw->private_data; + struct hdsp *hdsp = (struct hdsp *)hw->private_data; void __user *argp = (void __user *)arg; int err; @@ -4580,7 +4597,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne struct hdsp_config_info info; unsigned long flags; int i; - + err = hdsp_check_for_iobox(hdsp); if (err < 0) return err; @@ -4614,7 +4631,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne info.ad_gain = (unsigned char)hdsp_ad_gain(hdsp); info.phone_gain = (unsigned char)hdsp_phone_gain(hdsp); info.xlr_breakout_cable = (unsigned char)hdsp_xlr_breakout_cable(hdsp); - + } if (hdsp->io_type == H9632 || hdsp->io_type == H9652) info.analog_extension_board = (unsigned char)hdsp_aeb(hdsp); @@ -4625,7 +4642,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne } case SNDRV_HDSP_IOCTL_GET_9632_AEB: { struct hdsp_9632_aeb h9632_aeb; - + if (hdsp->io_type != H9632) return -EINVAL; h9632_aeb.aebi = hdsp->ss_in_channels - H9632_SS_CHANNELS; h9632_aeb.aebo = hdsp->ss_out_channels - H9632_SS_CHANNELS; @@ -4636,7 +4653,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne case SNDRV_HDSP_IOCTL_GET_VERSION: { struct hdsp_version hdsp_version; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; if (hdsp->io_type == Undefined) { if ((err = hdsp_get_iobox_version(hdsp)) < 0) @@ -4652,7 +4669,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne struct hdsp_firmware __user *firmware; u32 __user *firmware_data; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return -EINVAL; /* SNDRV_HDSP_IOCTL_GET_VERSION must have been called */ if (hdsp->io_type == Undefined) return -EINVAL; @@ -4665,25 +4682,25 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if (get_user(firmware_data, &firmware->firmware_data)) return -EFAULT; - + if (hdsp_check_for_iobox (hdsp)) return -EIO; if (copy_from_user(hdsp->firmware_cache, firmware_data, sizeof(hdsp->firmware_cache)) != 0) return -EFAULT; - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; - - snd_hdsp_initialize_channels(hdsp); + + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - + if ((err = snd_hdsp_create_alsa_devices(hdsp->card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error creating alsa devices\n"); return err; @@ -4730,18 +4747,16 @@ static int snd_hdsp_create_hwdep(struct snd_card *card, struct hdsp *hdsp) { struct snd_hwdep *hw; int err; - + if ((err = snd_hwdep_new(card, "HDSP hwdep", 0, &hw)) < 0) return err; - + hdsp->hwdep = hw; hw->private_data = hdsp; strcpy(hw->name, "HDSP hwdep interface"); - hw->ops.open = snd_hdsp_hwdep_dummy_op; hw->ops.ioctl = snd_hdsp_hwdep_ioctl; - hw->ops.release = snd_hdsp_hwdep_dummy_op; - + return 0; } @@ -4774,24 +4789,24 @@ static void snd_hdsp_9652_enable_mixer (struct hdsp *hdsp) static int snd_hdsp_enable_io (struct hdsp *hdsp) { int i; - + if (hdsp_fifo_wait (hdsp, 0, 100)) { snd_printk(KERN_ERR "Hammerfall-DSP: enable_io fifo_wait failed\n"); return -EIO; } - + for (i = 0; i < hdsp->max_channels; ++i) { hdsp_write (hdsp, HDSP_inputEnable + (4 * i), 1); hdsp_write (hdsp, HDSP_outputEnable + (4 * i), 1); } - + return 0; } static void snd_hdsp_initialize_channels(struct hdsp *hdsp) { int status, aebi_channels, aebo_channels; - + switch (hdsp->io_type) { case Digiface: hdsp->card_name = "RME Hammerfall DSP + Digiface"; @@ -4804,7 +4819,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_in_channels = hdsp->ss_out_channels = H9652_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = H9652_DS_CHANNELS; break; - + case H9632: status = hdsp_read(hdsp, HDSP_statusRegister); /* HDSP_AEBx bits are low when AEB are connected */ @@ -4824,7 +4839,7 @@ static void snd_hdsp_initialize_channels(struct hdsp *hdsp) hdsp->ss_in_channels = hdsp->ss_out_channels = MULTIFACE_SS_CHANNELS; hdsp->ds_in_channels = hdsp->ds_out_channels = MULTIFACE_DS_CHANNELS; break; - + default: /* should never get here */ break; @@ -4840,12 +4855,12 @@ static void snd_hdsp_initialize_midi_flush (struct hdsp *hdsp) static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp) { int err; - + if ((err = snd_hdsp_create_pcm(card, hdsp)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating pcm interface\n"); return err; } - + if ((err = snd_hdsp_create_midi(card, hdsp, 0)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: Error creating first midi interface\n"); @@ -4876,19 +4891,19 @@ static int snd_hdsp_create_alsa_devices(struct snd_card *card, struct hdsp *hdsp snd_printk(KERN_ERR "Hammerfall-DSP: Error setting default values\n"); return err; } - + if (!(hdsp->state & HDSP_InitializationComplete)) { strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); - + if ((err = snd_card_register(card)) < 0) { snd_printk(KERN_ERR "Hammerfall-DSP: error registering card\n"); return err; } hdsp->state |= HDSP_InitializationComplete; } - + return 0; } @@ -4899,7 +4914,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) const char *fwfile; const struct firmware *fw; int err; - + if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; if (hdsp->io_type == Undefined) { @@ -4908,7 +4923,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) if (hdsp->io_type == H9652 || hdsp->io_type == H9632) return 0; } - + /* caution: max length of firmware filename is 30! */ switch (hdsp->io_type) { case Multiface: @@ -4942,12 +4957,12 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) memcpy(hdsp->firmware_cache, fw->data, sizeof(hdsp->firmware_cache)); release_firmware(fw); - + hdsp->state |= HDSP_FirmwareCached; if ((err = snd_hdsp_load_firmware_from_cache(hdsp)) < 0) return err; - + if (!(hdsp->state & HDSP_InitializationComplete)) { if ((err = snd_hdsp_enable_io(hdsp)) < 0) return err; @@ -4994,14 +5009,14 @@ static int __devinit snd_hdsp_create(struct snd_card *card, hdsp->max_channels = 26; hdsp->card = card; - + spin_lock_init(&hdsp->lock); tasklet_init(&hdsp->midi_tasklet, hdsp_midi_tasklet, (unsigned long)hdsp); - + pci_read_config_word(hdsp->pci, PCI_CLASS_REVISION, &hdsp->firmware_rev); hdsp->firmware_rev &= 0xff; - + /* From Martin Bjoernsen : "It is important that the card's latency timer register in the PCI configuration space is set to a value much larger @@ -5010,7 +5025,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, to its maximum 255 to avoid problems with some computers." */ pci_write_config_byte(hdsp->pci, PCI_LATENCY_TIMER, 0xFF); - + strcpy(card->driver, "H-DSP"); strcpy(card->mixername, "Xilinx FPGA"); @@ -5024,7 +5039,7 @@ static int __devinit snd_hdsp_create(struct snd_card *card, } else { hdsp->card_name = "RME HDSP 9632"; hdsp->max_channels = 16; - is_9632 = 1; + is_9632 = 1; } if ((err = pci_enable_device(pci)) < 0) @@ -5053,12 +5068,12 @@ static int __devinit snd_hdsp_create(struct snd_card *card, if ((err = snd_hdsp_initialize_memory(hdsp)) < 0) return err; - + if (!is_9652 && !is_9632) { - /* we wait 2 seconds to let freshly inserted cardbus cards do their hardware init */ - ssleep(2); + /* we wait a maximum of 10 seconds to let freshly + * inserted cardbus cards do their hardware init */ + err = hdsp_wait_for_iobox(hdsp, 1000, 10); - err = hdsp_check_for_iobox(hdsp); if (err < 0) return err; @@ -5080,35 +5095,35 @@ static int __devinit snd_hdsp_create(struct snd_card *card, return err; return 0; } else { - snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); + snd_printk(KERN_INFO "Hammerfall-DSP: Firmware already present, initializing card.\n"); if (hdsp_read(hdsp, HDSP_status2Register) & HDSP_version1) hdsp->io_type = Multiface; - else + else hdsp->io_type = Digiface; } } - + if ((err = snd_hdsp_enable_io(hdsp)) != 0) return err; - + if (is_9652) hdsp->io_type = H9652; - + if (is_9632) hdsp->io_type = H9632; if ((err = snd_hdsp_create_hwdep(card, hdsp)) < 0) return err; - + snd_hdsp_initialize_channels(hdsp); snd_hdsp_initialize_midi_flush(hdsp); - hdsp->state |= HDSP_FirmwareLoaded; + hdsp->state |= HDSP_FirmwareLoaded; if ((err = snd_hdsp_create_alsa_devices(card, hdsp)) < 0) return err; - return 0; + return 0; } static int snd_hdsp_free(struct hdsp *hdsp) @@ -5124,13 +5139,13 @@ static int snd_hdsp_free(struct hdsp *hdsp) free_irq(hdsp->irq, (void *)hdsp); snd_hdsp_free_buffers(hdsp); - + if (hdsp->iobase) iounmap(hdsp->iobase); if (hdsp->port) pci_release_regions(hdsp->pci); - + pci_disable_device(hdsp->pci); return 0; } @@ -5175,7 +5190,7 @@ static int __devinit snd_hdsp_probe(struct pci_dev *pci, } strcpy(card->shortname, "Hammerfall DSP"); - sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, + sprintf(card->longname, "%s at 0x%lx, irq %d", hdsp->card_name, hdsp->port, hdsp->irq); if ((err = snd_card_register(card)) < 0) { diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d4b4e0d..bac2dc0 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -4100,13 +4100,6 @@ static int snd_hdspm_capture_release(struct snd_pcm_substream *substream) return 0; } -static int snd_hdspm_hwdep_dummy_op(struct snd_hwdep * hw, struct file *file) -{ - /* we have nothing to initialize but the call is required */ - return 0; -} - - static int snd_hdspm_hwdep_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg) { @@ -4213,9 +4206,7 @@ static int __devinit snd_hdspm_create_hwdep(struct snd_card *card, hw->private_data = hdspm; strcpy(hw->name, "HDSPM hwdep interface"); - hw->ops.open = snd_hdspm_hwdep_dummy_op; hw->ops.ioctl = snd_hdspm_hwdep_ioctl; - hw->ops.release = snd_hdspm_hwdep_dummy_op; return 0; } diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index baf6d8e..1a5ff06 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1300,7 +1300,7 @@ static int __devinit sis_chip_create(struct snd_card *card, if (rc) goto error_out; - if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) { printk(KERN_ERR "sis7019: architecture does not support " "30-bit PCI busmaster DMA"); goto error_out_enabled; diff --git a/sound/pci/sonicvibes.c b/sound/pci/sonicvibes.c index c5601b0..7dc60ad 100644 --- a/sound/pci/sonicvibes.c +++ b/sound/pci/sonicvibes.c @@ -273,7 +273,8 @@ static inline void snd_sonicvibes_setdmaa(struct sonicvibes * sonic, outl(count, sonic->dmaa_port + SV_DMA_COUNT0); outb(0x18, sonic->dmaa_port + SV_DMA_MODE); #if 0 - printk("program dmaa: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmaa_port + SV_DMA_ADDR0)); + printk(KERN_DEBUG "program dmaa: addr = 0x%x, paddr = 0x%x\n", + addr, inl(sonic->dmaa_port + SV_DMA_ADDR0)); #endif } @@ -288,7 +289,8 @@ static inline void snd_sonicvibes_setdmac(struct sonicvibes * sonic, outl(count, sonic->dmac_port + SV_DMA_COUNT0); outb(0x14, sonic->dmac_port + SV_DMA_MODE); #if 0 - printk("program dmac: addr = 0x%x, paddr = 0x%x\n", addr, inl(sonic->dmac_port + SV_DMA_ADDR0)); + printk(KERN_DEBUG "program dmac: addr = 0x%x, paddr = 0x%x\n", + addr, inl(sonic->dmac_port + SV_DMA_ADDR0)); #endif } @@ -355,71 +357,104 @@ static unsigned char snd_sonicvibes_in(struct sonicvibes * sonic, unsigned char #if 0 static void snd_sonicvibes_debug(struct sonicvibes * sonic) { - printk("SV REGS: INDEX = 0x%02x ", inb(SV_REG(sonic, INDEX))); + printk(KERN_DEBUG + "SV REGS: INDEX = 0x%02x ", inb(SV_REG(sonic, INDEX))); printk(" STATUS = 0x%02x\n", inb(SV_REG(sonic, STATUS))); - printk(" 0x00: left input = 0x%02x ", snd_sonicvibes_in(sonic, 0x00)); + printk(KERN_DEBUG + " 0x00: left input = 0x%02x ", snd_sonicvibes_in(sonic, 0x00)); printk(" 0x20: synth rate low = 0x%02x\n", snd_sonicvibes_in(sonic, 0x20)); - printk(" 0x01: right input = 0x%02x ", snd_sonicvibes_in(sonic, 0x01)); + printk(KERN_DEBUG + " 0x01: right input = 0x%02x ", snd_sonicvibes_in(sonic, 0x01)); printk(" 0x21: synth rate high = 0x%02x\n", snd_sonicvibes_in(sonic, 0x21)); - printk(" 0x02: left AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x02)); + printk(KERN_DEBUG + " 0x02: left AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x02)); printk(" 0x22: ADC clock = 0x%02x\n", snd_sonicvibes_in(sonic, 0x22)); - printk(" 0x03: right AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x03)); + printk(KERN_DEBUG + " 0x03: right AUX1 = 0x%02x ", snd_sonicvibes_in(sonic, 0x03)); printk(" 0x23: ADC alt rate = 0x%02x\n", snd_sonicvibes_in(sonic, 0x23)); - printk(" 0x04: left CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x04)); + printk(KERN_DEBUG + " 0x04: left CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x04)); printk(" 0x24: ADC pll M = 0x%02x\n", snd_sonicvibes_in(sonic, 0x24)); - printk(" 0x05: right CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x05)); + printk(KERN_DEBUG + " 0x05: right CD = 0x%02x ", snd_sonicvibes_in(sonic, 0x05)); printk(" 0x25: ADC pll N = 0x%02x\n", snd_sonicvibes_in(sonic, 0x25)); - printk(" 0x06: left line = 0x%02x ", snd_sonicvibes_in(sonic, 0x06)); + printk(KERN_DEBUG + " 0x06: left line = 0x%02x ", snd_sonicvibes_in(sonic, 0x06)); printk(" 0x26: Synth pll M = 0x%02x\n", snd_sonicvibes_in(sonic, 0x26)); - printk(" 0x07: right line = 0x%02x ", snd_sonicvibes_in(sonic, 0x07)); + printk(KERN_DEBUG + " 0x07: right line = 0x%02x ", snd_sonicvibes_in(sonic, 0x07)); printk(" 0x27: Synth pll N = 0x%02x\n", snd_sonicvibes_in(sonic, 0x27)); - printk(" 0x08: MIC = 0x%02x ", snd_sonicvibes_in(sonic, 0x08)); + printk(KERN_DEBUG + " 0x08: MIC = 0x%02x ", snd_sonicvibes_in(sonic, 0x08)); printk(" 0x28: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x28)); - printk(" 0x09: Game port = 0x%02x ", snd_sonicvibes_in(sonic, 0x09)); + printk(KERN_DEBUG + " 0x09: Game port = 0x%02x ", snd_sonicvibes_in(sonic, 0x09)); printk(" 0x29: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x29)); - printk(" 0x0a: left synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0a)); + printk(KERN_DEBUG + " 0x0a: left synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0a)); printk(" 0x2a: MPU401 = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2a)); - printk(" 0x0b: right synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0b)); + printk(KERN_DEBUG + " 0x0b: right synth = 0x%02x ", snd_sonicvibes_in(sonic, 0x0b)); printk(" 0x2b: drive ctrl = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2b)); - printk(" 0x0c: left AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0c)); + printk(KERN_DEBUG + " 0x0c: left AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0c)); printk(" 0x2c: SRS space = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2c)); - printk(" 0x0d: right AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0d)); + printk(KERN_DEBUG + " 0x0d: right AUX2 = 0x%02x ", snd_sonicvibes_in(sonic, 0x0d)); printk(" 0x2d: SRS center = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2d)); - printk(" 0x0e: left analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0e)); + printk(KERN_DEBUG + " 0x0e: left analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0e)); printk(" 0x2e: wave source = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2e)); - printk(" 0x0f: right analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0f)); + printk(KERN_DEBUG + " 0x0f: right analog = 0x%02x ", snd_sonicvibes_in(sonic, 0x0f)); printk(" 0x2f: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x2f)); - printk(" 0x10: left PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x10)); + printk(KERN_DEBUG + " 0x10: left PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x10)); printk(" 0x30: analog power = 0x%02x\n", snd_sonicvibes_in(sonic, 0x30)); - printk(" 0x11: right PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x11)); + printk(KERN_DEBUG + " 0x11: right PCM = 0x%02x ", snd_sonicvibes_in(sonic, 0x11)); printk(" 0x31: analog power = 0x%02x\n", snd_sonicvibes_in(sonic, 0x31)); - printk(" 0x12: DMA data format = 0x%02x ", snd_sonicvibes_in(sonic, 0x12)); + printk(KERN_DEBUG + " 0x12: DMA data format = 0x%02x ", snd_sonicvibes_in(sonic, 0x12)); printk(" 0x32: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x32)); - printk(" 0x13: P/C enable = 0x%02x ", snd_sonicvibes_in(sonic, 0x13)); + printk(KERN_DEBUG + " 0x13: P/C enable = 0x%02x ", snd_sonicvibes_in(sonic, 0x13)); printk(" 0x33: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x33)); - printk(" 0x14: U/D button = 0x%02x ", snd_sonicvibes_in(sonic, 0x14)); + printk(KERN_DEBUG + " 0x14: U/D button = 0x%02x ", snd_sonicvibes_in(sonic, 0x14)); printk(" 0x34: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x34)); - printk(" 0x15: revision = 0x%02x ", snd_sonicvibes_in(sonic, 0x15)); + printk(KERN_DEBUG + " 0x15: revision = 0x%02x ", snd_sonicvibes_in(sonic, 0x15)); printk(" 0x35: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x35)); - printk(" 0x16: ADC output ctrl = 0x%02x ", snd_sonicvibes_in(sonic, 0x16)); + printk(KERN_DEBUG + " 0x16: ADC output ctrl = 0x%02x ", snd_sonicvibes_in(sonic, 0x16)); printk(" 0x36: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x36)); - printk(" 0x17: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x17)); + printk(KERN_DEBUG + " 0x17: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x17)); printk(" 0x37: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x37)); - printk(" 0x18: DMA A upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x18)); + printk(KERN_DEBUG + " 0x18: DMA A upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x18)); printk(" 0x38: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x38)); - printk(" 0x19: DMA A lower cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x19)); + printk(KERN_DEBUG + " 0x19: DMA A lower cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x19)); printk(" 0x39: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x39)); - printk(" 0x1a: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1a)); + printk(KERN_DEBUG + " 0x1a: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1a)); printk(" 0x3a: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3a)); - printk(" 0x1b: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1b)); + printk(KERN_DEBUG + " 0x1b: --- = 0x%02x ", snd_sonicvibes_in(sonic, 0x1b)); printk(" 0x3b: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3b)); - printk(" 0x1c: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1c)); + printk(KERN_DEBUG + " 0x1c: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1c)); printk(" 0x3c: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3c)); - printk(" 0x1d: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1d)); + printk(KERN_DEBUG + " 0x1d: DMA C upper cnt = 0x%02x ", snd_sonicvibes_in(sonic, 0x1d)); printk(" 0x3d: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3d)); - printk(" 0x1e: PCM rate low = 0x%02x ", snd_sonicvibes_in(sonic, 0x1e)); + printk(KERN_DEBUG + " 0x1e: PCM rate low = 0x%02x ", snd_sonicvibes_in(sonic, 0x1e)); printk(" 0x3e: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3e)); - printk(" 0x1f: PCM rate high = 0x%02x ", snd_sonicvibes_in(sonic, 0x1f)); + printk(KERN_DEBUG + " 0x1f: PCM rate high = 0x%02x ", snd_sonicvibes_in(sonic, 0x1f)); printk(" 0x3f: --- = 0x%02x\n", snd_sonicvibes_in(sonic, 0x3f)); } @@ -476,8 +511,8 @@ static void snd_sonicvibes_pll(unsigned int rate, *res_m = m; *res_n = n; #if 0 - printk("metric = %i, xm = %i, xn = %i\n", metric, xm, xn); - printk("pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n); + printk(KERN_DEBUG "metric = %i, xm = %i, xn = %i\n", metric, xm, xn); + printk(KERN_DEBUG "pll: m = 0x%x, r = 0x%x, n = 0x%x\n", reg, m, r, n); #endif } @@ -1229,8 +1264,8 @@ static int __devinit snd_sonicvibes_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; /* check, if we can restrict PCI DMA transfers to 24 bits */ - if (pci_set_dma_mask(pci, DMA_24BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_24BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(24)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(24)) < 0) { snd_printk(KERN_ERR "architecture does not support 24bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/trident/trident_main.c b/sound/pci/trident/trident_main.c index c612b43..6d943f6 100644 --- a/sound/pci/trident/trident_main.c +++ b/sound/pci/trident/trident_main.c @@ -68,40 +68,40 @@ static void snd_trident_print_voice_regs(struct snd_trident *trident, int voice) { unsigned int val, tmp; - printk("Trident voice %i:\n", voice); + printk(KERN_DEBUG "Trident voice %i:\n", voice); outb(voice, TRID_REG(trident, T4D_LFO_GC_CIR)); val = inl(TRID_REG(trident, CH_LBA)); - printk("LBA: 0x%x\n", val); + printk(KERN_DEBUG "LBA: 0x%x\n", val); val = inl(TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC)); - printk("GVSel: %i\n", val >> 31); - printk("Pan: 0x%x\n", (val >> 24) & 0x7f); - printk("Vol: 0x%x\n", (val >> 16) & 0xff); - printk("CTRL: 0x%x\n", (val >> 12) & 0x0f); - printk("EC: 0x%x\n", val & 0x0fff); + printk(KERN_DEBUG "GVSel: %i\n", val >> 31); + printk(KERN_DEBUG "Pan: 0x%x\n", (val >> 24) & 0x7f); + printk(KERN_DEBUG "Vol: 0x%x\n", (val >> 16) & 0xff); + printk(KERN_DEBUG "CTRL: 0x%x\n", (val >> 12) & 0x0f); + printk(KERN_DEBUG "EC: 0x%x\n", val & 0x0fff); if (trident->device != TRIDENT_DEVICE_ID_NX) { val = inl(TRID_REG(trident, CH_DX_CSO_ALPHA_FMS)); - printk("CSO: 0x%x\n", val >> 16); + printk(KERN_DEBUG "CSO: 0x%x\n", val >> 16); printk("Alpha: 0x%x\n", (val >> 4) & 0x0fff); - printk("FMS: 0x%x\n", val & 0x0f); + printk(KERN_DEBUG "FMS: 0x%x\n", val & 0x0f); val = inl(TRID_REG(trident, CH_DX_ESO_DELTA)); - printk("ESO: 0x%x\n", val >> 16); - printk("Delta: 0x%x\n", val & 0xffff); + printk(KERN_DEBUG "ESO: 0x%x\n", val >> 16); + printk(KERN_DEBUG "Delta: 0x%x\n", val & 0xffff); val = inl(TRID_REG(trident, CH_DX_FMC_RVOL_CVOL)); } else { // TRIDENT_DEVICE_ID_NX val = inl(TRID_REG(trident, CH_NX_DELTA_CSO)); tmp = (val >> 24) & 0xff; - printk("CSO: 0x%x\n", val & 0x00ffffff); + printk(KERN_DEBUG "CSO: 0x%x\n", val & 0x00ffffff); val = inl(TRID_REG(trident, CH_NX_DELTA_ESO)); tmp |= (val >> 16) & 0xff00; - printk("Delta: 0x%x\n", tmp); - printk("ESO: 0x%x\n", val & 0x00ffffff); + printk(KERN_DEBUG "Delta: 0x%x\n", tmp); + printk(KERN_DEBUG "ESO: 0x%x\n", val & 0x00ffffff); val = inl(TRID_REG(trident, CH_NX_ALPHA_FMS_FMC_RVOL_CVOL)); - printk("Alpha: 0x%x\n", val >> 20); - printk("FMS: 0x%x\n", (val >> 16) & 0x0f); + printk(KERN_DEBUG "Alpha: 0x%x\n", val >> 20); + printk(KERN_DEBUG "FMS: 0x%x\n", (val >> 16) & 0x0f); } - printk("FMC: 0x%x\n", (val >> 14) & 3); - printk("RVol: 0x%x\n", (val >> 7) & 0x7f); - printk("CVol: 0x%x\n", val & 0x7f); + printk(KERN_DEBUG "FMC: 0x%x\n", (val >> 14) & 3); + printk(KERN_DEBUG "RVol: 0x%x\n", (val >> 7) & 0x7f); + printk(KERN_DEBUG "CVol: 0x%x\n", val & 0x7f); } #endif @@ -496,12 +496,17 @@ void snd_trident_write_voice_regs(struct snd_trident * trident, outl(regs[4], TRID_REG(trident, CH_START + 16)); #if 0 - printk("written %i channel:\n", voice->number); - printk(" regs[0] = 0x%x/0x%x\n", regs[0], inl(TRID_REG(trident, CH_START + 0))); - printk(" regs[1] = 0x%x/0x%x\n", regs[1], inl(TRID_REG(trident, CH_START + 4))); - printk(" regs[2] = 0x%x/0x%x\n", regs[2], inl(TRID_REG(trident, CH_START + 8))); - printk(" regs[3] = 0x%x/0x%x\n", regs[3], inl(TRID_REG(trident, CH_START + 12))); - printk(" regs[4] = 0x%x/0x%x\n", regs[4], inl(TRID_REG(trident, CH_START + 16))); + printk(KERN_DEBUG "written %i channel:\n", voice->number); + printk(KERN_DEBUG " regs[0] = 0x%x/0x%x\n", + regs[0], inl(TRID_REG(trident, CH_START + 0))); + printk(KERN_DEBUG " regs[1] = 0x%x/0x%x\n", + regs[1], inl(TRID_REG(trident, CH_START + 4))); + printk(KERN_DEBUG " regs[2] = 0x%x/0x%x\n", + regs[2], inl(TRID_REG(trident, CH_START + 8))); + printk(KERN_DEBUG " regs[3] = 0x%x/0x%x\n", + regs[3], inl(TRID_REG(trident, CH_START + 12))); + printk(KERN_DEBUG " regs[4] = 0x%x/0x%x\n", + regs[4], inl(TRID_REG(trident, CH_START + 16))); #endif } @@ -583,7 +588,7 @@ static void snd_trident_write_vol_reg(struct snd_trident * trident, outb(voice->Vol >> 2, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC + 2)); break; case TRIDENT_DEVICE_ID_SI7018: - // printk("voice->Vol = 0x%x\n", voice->Vol); + /* printk(KERN_DEBUG "voice->Vol = 0x%x\n", voice->Vol); */ outw((voice->CTRL << 12) | voice->Vol, TRID_REG(trident, CH_GVSEL_PAN_VOL_CTRL_EC)); break; @@ -3554,8 +3559,8 @@ int __devinit snd_trident_create(struct snd_card *card, if ((err = pci_enable_device(pci)) < 0) return err; /* check, if we can restrict PCI DMA transfers to 30 bits */ - if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0 || - pci_set_consistent_dma_mask(pci, DMA_30BIT_MASK) < 0) { + if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0 || + pci_set_consistent_dma_mask(pci, DMA_BIT_MASK(30)) < 0) { snd_printk(KERN_ERR "architecture does not support 30bit PCI busmaster DMA\n"); pci_disable_device(pci); return -ENXIO; diff --git a/sound/pci/via82xx.c b/sound/pci/via82xx.c index d870554..809b233 100644 --- a/sound/pci/via82xx.c +++ b/sound/pci/via82xx.c @@ -466,7 +466,10 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre flag = VIA_TBL_BIT_FLAG; /* period boundary */ } else flag = 0; /* period continues to the next */ - // printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); + /* + printk(KERN_DEBUG "via: tbl %d: at %d size %d " + "(rest %d)\n", idx, ofs, r, rest); + */ ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; @@ -2360,14 +2363,14 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = { SND_PCI_QUIRK(0x1019, 0x0996, "ESC Mobo", VIA_DXS_48K), SND_PCI_QUIRK(0x1019, 0x0a81, "ECS K7VTA3 v8.0", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x1019, 0x0a85, "ECS L7VMM2", VIA_DXS_NO_VRA), - SND_PCI_QUIRK(0x1019, 0, "ESC K8", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1019, "ESC K8", VIA_DXS_SRC), SND_PCI_QUIRK(0x1019, 0xaa01, "ESC K8T890-A", VIA_DXS_SRC), SND_PCI_QUIRK(0x1025, 0x0033, "Acer Inspire 1353LM", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x1025, 0x0046, "Acer Aspire 1524 WLMi", VIA_DXS_SRC), - SND_PCI_QUIRK(0x1043, 0, "ASUS A7/A8", VIA_DXS_NO_VRA), - SND_PCI_QUIRK(0x1071, 0, "Diverse Notebook", VIA_DXS_NO_VRA), + SND_PCI_QUIRK_VENDOR(0x1043, "ASUS A7/A8", VIA_DXS_NO_VRA), + SND_PCI_QUIRK_VENDOR(0x1071, "Diverse Notebook", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x10cf, 0x118e, "FSC Laptop", VIA_DXS_ENABLE), - SND_PCI_QUIRK(0x1106, 0, "ASRock", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1106, "ASRock", VIA_DXS_SRC), SND_PCI_QUIRK(0x1297, 0xa231, "Shuttle AK31v2", VIA_DXS_SRC), SND_PCI_QUIRK(0x1297, 0xa232, "Shuttle", VIA_DXS_SRC), SND_PCI_QUIRK(0x1297, 0xc160, "Shuttle Sk41G", VIA_DXS_SRC), @@ -2375,7 +2378,7 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = { SND_PCI_QUIRK(0x1462, 0x3800, "MSI KT266", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1462, 0x7120, "MSI KT4V", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1462, 0x7142, "MSI K8MM-V", VIA_DXS_ENABLE), - SND_PCI_QUIRK(0x1462, 0, "MSI Mobo", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1462, "MSI Mobo", VIA_DXS_SRC), SND_PCI_QUIRK(0x147b, 0x1401, "ABIT KD7(-RAID)", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x147b, 0x1411, "ABIT VA-20", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x147b, 0x1413, "ABIT KV8 Pro", VIA_DXS_ENABLE), @@ -2389,11 +2392,11 @@ static struct snd_pci_quirk dxs_whitelist[] __devinitdata = { SND_PCI_QUIRK(0x161f, 0x2032, "m680x machines", VIA_DXS_48K), SND_PCI_QUIRK(0x1631, 0xe004, "PB EasyNote 3174", VIA_DXS_ENABLE), SND_PCI_QUIRK(0x1695, 0x3005, "EPoX EP-8K9A", VIA_DXS_ENABLE), - SND_PCI_QUIRK(0x1695, 0, "EPoX mobo", VIA_DXS_SRC), - SND_PCI_QUIRK(0x16f3, 0, "Jetway K8", VIA_DXS_SRC), - SND_PCI_QUIRK(0x1734, 0, "FSC Laptop", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1695, "EPoX mobo", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x16f3, "Jetway K8", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1734, "FSC Laptop", VIA_DXS_SRC), SND_PCI_QUIRK(0x1849, 0x3059, "ASRock K7VM2", VIA_DXS_NO_VRA), - SND_PCI_QUIRK(0x1849, 0, "ASRock mobo", VIA_DXS_SRC), + SND_PCI_QUIRK_VENDOR(0x1849, "ASRock mobo", VIA_DXS_SRC), SND_PCI_QUIRK(0x1919, 0x200a, "Soltek SL-K8", VIA_DXS_NO_VRA), SND_PCI_QUIRK(0x4005, 0x4710, "MSI K7T266", VIA_DXS_SRC), { } /* terminator */ diff --git a/sound/pci/via82xx_modem.c b/sound/pci/via82xx_modem.c index c086b76..0d54e35 100644 --- a/sound/pci/via82xx_modem.c +++ b/sound/pci/via82xx_modem.c @@ -328,7 +328,10 @@ static int build_via_table(struct viadev *dev, struct snd_pcm_substream *substre flag = VIA_TBL_BIT_FLAG; /* period boundary */ } else flag = 0; /* period continues to the next */ - // printk("via: tbl %d: at %d size %d (rest %d)\n", idx, ofs, r, rest); + /* + printk(KERN_DEBUG "via: tbl %d: at %d size %d " + "(rest %d)\n", idx, ofs, r, rest); + */ ((u32 *)dev->table.area)[(idx<<1) + 1] = cpu_to_le32(r | flag); dev->idx_table[idx].offset = ofs; dev->idx_table[idx].size = r; diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index 7e87f39..c0efe44 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -107,7 +107,9 @@ static unsigned char vx2_inb(struct vx_core *chip, int offset) static void vx2_outb(struct vx_core *chip, int offset, unsigned char val) { outb(val, vx2_reg_addr(chip, offset)); - //printk("outb: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + /* + printk(KERN_DEBUG "outb: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + */ } /** @@ -126,7 +128,9 @@ static unsigned int vx2_inl(struct vx_core *chip, int offset) */ static void vx2_outl(struct vx_core *chip, int offset, unsigned int val) { - // printk("outl: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + /* + printk(KERN_DEBUG "outl: %x -> %x\n", val, vx2_reg_addr(chip, offset)); + */ outl(val, vx2_reg_addr(chip, offset)); } diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index 90d0d62..2f09252 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -318,7 +318,12 @@ static void snd_ymfpci_pcm_interrupt(struct snd_ymfpci *chip, struct snd_ymfpci_ ypcm->period_pos += delta; ypcm->last_pos = pos; if (ypcm->period_pos >= ypcm->period_size) { - // printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start); + /* + printk(KERN_DEBUG + "done - active_bank = 0x%x, start = 0x%x\n", + chip->active_bank, + voice->bank[chip->active_bank].start); + */ ypcm->period_pos %= ypcm->period_size; spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(ypcm->substream); @@ -366,7 +371,12 @@ static void snd_ymfpci_pcm_capture_interrupt(struct snd_pcm_substream *substream ypcm->last_pos = pos; if (ypcm->period_pos >= ypcm->period_size) { ypcm->period_pos %= ypcm->period_size; - // printk("done - active_bank = 0x%x, start = 0x%x\n", chip->active_bank, voice->bank[chip->active_bank].start); + /* + printk(KERN_DEBUG + "done - active_bank = 0x%x, start = 0x%x\n", + chip->active_bank, + voice->bank[chip->active_bank].start); + */ spin_unlock(&chip->reg_lock); snd_pcm_period_elapsed(substream); spin_lock(&chip->reg_lock); diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c index dfa40b0..5d2afa0 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c @@ -82,14 +82,21 @@ static void pdacf_ak4117_write(void *private_data, unsigned char reg, unsigned c #if 0 void pdacf_dump(struct snd_pdacf *chip) { - printk("PDAUDIOCF DUMP (0x%lx):\n", chip->port); - printk("WPD : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_WDP)); - printk("RDP : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_RDP)); - printk("TCR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_TCR)); - printk("SCR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_SCR)); - printk("ISR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_ISR)); - printk("IER : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_IER)); - printk("AK_IFR : 0x%x\n", inw(chip->port + PDAUDIOCF_REG_AK_IFR)); + printk(KERN_DEBUG "PDAUDIOCF DUMP (0x%lx):\n", chip->port); + printk(KERN_DEBUG "WPD : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_WDP)); + printk(KERN_DEBUG "RDP : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_RDP)); + printk(KERN_DEBUG "TCR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_TCR)); + printk(KERN_DEBUG "SCR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_SCR)); + printk(KERN_DEBUG "ISR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_ISR)); + printk(KERN_DEBUG "IER : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_IER)); + printk(KERN_DEBUG "AK_IFR : 0x%x\n", + inw(chip->port + PDAUDIOCF_REG_AK_IFR)); } #endif diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c index ea903c8..dcd3220 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_irq.c @@ -269,7 +269,7 @@ void pdacf_tasklet(unsigned long private_data) rdp = inw(chip->port + PDAUDIOCF_REG_RDP); wdp = inw(chip->port + PDAUDIOCF_REG_WDP); - // printk("TASKLET: rdp = %x, wdp = %x\n", rdp, wdp); + /* printk(KERN_DEBUG "TASKLET: rdp = %x, wdp = %x\n", rdp, wdp); */ size = wdp - rdp; if (size < 0) size += 0x10000; @@ -321,5 +321,5 @@ void pdacf_tasklet(unsigned long private_data) spin_lock(&chip->reg_lock); } spin_unlock(&chip->reg_lock); - // printk("TASKLET: end\n"); + /* printk(KERN_DEBUG "TASKLET: end\n"); */ } diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig index 777de2b..bd2338a 100644 --- a/sound/ppc/Kconfig +++ b/sound/ppc/Kconfig @@ -13,6 +13,7 @@ config SND_POWERMAC tristate "PowerMac (AWACS, DACA, Burgundy, Tumbler, Keywest)" depends on I2C && INPUT && PPC_PMAC select SND_PCM + select SND_VMASTER help Say Y here to include support for the integrated sound device. diff --git a/sound/ppc/awacs.c b/sound/ppc/awacs.c index 7bd33e6..80df9b1 100644 --- a/sound/ppc/awacs.c +++ b/sound/ppc/awacs.c @@ -608,9 +608,12 @@ static struct snd_kcontrol_new snd_pmac_screamer_mixers_beige[] __initdata = { AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_LINE, 0), }; -static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = { +static struct snd_kcontrol_new snd_pmac_screamer_mixers_lo[] __initdata = { AWACS_VOLUME("Line out Playback Volume", 2, 6, 1), - AWACS_VOLUME("Master Playback Volume", 5, 6, 1), +}; + +static struct snd_kcontrol_new snd_pmac_screamer_mixers_imac[] __initdata = { + AWACS_VOLUME("Play-through Playback Volume", 5, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), }; @@ -627,6 +630,10 @@ static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac7500[] __initdata = { AWACS_SWITCH("Line Capture Switch", 0, SHIFT_MUX_MIC, 0), }; +static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac5500[] __initdata = { + AWACS_VOLUME("Headphone Playback Volume", 2, 6, 1), +}; + static struct snd_kcontrol_new snd_pmac_awacs_mixers_pmac[] __initdata = { AWACS_VOLUME("Master Playback Volume", 2, 6, 1), AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), @@ -645,12 +652,19 @@ static struct snd_kcontrol_new snd_pmac_screamer_mixers2[] __initdata = { AWACS_SWITCH("Mic Capture Switch", 0, SHIFT_MUX_LINE, 0), }; +static struct snd_kcontrol_new snd_pmac_awacs_mixers2_pmac5500[] __initdata = { + AWACS_SWITCH("CD Capture Switch", 0, SHIFT_MUX_CD, 0), +}; + static struct snd_kcontrol_new snd_pmac_awacs_master_sw __initdata = AWACS_SWITCH("Master Playback Switch", 1, SHIFT_HDMUTE, 1); static struct snd_kcontrol_new snd_pmac_awacs_master_sw_imac __initdata = AWACS_SWITCH("Line out Playback Switch", 1, SHIFT_HDMUTE, 1); +static struct snd_kcontrol_new snd_pmac_awacs_master_sw_pmac5500 __initdata = +AWACS_SWITCH("Headphone Playback Switch", 1, SHIFT_HDMUTE, 1); + static struct snd_kcontrol_new snd_pmac_awacs_mic_boost[] __initdata = { AWACS_SWITCH("Mic Boost Capture Switch", 0, SHIFT_GAINLINE, 0), }; @@ -766,12 +780,16 @@ static void snd_pmac_awacs_resume(struct snd_pmac *chip) } #endif /* CONFIG_PM */ -#define IS_PM7500 (machine_is_compatible("AAPL,7500")) +#define IS_PM7500 (machine_is_compatible("AAPL,7500") \ + || machine_is_compatible("AAPL,8500") \ + || machine_is_compatible("AAPL,9500")) +#define IS_PM5500 (machine_is_compatible("AAPL,e411")) #define IS_BEIGE (machine_is_compatible("AAPL,Gossamer")) #define IS_IMAC1 (machine_is_compatible("PowerMac2,1")) #define IS_IMAC2 (machine_is_compatible("PowerMac2,2") \ || machine_is_compatible("PowerMac4,1")) #define IS_G4AGP (machine_is_compatible("PowerMac3,1")) +#define IS_LOMBARD (machine_is_compatible("PowerBook1,1")) static int imac1, imac2; @@ -858,10 +876,14 @@ int __init snd_pmac_awacs_init(struct snd_pmac *chip) { int pm7500 = IS_PM7500; + int pm5500 = IS_PM5500; int beige = IS_BEIGE; int g4agp = IS_G4AGP; + int lombard = IS_LOMBARD; int imac; int err, vol; + struct snd_kcontrol *vmaster_sw, *vmaster_vol; + struct snd_kcontrol *master_vol, *speaker_vol; imac1 = IS_IMAC1; imac2 = IS_IMAC2; @@ -915,7 +937,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) /* set headphone-jack detection bit */ switch (chip->model) { case PMAC_AWACS: - chip->hp_stat_mask = pm7500 ? MASK_HDPCONN + chip->hp_stat_mask = pm7500 || pm5500 ? MASK_HDPCONN : MASK_LOCONN; break; case PMAC_SCREAMER: @@ -954,7 +976,7 @@ snd_pmac_awacs_init(struct snd_pmac *chip) return err; if (beige || g4agp) ; - else if (chip->model == PMAC_SCREAMER) + else if (chip->model == PMAC_SCREAMER || pm5500) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers2), snd_pmac_screamer_mixers2); else if (!pm7500) @@ -962,19 +984,35 @@ snd_pmac_awacs_init(struct snd_pmac *chip) snd_pmac_awacs_mixers2); if (err < 0) return err; + if (pm5500) { + err = build_mixers(chip, + ARRAY_SIZE(snd_pmac_awacs_mixers2_pmac5500), + snd_pmac_awacs_mixers2_pmac5500); + if (err < 0) + return err; + } if (pm7500) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_mixers_pmac7500), snd_pmac_awacs_mixers_pmac7500); + else if (pm5500) + err = snd_ctl_add(chip->card, + (master_vol = snd_ctl_new1(snd_pmac_awacs_mixers_pmac5500, + chip))); else if (beige) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers_beige), snd_pmac_screamer_mixers_beige); - else if (imac) + else if (imac || lombard) { + err = snd_ctl_add(chip->card, + (master_vol = snd_ctl_new1(snd_pmac_screamer_mixers_lo, + chip))); + if (err < 0) + return err; err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers_imac), snd_pmac_screamer_mixers_imac); - else if (g4agp) + } else if (g4agp) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mixers_g4agp), snd_pmac_screamer_mixers_g4agp); @@ -984,8 +1022,10 @@ snd_pmac_awacs_init(struct snd_pmac *chip) snd_pmac_awacs_mixers_pmac); if (err < 0) return err; - chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac || g4agp) + chip->master_sw_ctl = snd_ctl_new1((pm7500 || imac || g4agp || lombard) ? &snd_pmac_awacs_master_sw_imac + : pm5500 + ? &snd_pmac_awacs_master_sw_pmac5500 : &snd_pmac_awacs_master_sw, chip); err = snd_ctl_add(chip->card, chip->master_sw_ctl); if (err < 0) @@ -1017,8 +1057,9 @@ snd_pmac_awacs_init(struct snd_pmac *chip) #endif /* PMAC_AMP_AVAIL */ { /* route A = headphone, route C = speaker */ - err = build_mixers(chip, ARRAY_SIZE(snd_pmac_awacs_speaker_vol), - snd_pmac_awacs_speaker_vol); + err = snd_ctl_add(chip->card, + (speaker_vol = snd_ctl_new1(snd_pmac_awacs_speaker_vol, + chip))); if (err < 0) return err; chip->speaker_sw_ctl = snd_ctl_new1(imac1 @@ -1031,6 +1072,33 @@ snd_pmac_awacs_init(struct snd_pmac *chip) return err; } + if (pm5500 || imac || lombard) { + vmaster_sw = snd_ctl_make_virtual_master( + "Master Playback Switch", (unsigned int *) NULL); + err = snd_ctl_add_slave_uncached(vmaster_sw, + chip->master_sw_ctl); + if (err < 0) + return err; + err = snd_ctl_add_slave_uncached(vmaster_sw, + chip->speaker_sw_ctl); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, vmaster_sw); + if (err < 0) + return err; + vmaster_vol = snd_ctl_make_virtual_master( + "Master Playback Volume", (unsigned int *) NULL); + err = snd_ctl_add_slave(vmaster_vol, master_vol); + if (err < 0) + return err; + err = snd_ctl_add_slave(vmaster_vol, speaker_vol); + if (err < 0) + return err; + err = snd_ctl_add(chip->card, vmaster_vol); + if (err < 0) + return err; + } + if (beige || g4agp) err = build_mixers(chip, ARRAY_SIZE(snd_pmac_screamer_mic_boost_beige), diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index f860d39..45a7629 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -35,7 +35,7 @@ snd_pmac_burgundy_busy_wait(struct snd_pmac *chip) int timeout = 50; while ((in_le32(&chip->awacs->codec_ctrl) & MASK_NEWECMD) && timeout--) udelay(1); - if (! timeout) + if (timeout < 0) printk(KERN_DEBUG "burgundy_busy_wait: timeout\n"); } diff --git a/sound/ppc/daca.c b/sound/ppc/daca.c index 8a5b290..f8d478c 100644 --- a/sound/ppc/daca.c +++ b/sound/ppc/daca.c @@ -82,7 +82,7 @@ static int daca_set_volume(struct pmac_daca *mix) data[1] |= mix->deemphasis ? 0x40 : 0; if (i2c_smbus_write_block_data(mix->i2c.client, DACA_REG_AVOL, 2, data) < 0) { - snd_printk("failed to set volume \n"); + snd_printk(KERN_ERR "failed to set volume \n"); return -EINVAL; } return 0; diff --git a/sound/ppc/pmac.c b/sound/ppc/pmac.c index af76ee8..9b4e9c3 100644 --- a/sound/ppc/pmac.c +++ b/sound/ppc/pmac.c @@ -299,7 +299,7 @@ static int snd_pmac_pcm_trigger(struct snd_pmac *chip, struct pmac_stream *rec, case SNDRV_PCM_TRIGGER_SUSPEND: spin_lock(&chip->reg_lock); rec->running = 0; - /*printk("stopped!!\n");*/ + /*printk(KERN_DEBUG "stopped!!\n");*/ snd_pmac_dma_stop(rec); for (i = 0, cp = rec->cmd.cmds; i < rec->nperiods; i++, cp++) out_le16(&cp->command, DBDMA_STOP); @@ -334,7 +334,7 @@ static snd_pcm_uframes_t snd_pmac_pcm_pointer(struct snd_pmac *chip, } #endif count += rec->cur_period * rec->period_size; - /*printk("pointer=%d\n", count);*/ + /*printk(KERN_DEBUG "pointer=%d\n", count);*/ return bytes_to_frames(subs->runtime, count); } @@ -486,7 +486,7 @@ static void snd_pmac_pcm_update(struct snd_pmac *chip, struct pmac_stream *rec) if (! (stat & ACTIVE)) break; - /*printk("update frag %d\n", rec->cur_period);*/ + /*printk(KERN_DEBUG "update frag %d\n", rec->cur_period);*/ st_le16(&cp->xfer_status, 0); st_le16(&cp->req_count, rec->period_size); /*st_le16(&cp->res_count, 0);*/ @@ -806,7 +806,7 @@ snd_pmac_ctrl_intr(int irq, void *devid) struct snd_pmac *chip = devid; int ctrl = in_le32(&chip->awacs->control); - /*printk("pmac: control interrupt.. 0x%x\n", ctrl);*/ + /*printk(KERN_DEBUG "pmac: control interrupt.. 0x%x\n", ctrl);*/ if (ctrl & MASK_PORTCHG) { /* do something when headphone is plugged/unplugged? */ if (chip->update_automute) @@ -1033,7 +1033,8 @@ static int __init snd_pmac_detect(struct snd_pmac *chip) } if (of_device_is_compatible(sound, "tumbler")) { chip->model = PMAC_TUMBLER; - chip->can_capture = machine_is_compatible("PowerMac4,2"); + chip->can_capture = machine_is_compatible("PowerMac4,2") + || machine_is_compatible("PowerBook4,1"); chip->can_duplex = 0; // chip->can_byte_swap = 0; /* FIXME: check this */ chip->num_freqs = ARRAY_SIZE(tumbler_freqs); diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 2e18ed0..a2b69b8 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -51,7 +51,7 @@ static struct platform_device *device; /* */ -static int __init snd_pmac_probe(struct platform_device *devptr) +static int __devinit snd_pmac_probe(struct platform_device *devptr) { struct snd_card *card; struct snd_pmac *chip; @@ -110,7 +110,7 @@ static int __init snd_pmac_probe(struct platform_device *devptr) goto __error; break; default: - snd_printk("unsupported hardware %d\n", chip->model); + snd_printk(KERN_ERR "unsupported hardware %d\n", chip->model); err = -EINVAL; goto __error; } diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index ef2c3f4..f361c26 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -477,7 +477,7 @@ static int snd_ps3_pcm_prepare(struct snd_pcm_substream *substream) card->dma_start_bus_addr[SND_PS3_CH_R] = runtime->dma_addr + (runtime->dma_bytes / 2); - pr_debug("%s: vaddr=%p bus=%#lx\n", __func__, + pr_debug("%s: vaddr=%p bus=%#llx\n", __func__, card->dma_start_vaddr[SND_PS3_CH_L], card->dma_start_bus_addr[SND_PS3_CH_L]); @@ -1028,7 +1028,7 @@ static int __init snd_ps3_driver_probe(struct ps3_system_bus_device *dev) pr_info("%s: nullbuffer alloc failed\n", __func__); goto clean_preallocate; } - pr_debug("%s: null vaddr=%p dma=%#lx\n", __func__, + pr_debug("%s: null vaddr=%p dma=%#llx\n", __func__, the_card.null_buffer_start_vaddr, the_card.null_buffer_start_dma_addr); /* set default sample rate/word width */ diff --git a/sound/ppc/tumbler.c b/sound/ppc/tumbler.c index 3eb2233..40222fc 100644 --- a/sound/ppc/tumbler.c +++ b/sound/ppc/tumbler.c @@ -41,7 +41,7 @@ #undef DEBUG #ifdef DEBUG -#define DBG(fmt...) printk(fmt) +#define DBG(fmt...) printk(KERN_DEBUG fmt) #else #define DBG(fmt...) #endif @@ -240,7 +240,7 @@ static int tumbler_set_master_volume(struct pmac_tumbler *mix) if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_VOL, 6, block) < 0) { - snd_printk("failed to set volume \n"); + snd_printk(KERN_ERR "failed to set volume \n"); return -EINVAL; } return 0; @@ -350,7 +350,7 @@ static int tumbler_set_drc(struct pmac_tumbler *mix) if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_DRC, 2, val) < 0) { - snd_printk("failed to set DRC\n"); + snd_printk(KERN_ERR "failed to set DRC\n"); return -EINVAL; } return 0; @@ -386,7 +386,7 @@ static int snapper_set_drc(struct pmac_tumbler *mix) if (i2c_smbus_write_i2c_block_data(mix->i2c.client, TAS_REG_DRC, 6, val) < 0) { - snd_printk("failed to set DRC\n"); + snd_printk(KERN_ERR "failed to set DRC\n"); return -EINVAL; } return 0; @@ -506,7 +506,8 @@ static int tumbler_set_mono_volume(struct pmac_tumbler *mix, block[i] = (vol >> ((info->bytes - i - 1) * 8)) & 0xff; if (i2c_smbus_write_i2c_block_data(mix->i2c.client, info->reg, info->bytes, block) < 0) { - snd_printk("failed to set mono volume %d\n", info->index); + snd_printk(KERN_ERR "failed to set mono volume %d\n", + info->index); return -EINVAL; } return 0; @@ -643,7 +644,7 @@ static int snapper_set_mix_vol1(struct pmac_tumbler *mix, int idx, int ch, int r } if (i2c_smbus_write_i2c_block_data(mix->i2c.client, reg, 9, block) < 0) { - snd_printk("failed to set mono volume %d\n", reg); + snd_printk(KERN_ERR "failed to set mono volume %d\n", reg); return -EINVAL; } return 0; diff --git a/sound/sh/Kconfig b/sound/sh/Kconfig index cfc1439..aed0f90 100644 --- a/sound/sh/Kconfig +++ b/sound/sh/Kconfig @@ -15,6 +15,7 @@ config SND_AICA tristate "Dreamcast Yamaha AICA sound" depends on SH_DREAMCAST select SND_PCM + select G2_DMA help ALSA Sound driver for the SEGA Dreamcast console. diff --git a/sound/sh/aica.c b/sound/sh/aica.c index f551233..583a369 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -565,7 +565,7 @@ static int load_aica_firmware(void) err = request_firmware(&fw_entry, "aica_firmware.bin", &pd->dev); if (unlikely(err)) return err; - /* write firware into memory */ + /* write firmware into memory */ spu_disable(); spu_memload(0, fw_entry->data, fw_entry->size); spu_enable(); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 3d2bb6f..3304f9d 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -32,6 +32,7 @@ source "sound/soc/fsl/Kconfig" source "sound/soc/omap/Kconfig" source "sound/soc/pxa/Kconfig" source "sound/soc/s3c24xx/Kconfig" +source "sound/soc/s6000/Kconfig" source "sound/soc/sh/Kconfig" # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 0237879..8943a14 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -10,4 +10,5 @@ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += omap/ obj-$(CONFIG_SND_SOC) += pxa/ obj-$(CONFIG_SND_SOC) += s3c24xx/ +obj-$(CONFIG_SND_SOC) += s6000/ obj-$(CONFIG_SND_SOC) += sh/ diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 0a2f8f9..811596f 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -42,7 +42,7 @@ config SND_BF5XX_AC97 You will also need to select the audio interfaces to support below. Note: - AC97 codecs which do not implment the slot-16 mode will not function + AC97 codecs which do not implement the slot-16 mode will not function properly with this driver. This driver is known to work with the Analog Devices line of AC97 codecs. diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 8cfed1a..cf0dfb7 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -413,7 +413,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) sport_done(sport_handle); } -static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -424,7 +424,7 @@ int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &bf5xx_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 1318c4f..62fbb84 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -245,7 +245,7 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) sport_done(sport_handle); } -static u64 bf5xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -256,7 +256,7 @@ int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &bf5xx_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = bf5xx_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b6c7f7a..121d63f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -35,7 +35,9 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C + select SND_SOC_WM8960 if I2C select SND_SOC_WM8971 if I2C + select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8990 if I2C select SND_SOC_WM9705 if SND_SOC_AC97_BUS select SND_SOC_WM9712 if SND_SOC_AC97_BUS @@ -138,9 +140,15 @@ config SND_SOC_WM8900 config SND_SOC_WM8903 tristate +config SND_SOC_WM8960 + tristate + config SND_SOC_WM8971 tristate +config SND_SOC_WM8988 + tristate + config SND_SOC_WM8990 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 030d245..d8e15a4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,7 +23,9 @@ snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o +snd-soc-wm8960-objs := wm8960.o snd-soc-wm8971-objs := wm8971.o +snd-soc-wm8988-objs := wm8988.o snd-soc-wm8990-objs := wm8990.o snd-soc-wm9705-objs := wm9705.o snd-soc-wm9712-objs := wm9712.o @@ -55,6 +57,8 @@ obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o +obj-$(CONFIG_SND_SOC_WM8960) += snd-soc-wm8960.o +obj-$(CONFIG_SND_SOC_WM8988) += snd-soc-wm8988.o obj-$(CONFIG_SND_SOC_WM8990) += snd-soc-wm8990.o obj-$(CONFIG_SND_SOC_WM8991) += snd-soc-wm8991.o obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index c3f4afb..21f69df 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -523,6 +523,8 @@ static int tlv320aic23_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_I2S: iface_reg |= TLV320AIC23_FOR_I2S; break; + case SND_SOC_DAIFMT_DSP_A: + iface_reg |= TLV320AIC23_LRP_ON; case SND_SOC_DAIFMT_DSP_B: iface_reg |= TLV320AIC23_FOR_DSP; break; diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 921b205..cc2968c 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -125,6 +125,11 @@ struct twl4030_priv { struct snd_pcm_substream *master_substream; struct snd_pcm_substream *slave_substream; + + unsigned int configured; + unsigned int rate; + unsigned int sample_bits; + unsigned int channels; }; /* @@ -1220,6 +1225,36 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec, return 0; } +static void twl4030_constraints(struct twl4030_priv *twl4030, + struct snd_pcm_substream *mst_substream) +{ + struct snd_pcm_substream *slv_substream; + + /* Pick the stream, which need to be constrained */ + if (mst_substream == twl4030->master_substream) + slv_substream = twl4030->slave_substream; + else if (mst_substream == twl4030->slave_substream) + slv_substream = twl4030->master_substream; + else /* This should not happen.. */ + return; + + /* Set the constraints according to the already configured stream */ + snd_pcm_hw_constraint_minmax(slv_substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + twl4030->rate, + twl4030->rate); + + snd_pcm_hw_constraint_minmax(slv_substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + twl4030->sample_bits, + twl4030->sample_bits); + + snd_pcm_hw_constraint_minmax(slv_substream->runtime, + SNDRV_PCM_HW_PARAM_CHANNELS, + twl4030->channels, + twl4030->channels); +} + static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1228,26 +1263,16 @@ static int twl4030_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = socdev->card->codec; struct twl4030_priv *twl4030 = codec->private_data; - /* If we already have a playback or capture going then constrain - * this substream to match it. - */ if (twl4030->master_substream) { - struct snd_pcm_runtime *master_runtime; - master_runtime = twl4030->master_substream->runtime; - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_RATE, - master_runtime->rate, - master_runtime->rate); - - snd_pcm_hw_constraint_minmax(substream->runtime, - SNDRV_PCM_HW_PARAM_SAMPLE_BITS, - master_runtime->sample_bits, - master_runtime->sample_bits); - twl4030->slave_substream = substream; - } else + /* The DAI has one configuration for playback and capture, so + * if the DAI has been already configured then constrain this + * substream to match it. */ + if (twl4030->configured) + twl4030_constraints(twl4030, twl4030->master_substream); + } else { twl4030->master_substream = substream; + } return 0; } @@ -1264,6 +1289,13 @@ static void twl4030_shutdown(struct snd_pcm_substream *substream, twl4030->master_substream = twl4030->slave_substream; twl4030->slave_substream = NULL; + + /* If all streams are closed, or the remaining stream has not yet + * been configured than set the DAI as not configured. */ + if (!twl4030->master_substream) + twl4030->configured = 0; + else if (!twl4030->master_substream->runtime->channels) + twl4030->configured = 0; } static int twl4030_hw_params(struct snd_pcm_substream *substream, @@ -1276,8 +1308,8 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, struct twl4030_priv *twl4030 = codec->private_data; u8 mode, old_mode, format, old_format; - if (substream == twl4030->slave_substream) - /* Ignoring hw_params for slave substream */ + if (twl4030->configured) + /* Ignoring hw_params for already configured DAI */ return 0; /* bit rate */ @@ -1357,6 +1389,21 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream, /* set CODECPDZ afterwards */ twl4030_codec_enable(codec, 1); } + + /* Store the important parameters for the DAI configuration and set + * the DAI as configured */ + twl4030->configured = 1; + twl4030->rate = params_rate(params); + twl4030->sample_bits = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min; + twl4030->channels = params_channels(params); + + /* If both playback and capture streams are open, and one of them + * is setting the hw parameters right now (since we are here), set + * constraints to the other stream to match the current one. */ + if (twl4030->slave_substream) + twl4030_constraints(twl4030, substream); + return 0; } @@ -1437,6 +1484,144 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +static int twl4030_voice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 infreq; + u8 mode; + + /* If the system master clock is not 26MHz, the voice PCM interface is + * not avilable. + */ + infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) + & TWL4030_APLL_INFREQ; + + if (infreq != TWL4030_APLL_INFREQ_26000KHZ) { + printk(KERN_ERR "TWL4030 voice startup: " + "MCLK is not 26MHz, call set_sysclk() on init\n"); + return -EINVAL; + } + + /* If the codec mode is not option2, the voice PCM interface is not + * avilable. + */ + mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) + & TWL4030_OPT_MODE; + + if (mode != TWL4030_OPTION_2) { + printk(KERN_ERR "TWL4030 voice startup: " + "the codec mode is not option2\n"); + return -EINVAL; + } + + return 0; +} + +static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 old_mode, mode; + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, TWL4030_REG_CODEC_MODE) + & ~(TWL4030_CODECPDZ); + mode = old_mode; + + switch (params_rate(params)) { + case 8000: + mode &= ~(TWL4030_SEL_16K); + break; + case 16000: + mode |= TWL4030_SEL_16K; + break; + default: + printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + +static int twl4030_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 infreq; + + switch (freq) { + case 26000000: + infreq = TWL4030_APLL_INFREQ_26000KHZ; + break; + default: + printk(KERN_ERR "TWL4030 voice set sysclk: unknown rate %d\n", + freq); + return -EINVAL; + } + + infreq |= TWL4030_APLL_EN; + twl4030_write(codec, TWL4030_REG_APLL_CTL, infreq); + + return 0; +} + +static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: + format &= ~(TWL4030_VIF_SLAVE_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + format |= TWL4030_VIF_SLAVE_EN; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_NF: + format &= ~(TWL4030_VIF_FORMAT); + break; + case SND_SOC_DAIFMT_NB_IF: + format |= TWL4030_VIF_FORMAT; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + /* change format and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE) @@ -1448,7 +1633,15 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .set_fmt = twl4030_set_dai_fmt, }; -struct snd_soc_dai twl4030_dai = { +static struct snd_soc_dai_ops twl4030_dai_voice_ops = { + .startup = twl4030_voice_startup, + .hw_params = twl4030_voice_hw_params, + .set_sysclk = twl4030_voice_set_dai_sysclk, + .set_fmt = twl4030_voice_set_dai_fmt, +}; + +struct snd_soc_dai twl4030_dai[] = { +{ .name = "twl4030", .playback = { .stream_name = "Playback", @@ -1463,6 +1656,23 @@ struct snd_soc_dai twl4030_dai = { .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .ops = &twl4030_dai_ops, +}, +{ + .name = "twl4030 Voice", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &twl4030_dai_voice_ops, +}, }; EXPORT_SYMBOL_GPL(twl4030_dai); @@ -1503,8 +1713,8 @@ static int twl4030_init(struct snd_soc_device *socdev) codec->read = twl4030_read_reg_cache; codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; - codec->dai = &twl4030_dai; - codec->num_dai = 1; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); @@ -1598,13 +1808,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030); static int __init twl4030_modinit(void) { - return snd_soc_register_dai(&twl4030_dai); + return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_init(twl4030_modinit); static void __exit twl4030_exit(void) { - snd_soc_unregister_dai(&twl4030_dai); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_exit(twl4030_exit); diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index cb63765..981ec60 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -113,6 +113,8 @@ #define TWL4030_SEL_16K 0x04 #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 +#define TWL4030_OPTION_1 (1 << 0) +#define TWL4030_OPTION_2 (0 << 0) /* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ @@ -171,6 +173,17 @@ #define TWL4030_CLK256FS_EN 0x02 #define TWL4030_AIF_EN 0x01 +/* VOICE_IF (0x0F) Fields */ + +#define TWL4030_VIF_SLAVE_EN 0x80 +#define TWL4030_VIF_DIN_EN 0x40 +#define TWL4030_VIF_DOUT_EN 0x20 +#define TWL4030_VIF_SWAP 0x10 +#define TWL4030_VIF_FORMAT 0x08 +#define TWL4030_VIF_TRI_EN 0x04 +#define TWL4030_VIF_SUB_EN 0x02 +#define TWL4030_VIF_EN 0x01 + /* EAR_CTL (0x21) */ #define TWL4030_EAR_GAIN 0x30 @@ -236,7 +249,10 @@ #define TWL4030_SMOOTH_ANAVOL_EN 0x02 #define TWL4030_DIGMIC_LR_SWAP_EN 0x01 -extern struct snd_soc_dai twl4030_dai; +#define TWL4030_DAI_HIFI 0 +#define TWL4030_DAI_VOICE 1 + +extern struct snd_soc_dai twl4030_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_twl4030; #endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 8cf571f..c539184 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1523,6 +1523,7 @@ struct snd_soc_dai wm8903_dai = { .formats = WM8903_FORMATS, }, .ops = &wm8903_dai_ops, + .symmetric_rates = 1, }; EXPORT_SYMBOL_GPL(wm8903_dai); diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c new file mode 100644 index 0000000..e224d8a --- /dev/null +++ b/sound/soc/codecs/wm8960.c @@ -0,0 +1,969 @@ +/* + * wm8960.c -- WM8960 ALSA SoC Audio driver + * + * Author: Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8960.h" + +#define AUDIO_NAME "wm8960" + +struct snd_soc_codec_device soc_codec_dev_wm8960; + +/* R25 - Power 1 */ +#define WM8960_VREF 0x40 + +/* R28 - Anti-pop 1 */ +#define WM8960_POBCTRL 0x80 +#define WM8960_BUFDCOPEN 0x10 +#define WM8960_BUFIOEN 0x08 +#define WM8960_SOFT_ST 0x04 +#define WM8960_HPSTBY 0x01 + +/* R29 - Anti-pop 2 */ +#define WM8960_DISOP 0x40 + +/* + * wm8960 register cache + * We can't read the WM8960 register space when we are + * using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8960_reg[WM8960_CACHEREGNUM] = { + 0x0097, 0x0097, 0x0000, 0x0000, + 0x0000, 0x0008, 0x0000, 0x000a, + 0x01c0, 0x0000, 0x00ff, 0x00ff, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x007b, 0x0100, 0x0032, + 0x0000, 0x00c3, 0x00c3, 0x01c0, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0000, 0x0000, + 0x0100, 0x0100, 0x0050, 0x0050, + 0x0050, 0x0050, 0x0000, 0x0000, + 0x0000, 0x0000, 0x0040, 0x0000, + 0x0000, 0x0050, 0x0050, 0x0000, + 0x0002, 0x0037, 0x004d, 0x0080, + 0x0008, 0x0031, 0x0026, 0x00e9, +}; + +struct wm8960_priv { + u16 reg_cache[WM8960_CACHEREGNUM]; + struct snd_soc_codec codec; +}; + +/* + * read wm8960 register cache + */ +static inline unsigned int wm8960_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg == WM8960_RESET) + return 0; + if (reg >= WM8960_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8960 register cache + */ +static inline void wm8960_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8960_CACHEREGNUM) + return; + cache[reg] = value; +} + +static inline unsigned int wm8960_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + return wm8960_read_reg_cache(codec, reg); +} + +/* + * write to the WM8960 register space + */ +static int wm8960_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8960 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8960_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8960_reset(c) wm8960_write(c, WM8960_RESET, 0) + +/* enumerated controls */ +static const char *wm8960_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const char *wm8960_polarity[] = {"No Inversion", "Left Inverted", + "Right Inverted", "Stereo Inversion"}; +static const char *wm8960_3d_upper_cutoff[] = {"High", "Low"}; +static const char *wm8960_3d_lower_cutoff[] = {"Low", "High"}; +static const char *wm8960_alcfunc[] = {"Off", "Right", "Left", "Stereo"}; +static const char *wm8960_alcmode[] = {"ALC", "Limiter"}; + +static const struct soc_enum wm8960_enum[] = { + SOC_ENUM_SINGLE(WM8960_DACCTL1, 1, 4, wm8960_deemph), + SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), + SOC_ENUM_SINGLE(WM8960_DACCTL2, 5, 4, wm8960_polarity), + SOC_ENUM_SINGLE(WM8960_3D, 6, 2, wm8960_3d_upper_cutoff), + SOC_ENUM_SINGLE(WM8960_3D, 5, 2, wm8960_3d_lower_cutoff), + SOC_ENUM_SINGLE(WM8960_ALC1, 7, 4, wm8960_alcfunc), + SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), +}; + +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); + +static const struct snd_kcontrol_new wm8960_snd_controls[] = { +SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, + 0, 63, 0, adc_tlv), +SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL, + 6, 1, 0), +SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL, + 7, 1, 0), + +SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC, + 0, 255, 0, dac_tlv), + +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8960_LOUT1, WM8960_ROUT1, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Headphone Playback ZC Switch", WM8960_LOUT1, WM8960_ROUT1, + 7, 1, 0), + +SOC_DOUBLE_R_TLV("Speaker Playback Volume", WM8960_LOUT2, WM8960_ROUT2, + 0, 127, 0, out_tlv), +SOC_DOUBLE_R("Speaker Playback ZC Switch", WM8960_LOUT2, WM8960_ROUT2, + 7, 1, 0), +SOC_SINGLE("Speaker DC Volume", WM8960_CLASSD3, 3, 5, 0), +SOC_SINGLE("Speaker AC Volume", WM8960_CLASSD3, 0, 5, 0), + +SOC_SINGLE("PCM Playback -6dB Switch", WM8960_DACCTL1, 7, 1, 0), +SOC_ENUM("ADC Polarity", wm8960_enum[1]), +SOC_ENUM("Playback De-emphasis", wm8960_enum[0]), +SOC_SINGLE("ADC High Pass Filter Switch", WM8960_DACCTL1, 0, 1, 0), + +SOC_ENUM("DAC Polarity", wm8960_enum[2]), + +SOC_ENUM("3D Filter Upper Cut-Off", wm8960_enum[3]), +SOC_ENUM("3D Filter Lower Cut-Off", wm8960_enum[4]), +SOC_SINGLE("3D Volume", WM8960_3D, 1, 15, 0), +SOC_SINGLE("3D Switch", WM8960_3D, 0, 1, 0), + +SOC_ENUM("ALC Function", wm8960_enum[5]), +SOC_SINGLE("ALC Max Gain", WM8960_ALC1, 4, 7, 0), +SOC_SINGLE("ALC Target", WM8960_ALC1, 0, 15, 1), +SOC_SINGLE("ALC Min Gain", WM8960_ALC2, 4, 7, 0), +SOC_SINGLE("ALC Hold Time", WM8960_ALC2, 0, 15, 0), +SOC_ENUM("ALC Mode", wm8960_enum[6]), +SOC_SINGLE("ALC Decay", WM8960_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Attack", WM8960_ALC3, 0, 15, 0), + +SOC_SINGLE("Noise Gate Threshold", WM8960_NOISEG, 3, 31, 0), +SOC_SINGLE("Noise Gate Switch", WM8960_NOISEG, 0, 1, 0), + +SOC_DOUBLE_R("ADC PCM Capture Volume", WM8960_LINPATH, WM8960_RINPATH, + 0, 127, 0), + +SOC_SINGLE_TLV("Left Output Mixer Boost Bypass Volume", + WM8960_BYPASS1, 4, 7, 1, bypass_tlv), +SOC_SINGLE_TLV("Left Output Mixer LINPUT3 Volume", + WM8960_LOUTMIX, 4, 7, 1, bypass_tlv), +SOC_SINGLE_TLV("Right Output Mixer Boost Bypass Volume", + WM8960_BYPASS2, 4, 7, 1, bypass_tlv), +SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", + WM8960_ROUTMIX, 4, 7, 1, bypass_tlv), +}; + +static const struct snd_kcontrol_new wm8960_lin_boost[] = { +SOC_DAPM_SINGLE("LINPUT2 Switch", WM8960_LINPATH, 6, 1, 0), +SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LINPATH, 7, 1, 0), +SOC_DAPM_SINGLE("LINPUT1 Switch", WM8960_LINPATH, 8, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_lin[] = { +SOC_DAPM_SINGLE("Boost Switch", WM8960_LINPATH, 3, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_rin_boost[] = { +SOC_DAPM_SINGLE("RINPUT2 Switch", WM8960_RINPATH, 6, 1, 0), +SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_RINPATH, 7, 1, 0), +SOC_DAPM_SINGLE("RINPUT1 Switch", WM8960_RINPATH, 8, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_rin[] = { +SOC_DAPM_SINGLE("Boost Switch", WM8960_RINPATH, 3, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_loutput_mixer[] = { +SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_LOUTMIX, 8, 1, 0), +SOC_DAPM_SINGLE("LINPUT3 Switch", WM8960_LOUTMIX, 7, 1, 0), +SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS1, 7, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_routput_mixer[] = { +SOC_DAPM_SINGLE("PCM Playback Switch", WM8960_ROUTMIX, 8, 1, 0), +SOC_DAPM_SINGLE("RINPUT3 Switch", WM8960_ROUTMIX, 7, 1, 0), +SOC_DAPM_SINGLE("Boost Bypass Switch", WM8960_BYPASS2, 7, 1, 0), +}; + +static const struct snd_kcontrol_new wm8960_mono_out[] = { +SOC_DAPM_SINGLE("Left Switch", WM8960_MONOMIX1, 7, 1, 0), +SOC_DAPM_SINGLE("Right Switch", WM8960_MONOMIX2, 7, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8960_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("LINPUT1"), +SND_SOC_DAPM_INPUT("RINPUT1"), +SND_SOC_DAPM_INPUT("LINPUT2"), +SND_SOC_DAPM_INPUT("RINPUT2"), +SND_SOC_DAPM_INPUT("LINPUT3"), +SND_SOC_DAPM_INPUT("RINPUT3"), + +SND_SOC_DAPM_MICBIAS("MICB", WM8960_POWER1, 1, 0), + +SND_SOC_DAPM_MIXER("Left Boost Mixer", WM8960_POWER1, 5, 0, + wm8960_lin_boost, ARRAY_SIZE(wm8960_lin_boost)), +SND_SOC_DAPM_MIXER("Right Boost Mixer", WM8960_POWER1, 4, 0, + wm8960_rin_boost, ARRAY_SIZE(wm8960_rin_boost)), + +SND_SOC_DAPM_MIXER("Left Input Mixer", WM8960_POWER3, 5, 0, + wm8960_lin, ARRAY_SIZE(wm8960_lin)), +SND_SOC_DAPM_MIXER("Right Input Mixer", WM8960_POWER3, 4, 0, + wm8960_rin, ARRAY_SIZE(wm8960_rin)), + +SND_SOC_DAPM_ADC("Left ADC", "Capture", WM8960_POWER2, 3, 0), +SND_SOC_DAPM_ADC("Right ADC", "Capture", WM8960_POWER2, 2, 0), + +SND_SOC_DAPM_DAC("Left DAC", "Playback", WM8960_POWER2, 8, 0), +SND_SOC_DAPM_DAC("Right DAC", "Playback", WM8960_POWER2, 7, 0), + +SND_SOC_DAPM_MIXER("Left Output Mixer", WM8960_POWER3, 3, 0, + &wm8960_loutput_mixer[0], + ARRAY_SIZE(wm8960_loutput_mixer)), +SND_SOC_DAPM_MIXER("Right Output Mixer", WM8960_POWER3, 2, 0, + &wm8960_routput_mixer[0], + ARRAY_SIZE(wm8960_routput_mixer)), + +SND_SOC_DAPM_MIXER("Mono Output Mixer", WM8960_POWER2, 1, 0, + &wm8960_mono_out[0], + ARRAY_SIZE(wm8960_mono_out)), + +SND_SOC_DAPM_PGA("LOUT1 PGA", WM8960_POWER2, 6, 0, NULL, 0), +SND_SOC_DAPM_PGA("ROUT1 PGA", WM8960_POWER2, 5, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Left Speaker PGA", WM8960_POWER2, 4, 0, NULL, 0), +SND_SOC_DAPM_PGA("Right Speaker PGA", WM8960_POWER2, 3, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Right Speaker Output", WM8960_CLASSD1, 7, 0, NULL, 0), +SND_SOC_DAPM_PGA("Left Speaker Output", WM8960_CLASSD1, 6, 0, NULL, 0), + +SND_SOC_DAPM_OUTPUT("SPK_LP"), +SND_SOC_DAPM_OUTPUT("SPK_LN"), +SND_SOC_DAPM_OUTPUT("HP_L"), +SND_SOC_DAPM_OUTPUT("HP_R"), +SND_SOC_DAPM_OUTPUT("SPK_RP"), +SND_SOC_DAPM_OUTPUT("SPK_RN"), +SND_SOC_DAPM_OUTPUT("OUT3"), +}; + +static const struct snd_soc_dapm_route audio_paths[] = { + { "Left Boost Mixer", "LINPUT1 Switch", "LINPUT1" }, + { "Left Boost Mixer", "LINPUT2 Switch", "LINPUT2" }, + { "Left Boost Mixer", "LINPUT3 Switch", "LINPUT3" }, + + { "Left Input Mixer", "Boost Switch", "Left Boost Mixer", }, + { "Left Input Mixer", NULL, "LINPUT1", }, /* Really Boost Switch */ + { "Left Input Mixer", NULL, "LINPUT2" }, + { "Left Input Mixer", NULL, "LINPUT3" }, + + { "Right Boost Mixer", "RINPUT1 Switch", "RINPUT1" }, + { "Right Boost Mixer", "RINPUT2 Switch", "RINPUT2" }, + { "Right Boost Mixer", "RINPUT3 Switch", "RINPUT3" }, + + { "Right Input Mixer", "Boost Switch", "Right Boost Mixer", }, + { "Right Input Mixer", NULL, "RINPUT1", }, /* Really Boost Switch */ + { "Right Input Mixer", NULL, "RINPUT2" }, + { "Right Input Mixer", NULL, "LINPUT3" }, + + { "Left ADC", NULL, "Left Input Mixer" }, + { "Right ADC", NULL, "Right Input Mixer" }, + + { "Left Output Mixer", "LINPUT3 Switch", "LINPUT3" }, + { "Left Output Mixer", "Boost Bypass Switch", "Left Boost Mixer"} , + { "Left Output Mixer", "PCM Playback Switch", "Left DAC" }, + + { "Right Output Mixer", "RINPUT3 Switch", "RINPUT3" }, + { "Right Output Mixer", "Boost Bypass Switch", "Right Boost Mixer" } , + { "Right Output Mixer", "PCM Playback Switch", "Right DAC" }, + + { "Mono Output Mixer", "Left Switch", "Left Output Mixer" }, + { "Mono Output Mixer", "Right Switch", "Right Output Mixer" }, + + { "LOUT1 PGA", NULL, "Left Output Mixer" }, + { "ROUT1 PGA", NULL, "Right Output Mixer" }, + + { "HP_L", NULL, "LOUT1 PGA" }, + { "HP_R", NULL, "ROUT1 PGA" }, + + { "Left Speaker PGA", NULL, "Left Output Mixer" }, + { "Right Speaker PGA", NULL, "Right Output Mixer" }, + + { "Left Speaker Output", NULL, "Left Speaker PGA" }, + { "Right Speaker Output", NULL, "Right Speaker PGA" }, + + { "SPK_LN", NULL, "Left Speaker Output" }, + { "SPK_LP", NULL, "Left Speaker Output" }, + { "SPK_RN", NULL, "Right Speaker Output" }, + { "SPK_RP", NULL, "Right Speaker Output" }, + + { "OUT3", NULL, "Mono Output Mixer", } +}; + +static int wm8960_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, wm8960_dapm_widgets, + ARRAY_SIZE(wm8960_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int wm8960_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface |= 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + /* set iface */ + wm8960_write(codec, WM8960_IFACE1, iface); + return 0; +} + +static int wm8960_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u16 iface = wm8960_read(codec, WM8960_IFACE1) & 0xfff3; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + } + + /* set iface */ + wm8960_write(codec, WM8960_IFACE1, iface); + return 0; +} + +static int wm8960_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8960_read(codec, WM8960_DACCTL1) & 0xfff7; + + if (mute) + wm8960_write(codec, WM8960_DACCTL1, mute_reg | 0x8); + else + wm8960_write(codec, WM8960_DACCTL1, mute_reg); + return 0; +} + +static int wm8960_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm8960_data *pdata = codec->dev->platform_data; + u16 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Set VMID to 2x50k */ + reg = wm8960_read(codec, WM8960_POWER1); + reg &= ~0x180; + reg |= 0x80; + wm8960_write(codec, WM8960_POWER1, reg); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Enable anti-pop features */ + wm8960_write(codec, WM8960_APOP1, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN | WM8960_BUFIOEN); + + /* Discharge HP output */ + reg = WM8960_DISOP; + if (pdata) + reg |= pdata->dres << 4; + wm8960_write(codec, WM8960_APOP2, reg); + + msleep(400); + + wm8960_write(codec, WM8960_APOP2, 0); + + /* Enable & ramp VMID at 2x50k */ + reg = wm8960_read(codec, WM8960_POWER1); + reg |= 0x80; + wm8960_write(codec, WM8960_POWER1, reg); + msleep(100); + + /* Enable VREF */ + wm8960_write(codec, WM8960_POWER1, reg | WM8960_VREF); + + /* Disable anti-pop features */ + wm8960_write(codec, WM8960_APOP1, WM8960_BUFIOEN); + } + + /* Set VMID to 2x250k */ + reg = wm8960_read(codec, WM8960_POWER1); + reg &= ~0x180; + reg |= 0x100; + wm8960_write(codec, WM8960_POWER1, reg); + break; + + case SND_SOC_BIAS_OFF: + /* Enable anti-pop features */ + wm8960_write(codec, WM8960_APOP1, + WM8960_POBCTRL | WM8960_SOFT_ST | + WM8960_BUFDCOPEN | WM8960_BUFIOEN); + + /* Disable VMID and VREF, let them discharge */ + wm8960_write(codec, WM8960_POWER1, 0); + msleep(600); + + wm8960_write(codec, WM8960_APOP1, 0); + break; + } + + codec->bias_level = level; + + return 0; +} + +/* PLL divisors */ +struct _pll_div { + u32 pre_div:1; + u32 n:4; + u32 k:24; +}; + +/* The size in bits of the pll divide multiplied by 10 + * to allow rounding later */ +#define FIXED_PLL_SIZE ((1 << 24) * 10) + +static int pll_factors(unsigned int source, unsigned int target, + struct _pll_div *pll_div) +{ + unsigned long long Kpart; + unsigned int K, Ndiv, Nmod; + + pr_debug("WM8960 PLL: setting %dHz->%dHz\n", source, target); + + /* Scale up target to PLL operating frequency */ + target *= 4; + + Ndiv = target / source; + if (Ndiv < 6) { + source >>= 1; + pll_div->pre_div = 1; + Ndiv = target / source; + } else + pll_div->pre_div = 0; + + if ((Ndiv < 6) || (Ndiv > 12)) { + pr_err("WM8960 PLL: Unsupported N=%d\n", Ndiv); + return -EINVAL; + } + + pll_div->n = Ndiv; + Nmod = target % source; + Kpart = FIXED_PLL_SIZE * (long long)Nmod; + + do_div(Kpart, source); + + K = Kpart & 0xFFFFFFFF; + + /* Check if we need to round */ + if ((K % 10) >= 5) + K += 5; + + /* Move down to proper range now rounding is done */ + K /= 10; + + pll_div->k = K; + + pr_debug("WM8960 PLL: N=%x K=%x pre_div=%d\n", + pll_div->n, pll_div->k, pll_div->pre_div); + + return 0; +} + +static int wm8960_set_dai_pll(struct snd_soc_dai *codec_dai, + int pll_id, unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + static struct _pll_div pll_div; + int ret; + + if (freq_in && freq_out) { + ret = pll_factors(freq_in, freq_out, &pll_div); + if (ret != 0) + return ret; + } + + /* Disable the PLL: even if we are changing the frequency the + * PLL needs to be disabled while we do so. */ + wm8960_write(codec, WM8960_CLOCK1, + wm8960_read(codec, WM8960_CLOCK1) & ~1); + wm8960_write(codec, WM8960_POWER2, + wm8960_read(codec, WM8960_POWER2) & ~1); + + if (!freq_in || !freq_out) + return 0; + + reg = wm8960_read(codec, WM8960_PLL1) & ~0x3f; + reg |= pll_div.pre_div << 4; + reg |= pll_div.n; + + if (pll_div.k) { + reg |= 0x20; + + wm8960_write(codec, WM8960_PLL2, (pll_div.k >> 18) & 0x3f); + wm8960_write(codec, WM8960_PLL3, (pll_div.k >> 9) & 0x1ff); + wm8960_write(codec, WM8960_PLL4, pll_div.k & 0x1ff); + } + wm8960_write(codec, WM8960_PLL1, reg); + + /* Turn it on */ + wm8960_write(codec, WM8960_POWER2, + wm8960_read(codec, WM8960_POWER2) | 1); + msleep(250); + wm8960_write(codec, WM8960_CLOCK1, + wm8960_read(codec, WM8960_CLOCK1) | 1); + + return 0; +} + +static int wm8960_set_dai_clkdiv(struct snd_soc_dai *codec_dai, + int div_id, int div) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 reg; + + switch (div_id) { + case WM8960_SYSCLKSEL: + reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1fe; + wm8960_write(codec, WM8960_CLOCK1, reg | div); + break; + case WM8960_SYSCLKDIV: + reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1f9; + wm8960_write(codec, WM8960_CLOCK1, reg | div); + break; + case WM8960_DACDIV: + reg = wm8960_read(codec, WM8960_CLOCK1) & 0x1c7; + wm8960_write(codec, WM8960_CLOCK1, reg | div); + break; + case WM8960_OPCLKDIV: + reg = wm8960_read(codec, WM8960_PLL1) & 0x03f; + wm8960_write(codec, WM8960_PLL1, reg | div); + break; + case WM8960_DCLKDIV: + reg = wm8960_read(codec, WM8960_CLOCK2) & 0x03f; + wm8960_write(codec, WM8960_CLOCK2, reg | div); + break; + case WM8960_TOCLKSEL: + reg = wm8960_read(codec, WM8960_ADDCTL1) & 0x1fd; + wm8960_write(codec, WM8960_ADDCTL1, reg | div); + break; + default: + return -EINVAL; + } + + return 0; +} + +#define WM8960_RATES SNDRV_PCM_RATE_8000_48000 + +#define WM8960_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8960_dai_ops = { + .hw_params = wm8960_hw_params, + .digital_mute = wm8960_mute, + .set_fmt = wm8960_set_dai_fmt, + .set_clkdiv = wm8960_set_dai_clkdiv, + .set_pll = wm8960_set_dai_pll, +}; + +struct snd_soc_dai wm8960_dai = { + .name = "WM8960", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8960_RATES, + .formats = WM8960_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8960_RATES, + .formats = WM8960_FORMATS,}, + .ops = &wm8960_dai_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8960_dai); + +static int wm8960_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8960_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8960_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8960_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + wm8960_set_bias_level(codec, codec->suspend_bias_level); + return 0; +} + +static struct snd_soc_codec *wm8960_codec; + +static int wm8960_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8960_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8960_codec; + codec = wm8960_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8960_snd_controls, + ARRAY_SIZE(wm8960_snd_controls)); + wm8960_add_widgets(codec); + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int wm8960_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8960 = { + .probe = wm8960_probe, + .remove = wm8960_remove, + .suspend = wm8960_suspend, + .resume = wm8960_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8960); + +static int wm8960_register(struct wm8960_priv *wm8960) +{ + struct wm8960_data *pdata = wm8960->codec.dev->platform_data; + struct snd_soc_codec *codec = &wm8960->codec; + int ret; + u16 reg; + + if (wm8960_codec) { + dev_err(codec->dev, "Another WM8960 is registered\n"); + return -EINVAL; + } + + if (!pdata) { + dev_warn(codec->dev, "No platform data supplied\n"); + } else { + if (pdata->dres > WM8960_DRES_MAX) { + dev_err(codec->dev, "Invalid DRES: %d\n", pdata->dres); + pdata->dres = 0; + } + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8960; + codec->name = "WM8960"; + codec->owner = THIS_MODULE; + codec->read = wm8960_read_reg_cache; + codec->write = wm8960_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8960_set_bias_level; + codec->dai = &wm8960_dai; + codec->num_dai = 1; + codec->reg_cache_size = WM8960_CACHEREGNUM; + codec->reg_cache = &wm8960->reg_cache; + + memcpy(codec->reg_cache, wm8960_reg, sizeof(wm8960_reg)); + + ret = wm8960_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + wm8960_dai.dev = codec->dev; + + wm8960_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits */ + reg = wm8960_read(codec, WM8960_LINVOL); + wm8960_write(codec, WM8960_LINVOL, reg | 0x100); + reg = wm8960_read(codec, WM8960_RINVOL); + wm8960_write(codec, WM8960_RINVOL, reg | 0x100); + reg = wm8960_read(codec, WM8960_LADC); + wm8960_write(codec, WM8960_LADC, reg | 0x100); + reg = wm8960_read(codec, WM8960_RADC); + wm8960_write(codec, WM8960_RADC, reg | 0x100); + reg = wm8960_read(codec, WM8960_LDAC); + wm8960_write(codec, WM8960_LDAC, reg | 0x100); + reg = wm8960_read(codec, WM8960_RDAC); + wm8960_write(codec, WM8960_RDAC, reg | 0x100); + reg = wm8960_read(codec, WM8960_LOUT1); + wm8960_write(codec, WM8960_LOUT1, reg | 0x100); + reg = wm8960_read(codec, WM8960_ROUT1); + wm8960_write(codec, WM8960_ROUT1, reg | 0x100); + reg = wm8960_read(codec, WM8960_LOUT2); + wm8960_write(codec, WM8960_LOUT2, reg | 0x100); + reg = wm8960_read(codec, WM8960_ROUT2); + wm8960_write(codec, WM8960_ROUT2, reg | 0x100); + + wm8960_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8960_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; +} + +static void wm8960_unregister(struct wm8960_priv *wm8960) +{ + wm8960_set_bias_level(&wm8960->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8960_dai); + snd_soc_unregister_codec(&wm8960->codec); + kfree(wm8960); + wm8960_codec = NULL; +} + +static __devinit int wm8960_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8960_priv *wm8960; + struct snd_soc_codec *codec; + + wm8960 = kzalloc(sizeof(struct wm8960_priv), GFP_KERNEL); + if (wm8960 == NULL) + return -ENOMEM; + + codec = &wm8960->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8960); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8960_register(wm8960); +} + +static __devexit int wm8960_i2c_remove(struct i2c_client *client) +{ + struct wm8960_priv *wm8960 = i2c_get_clientdata(client); + wm8960_unregister(wm8960); + return 0; +} + +static const struct i2c_device_id wm8960_i2c_id[] = { + { "wm8960", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8960_i2c_id); + +static struct i2c_driver wm8960_i2c_driver = { + .driver = { + .name = "WM8960 I2C Codec", + .owner = THIS_MODULE, + }, + .probe = wm8960_i2c_probe, + .remove = __devexit_p(wm8960_i2c_remove), + .id_table = wm8960_i2c_id, +}; + +static int __init wm8960_modinit(void) +{ + int ret; + + ret = i2c_add_driver(&wm8960_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8960 I2C driver: %d\n", + ret); + } + + return ret; +} +module_init(wm8960_modinit); + +static void __exit wm8960_exit(void) +{ + i2c_del_driver(&wm8960_i2c_driver); +} +module_exit(wm8960_exit); + + +MODULE_DESCRIPTION("ASoC WM8960 driver"); +MODULE_AUTHOR("Liam Girdwood"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8960.h b/sound/soc/codecs/wm8960.h new file mode 100644 index 0000000..c9af56c --- /dev/null +++ b/sound/soc/codecs/wm8960.h @@ -0,0 +1,127 @@ +/* + * wm8960.h -- WM8960 Soc Audio driver + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8960_H +#define _WM8960_H + +/* WM8960 register space */ + + +#define WM8960_CACHEREGNUM 56 + +#define WM8960_LINVOL 0x0 +#define WM8960_RINVOL 0x1 +#define WM8960_LOUT1 0x2 +#define WM8960_ROUT1 0x3 +#define WM8960_CLOCK1 0x4 +#define WM8960_DACCTL1 0x5 +#define WM8960_DACCTL2 0x6 +#define WM8960_IFACE1 0x7 +#define WM8960_CLOCK2 0x8 +#define WM8960_IFACE2 0x9 +#define WM8960_LDAC 0xa +#define WM8960_RDAC 0xb + +#define WM8960_RESET 0xf +#define WM8960_3D 0x10 +#define WM8960_ALC1 0x11 +#define WM8960_ALC2 0x12 +#define WM8960_ALC3 0x13 +#define WM8960_NOISEG 0x14 +#define WM8960_LADC 0x15 +#define WM8960_RADC 0x16 +#define WM8960_ADDCTL1 0x17 +#define WM8960_ADDCTL2 0x18 +#define WM8960_POWER1 0x19 +#define WM8960_POWER2 0x1a +#define WM8960_ADDCTL3 0x1b +#define WM8960_APOP1 0x1c +#define WM8960_APOP2 0x1d + +#define WM8960_LINPATH 0x20 +#define WM8960_RINPATH 0x21 +#define WM8960_LOUTMIX 0x22 + +#define WM8960_ROUTMIX 0x25 +#define WM8960_MONOMIX1 0x26 +#define WM8960_MONOMIX2 0x27 +#define WM8960_LOUT2 0x28 +#define WM8960_ROUT2 0x29 +#define WM8960_MONO 0x2a +#define WM8960_INBMIX1 0x2b +#define WM8960_INBMIX2 0x2c +#define WM8960_BYPASS1 0x2d +#define WM8960_BYPASS2 0x2e +#define WM8960_POWER3 0x2f +#define WM8960_ADDCTL4 0x30 +#define WM8960_CLASSD1 0x31 + +#define WM8960_CLASSD3 0x33 +#define WM8960_PLL1 0x34 +#define WM8960_PLL2 0x35 +#define WM8960_PLL3 0x36 +#define WM8960_PLL4 0x37 + + +/* + * WM8960 Clock dividers + */ +#define WM8960_SYSCLKDIV 0 +#define WM8960_DACDIV 1 +#define WM8960_OPCLKDIV 2 +#define WM8960_DCLKDIV 3 +#define WM8960_TOCLKSEL 4 +#define WM8960_SYSCLKSEL 5 + +#define WM8960_SYSCLK_DIV_1 (0 << 1) +#define WM8960_SYSCLK_DIV_2 (2 << 1) + +#define WM8960_SYSCLK_MCLK (0 << 0) +#define WM8960_SYSCLK_PLL (1 << 0) + +#define WM8960_DAC_DIV_1 (0 << 3) +#define WM8960_DAC_DIV_1_5 (1 << 3) +#define WM8960_DAC_DIV_2 (2 << 3) +#define WM8960_DAC_DIV_3 (3 << 3) +#define WM8960_DAC_DIV_4 (4 << 3) +#define WM8960_DAC_DIV_5_5 (5 << 3) +#define WM8960_DAC_DIV_6 (6 << 3) + +#define WM8960_DCLK_DIV_1_5 (0 << 6) +#define WM8960_DCLK_DIV_2 (1 << 6) +#define WM8960_DCLK_DIV_3 (2 << 6) +#define WM8960_DCLK_DIV_4 (3 << 6) +#define WM8960_DCLK_DIV_6 (4 << 6) +#define WM8960_DCLK_DIV_8 (5 << 6) +#define WM8960_DCLK_DIV_12 (6 << 6) +#define WM8960_DCLK_DIV_16 (7 << 6) + +#define WM8960_TOCLK_F19 (0 << 1) +#define WM8960_TOCLK_F21 (1 << 1) + +#define WM8960_OPCLK_DIV_1 (0 << 0) +#define WM8960_OPCLK_DIV_2 (1 << 0) +#define WM8960_OPCLK_DIV_3 (2 << 0) +#define WM8960_OPCLK_DIV_4 (3 << 0) +#define WM8960_OPCLK_DIV_5_5 (4 << 0) +#define WM8960_OPCLK_DIV_6 (5 << 0) + +extern struct snd_soc_dai wm8960_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8960; + +#define WM8960_DRES_400R 0 +#define WM8960_DRES_200R 1 +#define WM8960_DRES_600R 2 +#define WM8960_DRES_150R 3 +#define WM8960_DRES_MAX 3 + +struct wm8960_data { + int dres; +}; + +#endif diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c new file mode 100644 index 0000000..c05f718 --- /dev/null +++ b/sound/soc/codecs/wm8988.c @@ -0,0 +1,1097 @@ +/* + * wm8988.c -- WM8988 ALSA SoC audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2005 Openedhand Ltd. + * + * Author: Mark Brown <broonie@opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> + +#include "wm8988.h" + +/* + * wm8988 register cache + * We can't read the WM8988 register space when we + * are using 2 wire for device control, so we cache them instead. + */ +static const u16 wm8988_reg[] = { + 0x0097, 0x0097, 0x0079, 0x0079, /* 0 */ + 0x0000, 0x0008, 0x0000, 0x000a, /* 4 */ + 0x0000, 0x0000, 0x00ff, 0x00ff, /* 8 */ + 0x000f, 0x000f, 0x0000, 0x0000, /* 12 */ + 0x0000, 0x007b, 0x0000, 0x0032, /* 16 */ + 0x0000, 0x00c3, 0x00c3, 0x00c0, /* 20 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 24 */ + 0x0000, 0x0000, 0x0000, 0x0000, /* 28 */ + 0x0000, 0x0000, 0x0050, 0x0050, /* 32 */ + 0x0050, 0x0050, 0x0050, 0x0050, /* 36 */ + 0x0079, 0x0079, 0x0079, /* 40 */ +}; + +/* codec private data */ +struct wm8988_priv { + unsigned int sysclk; + struct snd_soc_codec codec; + struct snd_pcm_hw_constraint_list *sysclk_constraints; + u16 reg_cache[WM8988_NUM_REG]; +}; + + +/* + * read wm8988 register cache + */ +static inline unsigned int wm8988_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg > WM8988_NUM_REG) + return -1; + return cache[reg]; +} + +/* + * write wm8988 register cache + */ +static inline void wm8988_write_reg_cache(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg > WM8988_NUM_REG) + return; + cache[reg] = value; +} + +static int wm8988_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + /* data is + * D15..D9 WM8753 register offset + * D8...D0 register data + */ + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8988_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +#define wm8988_reset(c) wm8988_write(c, WM8988_RESET, 0) + +/* + * WM8988 Controls + */ + +static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"}; +static const struct soc_enum bass_boost = + SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt); + +static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" }; +static const struct soc_enum bass_filter = + SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt); + +static const char *treble_txt[] = {"8kHz", "4kHz"}; +static const struct soc_enum treble = + SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt); + +static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"}; +static const struct soc_enum stereo_3d_lc = + SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt); + +static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"}; +static const struct soc_enum stereo_3d_uc = + SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt); + +static const char *stereo_3d_func_txt[] = {"Capture", "Playback"}; +static const struct soc_enum stereo_3d_func = + SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt); + +static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"}; +static const struct soc_enum alc_func = + SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt); + +static const char *ng_type_txt[] = {"Constant PGA Gain", + "Mute ADC Output"}; +static const struct soc_enum ng_type = + SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt); + +static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"}; +static const struct soc_enum deemph = + SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt); + +static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert", + "L + R Invert"}; +static const struct soc_enum adcpol = + SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt); + +static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); + +static const struct snd_kcontrol_new wm8988_snd_controls[] = { + +SOC_ENUM("Bass Boost", bass_boost), +SOC_ENUM("Bass Filter", bass_filter), +SOC_SINGLE("Bass Volume", WM8988_BASS, 0, 15, 1), + +SOC_SINGLE("Treble Volume", WM8988_TREBLE, 0, 15, 0), +SOC_ENUM("Treble Cut-off", treble), + +SOC_SINGLE("3D Switch", WM8988_3D, 0, 1, 0), +SOC_SINGLE("3D Volume", WM8988_3D, 1, 15, 0), +SOC_ENUM("3D Lower Cut-off", stereo_3d_lc), +SOC_ENUM("3D Upper Cut-off", stereo_3d_uc), +SOC_ENUM("3D Mode", stereo_3d_func), + +SOC_SINGLE("ALC Capture Target Volume", WM8988_ALC1, 0, 7, 0), +SOC_SINGLE("ALC Capture Max Volume", WM8988_ALC1, 4, 7, 0), +SOC_ENUM("ALC Capture Function", alc_func), +SOC_SINGLE("ALC Capture ZC Switch", WM8988_ALC2, 7, 1, 0), +SOC_SINGLE("ALC Capture Hold Time", WM8988_ALC2, 0, 15, 0), +SOC_SINGLE("ALC Capture Decay Time", WM8988_ALC3, 4, 15, 0), +SOC_SINGLE("ALC Capture Attack Time", WM8988_ALC3, 0, 15, 0), +SOC_SINGLE("ALC Capture NG Threshold", WM8988_NGATE, 3, 31, 0), +SOC_ENUM("ALC Capture NG Type", ng_type), +SOC_SINGLE("ALC Capture NG Switch", WM8988_NGATE, 0, 1, 0), + +SOC_SINGLE("ZC Timeout Switch", WM8988_ADCTL1, 0, 1, 0), + +SOC_DOUBLE_R_TLV("Capture Digital Volume", WM8988_LADC, WM8988_RADC, + 0, 255, 0, adc_tlv), +SOC_DOUBLE_R_TLV("Capture Volume", WM8988_LINVOL, WM8988_RINVOL, + 0, 63, 0, pga_tlv), +SOC_DOUBLE_R("Capture ZC Switch", WM8988_LINVOL, WM8988_RINVOL, 6, 1, 0), +SOC_DOUBLE_R("Capture Switch", WM8988_LINVOL, WM8988_RINVOL, 7, 1, 1), + +SOC_ENUM("Playback De-emphasis", deemph), + +SOC_ENUM("Capture Polarity", adcpol), +SOC_SINGLE("Playback 6dB Attenuate", WM8988_ADCDAC, 7, 1, 0), +SOC_SINGLE("Capture 6dB Attenuate", WM8988_ADCDAC, 8, 1, 0), + +SOC_DOUBLE_R_TLV("PCM Volume", WM8988_LDAC, WM8988_RDAC, 0, 255, 0, dac_tlv), + +SOC_SINGLE_TLV("Left Mixer Left Bypass Volume", WM8988_LOUTM1, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Left Mixer Right Bypass Volume", WM8988_LOUTM2, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Right Mixer Left Bypass Volume", WM8988_ROUTM1, 4, 7, 1, + bypass_tlv), +SOC_SINGLE_TLV("Right Mixer Right Bypass Volume", WM8988_ROUTM2, 4, 7, 1, + bypass_tlv), + +SOC_DOUBLE_R("Output 1 Playback ZC Switch", WM8988_LOUT1V, + WM8988_ROUT1V, 7, 1, 0), +SOC_DOUBLE_R_TLV("Output 1 Playback Volume", WM8988_LOUT1V, WM8988_ROUT1V, + 0, 127, 0, out_tlv), + +SOC_DOUBLE_R("Output 2 Playback ZC Switch", WM8988_LOUT2V, + WM8988_ROUT2V, 7, 1, 0), +SOC_DOUBLE_R_TLV("Output 2 Playback Volume", WM8988_LOUT2V, WM8988_ROUT2V, + 0, 127, 0, out_tlv), + +}; + +/* + * DAPM Controls + */ + +static int wm8988_lrc_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 adctl2 = wm8988_read_reg_cache(codec, WM8988_ADCTL2); + + /* Use the DAC to gate LRC if active, otherwise use ADC */ + if (wm8988_read_reg_cache(codec, WM8988_PWR2) & 0x180) + adctl2 &= ~0x4; + else + adctl2 |= 0x4; + + return wm8988_write(codec, WM8988_ADCTL2, adctl2); +} + +static const char *wm8988_line_texts[] = { + "Line 1", "Line 2", "PGA", "Differential"}; + +static const unsigned int wm8988_line_values[] = { + 0, 1, 3, 4}; + +static const struct soc_enum wm8988_lline_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_LOUTM1, 0, 7, + ARRAY_SIZE(wm8988_line_texts), + wm8988_line_texts, + wm8988_line_values); +static const struct snd_kcontrol_new wm8988_left_line_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum); + +static const struct soc_enum wm8988_rline_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_ROUTM1, 0, 7, + ARRAY_SIZE(wm8988_line_texts), + wm8988_line_texts, + wm8988_line_values); +static const struct snd_kcontrol_new wm8988_right_line_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_lline_enum); + +/* Left Mixer */ +static const struct snd_kcontrol_new wm8988_left_mixer_controls[] = { + SOC_DAPM_SINGLE("Playback Switch", WM8988_LOUTM1, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_LOUTM1, 7, 1, 0), + SOC_DAPM_SINGLE("Right Playback Switch", WM8988_LOUTM2, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_LOUTM2, 7, 1, 0), +}; + +/* Right Mixer */ +static const struct snd_kcontrol_new wm8988_right_mixer_controls[] = { + SOC_DAPM_SINGLE("Left Playback Switch", WM8988_ROUTM1, 8, 1, 0), + SOC_DAPM_SINGLE("Left Bypass Switch", WM8988_ROUTM1, 7, 1, 0), + SOC_DAPM_SINGLE("Playback Switch", WM8988_ROUTM2, 8, 1, 0), + SOC_DAPM_SINGLE("Right Bypass Switch", WM8988_ROUTM2, 7, 1, 0), +}; + +static const char *wm8988_pga_sel[] = {"Line 1", "Line 2", "Differential"}; +static const unsigned int wm8988_pga_val[] = { 0, 1, 3 }; + +/* Left PGA Mux */ +static const struct soc_enum wm8988_lpga_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_LADCIN, 6, 3, + ARRAY_SIZE(wm8988_pga_sel), + wm8988_pga_sel, + wm8988_pga_val); +static const struct snd_kcontrol_new wm8988_left_pga_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_lpga_enum); + +/* Right PGA Mux */ +static const struct soc_enum wm8988_rpga_enum = + SOC_VALUE_ENUM_SINGLE(WM8988_RADCIN, 6, 3, + ARRAY_SIZE(wm8988_pga_sel), + wm8988_pga_sel, + wm8988_pga_val); +static const struct snd_kcontrol_new wm8988_right_pga_controls = + SOC_DAPM_VALUE_ENUM("Route", wm8988_rpga_enum); + +/* Differential Mux */ +static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"}; +static const struct soc_enum diffmux = + SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel); +static const struct snd_kcontrol_new wm8988_diffmux_controls = + SOC_DAPM_ENUM("Route", diffmux); + +/* Mono ADC Mux */ +static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)", + "Mono (Right)", "Digital Mono"}; +static const struct soc_enum monomux = + SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux); +static const struct snd_kcontrol_new wm8988_monomux_controls = + SOC_DAPM_ENUM("Route", monomux); + +static const struct snd_soc_dapm_widget wm8988_dapm_widgets[] = { + SND_SOC_DAPM_MICBIAS("Mic Bias", WM8988_PWR1, 1, 0), + + SND_SOC_DAPM_MUX("Differential Mux", SND_SOC_NOPM, 0, 0, + &wm8988_diffmux_controls), + SND_SOC_DAPM_MUX("Left ADC Mux", SND_SOC_NOPM, 0, 0, + &wm8988_monomux_controls), + SND_SOC_DAPM_MUX("Right ADC Mux", SND_SOC_NOPM, 0, 0, + &wm8988_monomux_controls), + + SND_SOC_DAPM_MUX("Left PGA Mux", WM8988_PWR1, 5, 0, + &wm8988_left_pga_controls), + SND_SOC_DAPM_MUX("Right PGA Mux", WM8988_PWR1, 4, 0, + &wm8988_right_pga_controls), + + SND_SOC_DAPM_MUX("Left Line Mux", SND_SOC_NOPM, 0, 0, + &wm8988_left_line_controls), + SND_SOC_DAPM_MUX("Right Line Mux", SND_SOC_NOPM, 0, 0, + &wm8988_right_line_controls), + + SND_SOC_DAPM_ADC("Right ADC", "Right Capture", WM8988_PWR1, 2, 0), + SND_SOC_DAPM_ADC("Left ADC", "Left Capture", WM8988_PWR1, 3, 0), + + SND_SOC_DAPM_DAC("Right DAC", "Right Playback", WM8988_PWR2, 7, 0), + SND_SOC_DAPM_DAC("Left DAC", "Left Playback", WM8988_PWR2, 8, 0), + + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + &wm8988_left_mixer_controls[0], + ARRAY_SIZE(wm8988_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + &wm8988_right_mixer_controls[0], + ARRAY_SIZE(wm8988_right_mixer_controls)), + + SND_SOC_DAPM_PGA("Right Out 2", WM8988_PWR2, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 2", WM8988_PWR2, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Right Out 1", WM8988_PWR2, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Left Out 1", WM8988_PWR2, 6, 0, NULL, 0), + + SND_SOC_DAPM_POST("LRC control", wm8988_lrc_control), + + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("VREF"), + + SND_SOC_DAPM_INPUT("LINPUT1"), + SND_SOC_DAPM_INPUT("LINPUT2"), + SND_SOC_DAPM_INPUT("RINPUT1"), + SND_SOC_DAPM_INPUT("RINPUT2"), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left PGA Mux", "Line 1", "LINPUT1" }, + { "Left PGA Mux", "Line 2", "LINPUT2" }, + { "Left PGA Mux", "Differential", "Differential Mux" }, + + { "Right PGA Mux", "Line 1", "RINPUT1" }, + { "Right PGA Mux", "Line 2", "RINPUT2" }, + { "Right PGA Mux", "Differential", "Differential Mux" }, + + { "Differential Mux", "Line 1", "LINPUT1" }, + { "Differential Mux", "Line 1", "RINPUT1" }, + { "Differential Mux", "Line 2", "LINPUT2" }, + { "Differential Mux", "Line 2", "RINPUT2" }, + + { "Left ADC Mux", "Stereo", "Left PGA Mux" }, + { "Left ADC Mux", "Mono (Left)", "Left PGA Mux" }, + { "Left ADC Mux", "Digital Mono", "Left PGA Mux" }, + + { "Right ADC Mux", "Stereo", "Right PGA Mux" }, + { "Right ADC Mux", "Mono (Right)", "Right PGA Mux" }, + { "Right ADC Mux", "Digital Mono", "Right PGA Mux" }, + + { "Left ADC", NULL, "Left ADC Mux" }, + { "Right ADC", NULL, "Right ADC Mux" }, + + { "Left Line Mux", "Line 1", "LINPUT1" }, + { "Left Line Mux", "Line 2", "LINPUT2" }, + { "Left Line Mux", "PGA", "Left PGA Mux" }, + { "Left Line Mux", "Differential", "Differential Mux" }, + + { "Right Line Mux", "Line 1", "RINPUT1" }, + { "Right Line Mux", "Line 2", "RINPUT2" }, + { "Right Line Mux", "PGA", "Right PGA Mux" }, + { "Right Line Mux", "Differential", "Differential Mux" }, + + { "Left Mixer", "Playback Switch", "Left DAC" }, + { "Left Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Left Mixer", "Right Playback Switch", "Right DAC" }, + { "Left Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "Right Mixer", "Left Playback Switch", "Left DAC" }, + { "Right Mixer", "Left Bypass Switch", "Left Line Mux" }, + { "Right Mixer", "Playback Switch", "Right DAC" }, + { "Right Mixer", "Right Bypass Switch", "Right Line Mux" }, + + { "Left Out 1", NULL, "Left Mixer" }, + { "LOUT1", NULL, "Left Out 1" }, + { "Right Out 1", NULL, "Right Mixer" }, + { "ROUT1", NULL, "Right Out 1" }, + + { "Left Out 2", NULL, "Left Mixer" }, + { "LOUT2", NULL, "Left Out 2" }, + { "Right Out 2", NULL, "Right Mixer" }, + { "ROUT2", NULL, "Right Out 2" }, +}; + +struct _coeff_div { + u32 mclk; + u32 rate; + u16 fs; + u8 sr:5; + u8 usb:1; +}; + +/* codec hifi mclk clock divider coefficients */ +static const struct _coeff_div coeff_div[] = { + /* 8k */ + {12288000, 8000, 1536, 0x6, 0x0}, + {11289600, 8000, 1408, 0x16, 0x0}, + {18432000, 8000, 2304, 0x7, 0x0}, + {16934400, 8000, 2112, 0x17, 0x0}, + {12000000, 8000, 1500, 0x6, 0x1}, + + /* 11.025k */ + {11289600, 11025, 1024, 0x18, 0x0}, + {16934400, 11025, 1536, 0x19, 0x0}, + {12000000, 11025, 1088, 0x19, 0x1}, + + /* 16k */ + {12288000, 16000, 768, 0xa, 0x0}, + {18432000, 16000, 1152, 0xb, 0x0}, + {12000000, 16000, 750, 0xa, 0x1}, + + /* 22.05k */ + {11289600, 22050, 512, 0x1a, 0x0}, + {16934400, 22050, 768, 0x1b, 0x0}, + {12000000, 22050, 544, 0x1b, 0x1}, + + /* 32k */ + {12288000, 32000, 384, 0xc, 0x0}, + {18432000, 32000, 576, 0xd, 0x0}, + {12000000, 32000, 375, 0xa, 0x1}, + + /* 44.1k */ + {11289600, 44100, 256, 0x10, 0x0}, + {16934400, 44100, 384, 0x11, 0x0}, + {12000000, 44100, 272, 0x11, 0x1}, + + /* 48k */ + {12288000, 48000, 256, 0x0, 0x0}, + {18432000, 48000, 384, 0x1, 0x0}, + {12000000, 48000, 250, 0x0, 0x1}, + + /* 88.2k */ + {11289600, 88200, 128, 0x1e, 0x0}, + {16934400, 88200, 192, 0x1f, 0x0}, + {12000000, 88200, 136, 0x1f, 0x1}, + + /* 96k */ + {12288000, 96000, 128, 0xe, 0x0}, + {18432000, 96000, 192, 0xf, 0x0}, + {12000000, 96000, 125, 0xe, 0x1}, +}; + +static inline int get_coeff(int mclk, int rate) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(coeff_div); i++) { + if (coeff_div[i].rate == rate && coeff_div[i].mclk == mclk) + return i; + } + + return -EINVAL; +} + +/* The set of rates we can generate from the above for each SYSCLK */ + +static unsigned int rates_12288[] = { + 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12288 = { + .count = ARRAY_SIZE(rates_12288), + .list = rates_12288, +}; + +static unsigned int rates_112896[] = { + 8000, 11025, 22050, 44100, +}; + +static struct snd_pcm_hw_constraint_list constraints_112896 = { + .count = ARRAY_SIZE(rates_112896), + .list = rates_112896, +}; + +static unsigned int rates_12[] = { + 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, + 48000, 88235, 96000, +}; + +static struct snd_pcm_hw_constraint_list constraints_12 = { + .count = ARRAY_SIZE(rates_12), + .list = rates_12, +}; + +/* + * Note that this should be called from init rather than from hw_params. + */ +static int wm8988_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct wm8988_priv *wm8988 = codec->private_data; + + switch (freq) { + case 11289600: + case 18432000: + case 22579200: + case 36864000: + wm8988->sysclk_constraints = &constraints_112896; + wm8988->sysclk = freq; + return 0; + + case 12288000: + case 16934400: + case 24576000: + case 33868800: + wm8988->sysclk_constraints = &constraints_12288; + wm8988->sysclk = freq; + return 0; + + case 12000000: + case 24000000: + wm8988->sysclk_constraints = &constraints_12; + wm8988->sysclk = freq; + return 0; + } + return -EINVAL; +} + +static int wm8988_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u16 iface = 0; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface = 0x0040; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0013; + break; + default: + return -EINVAL; + } + + /* clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x0090; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x0080; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x0010; + break; + default: + return -EINVAL; + } + + wm8988_write(codec, WM8988_IFACE, iface); + return 0; +} + +static int wm8988_pcm_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8988_priv *wm8988 = codec->private_data; + + /* The set of sample rates that can be supported depends on the + * MCLK supplied to the CODEC - enforce this. + */ + if (!wm8988->sysclk) { + dev_err(codec->dev, + "No MCLK configured, call set_sysclk() on init\n"); + return -EINVAL; + } + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + wm8988->sysclk_constraints); + + return 0; +} + +static int wm8988_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct wm8988_priv *wm8988 = codec->private_data; + u16 iface = wm8988_read_reg_cache(codec, WM8988_IFACE) & 0x1f3; + u16 srate = wm8988_read_reg_cache(codec, WM8988_SRATE) & 0x180; + int coeff; + + coeff = get_coeff(wm8988->sysclk, params_rate(params)); + if (coeff < 0) { + coeff = get_coeff(wm8988->sysclk / 2, params_rate(params)); + srate |= 0x40; + } + if (coeff < 0) { + dev_err(codec->dev, + "Unable to configure sample rate %dHz with %dHz MCLK\n", + params_rate(params), wm8988->sysclk); + return coeff; + } + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x0004; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x0008; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x000c; + break; + } + + /* set iface & srate */ + wm8988_write(codec, WM8988_IFACE, iface); + if (coeff >= 0) + wm8988_write(codec, WM8988_SRATE, srate | + (coeff_div[coeff].sr << 1) | coeff_div[coeff].usb); + + return 0; +} + +static int wm8988_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + u16 mute_reg = wm8988_read_reg_cache(codec, WM8988_ADCDAC) & 0xfff7; + + if (mute) + wm8988_write(codec, WM8988_ADCDAC, mute_reg | 0x8); + else + wm8988_write(codec, WM8988_ADCDAC, mute_reg); + return 0; +} + +static int wm8988_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u16 pwr_reg = wm8988_read_reg_cache(codec, WM8988_PWR1) & ~0x1c1; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* VREF, VMID=2x50k, digital enabled */ + wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x00c0); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* VREF, VMID=2x5k */ + wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x1c1); + + /* Charge caps */ + msleep(100); + } + + /* VREF, VMID=2*500k, digital stopped */ + wm8988_write(codec, WM8988_PWR1, pwr_reg | 0x0141); + break; + + case SND_SOC_BIAS_OFF: + wm8988_write(codec, WM8988_PWR1, 0x0000); + break; + } + codec->bias_level = level; + return 0; +} + +#define WM8988_RATES SNDRV_PCM_RATE_8000_96000 + +#define WM8988_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_ops wm8988_ops = { + .startup = wm8988_pcm_startup, + .hw_params = wm8988_pcm_hw_params, + .set_fmt = wm8988_set_dai_fmt, + .set_sysclk = wm8988_set_dai_sysclk, + .digital_mute = wm8988_mute, +}; + +struct snd_soc_dai wm8988_dai = { + .name = "WM8988", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM8988_RATES, + .formats = WM8988_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM8988_RATES, + .formats = WM8988_FORMATS, + }, + .ops = &wm8988_ops, + .symmetric_rates = 1, +}; +EXPORT_SYMBOL_GPL(wm8988_dai); + +static int wm8988_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8988_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int wm8988_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < WM8988_NUM_REG; i++) { + if (i == WM8988_RESET) + continue; + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8988_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +static struct snd_soc_codec *wm8988_codec; + +static int wm8988_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8988_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8988_codec; + codec = wm8988_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, wm8988_snd_controls, + ARRAY_SIZE(wm8988_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8988_dapm_widgets, + ARRAY_SIZE(wm8988_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int wm8988_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_wm8988 = { + .probe = wm8988_probe, + .remove = wm8988_remove, + .suspend = wm8988_suspend, + .resume = wm8988_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8988); + +static int wm8988_register(struct wm8988_priv *wm8988) +{ + struct snd_soc_codec *codec = &wm8988->codec; + int ret; + u16 reg; + + if (wm8988_codec) { + dev_err(codec->dev, "Another WM8988 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8988; + codec->name = "WM8988"; + codec->owner = THIS_MODULE; + codec->read = wm8988_read_reg_cache; + codec->write = wm8988_write; + codec->dai = &wm8988_dai; + codec->num_dai = 1; + codec->reg_cache_size = ARRAY_SIZE(wm8988->reg_cache); + codec->reg_cache = &wm8988->reg_cache; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8988_set_bias_level; + + memcpy(codec->reg_cache, wm8988_reg, + sizeof(wm8988_reg)); + + ret = wm8988_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset\n"); + return ret; + } + + /* set the update bits (we always update left then right) */ + reg = wm8988_read_reg_cache(codec, WM8988_RADC); + wm8988_write(codec, WM8988_RADC, reg | 0x100); + reg = wm8988_read_reg_cache(codec, WM8988_RDAC); + wm8988_write(codec, WM8988_RDAC, reg | 0x0100); + reg = wm8988_read_reg_cache(codec, WM8988_ROUT1V); + wm8988_write(codec, WM8988_ROUT1V, reg | 0x0100); + reg = wm8988_read_reg_cache(codec, WM8988_ROUT2V); + wm8988_write(codec, WM8988_ROUT2V, reg | 0x0100); + reg = wm8988_read_reg_cache(codec, WM8988_RINVOL); + wm8988_write(codec, WM8988_RINVOL, reg | 0x0100); + + wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_STANDBY); + + wm8988_dai.dev = codec->dev; + + wm8988_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + return ret; + } + + ret = snd_soc_register_dai(&wm8988_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + return ret; + } + + return 0; + +err: + kfree(wm8988); + return ret; +} + +static void wm8988_unregister(struct wm8988_priv *wm8988) +{ + wm8988_set_bias_level(&wm8988->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&wm8988_dai); + snd_soc_unregister_codec(&wm8988->codec); + kfree(wm8988); + wm8988_codec = NULL; +} + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static int wm8988_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8988_priv *wm8988; + struct snd_soc_codec *codec; + + wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL); + if (wm8988 == NULL) + return -ENOMEM; + + codec = &wm8988->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8988); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8988_register(wm8988); +} + +static int wm8988_i2c_remove(struct i2c_client *client) +{ + struct wm8988_priv *wm8988 = i2c_get_clientdata(client); + wm8988_unregister(wm8988); + return 0; +} + +static const struct i2c_device_id wm8988_i2c_id[] = { + { "wm8988", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8988_i2c_id); + +static struct i2c_driver wm8988_i2c_driver = { + .driver = { + .name = "WM8988", + .owner = THIS_MODULE, + }, + .probe = wm8988_i2c_probe, + .remove = wm8988_i2c_remove, + .id_table = wm8988_i2c_id, +}; +#endif + +#if defined(CONFIG_SPI_MASTER) +static int wm8988_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} + +static int __devinit wm8988_spi_probe(struct spi_device *spi) +{ + struct wm8988_priv *wm8988; + struct snd_soc_codec *codec; + + wm8988 = kzalloc(sizeof(struct wm8988_priv), GFP_KERNEL); + if (wm8988 == NULL) + return -ENOMEM; + + codec = &wm8988->codec; + codec->hw_write = (hw_write_t)wm8988_spi_write; + codec->control_data = spi; + codec->dev = &spi->dev; + + spi->dev.driver_data = wm8988; + + return wm8988_register(wm8988); +} + +static int __devexit wm8988_spi_remove(struct spi_device *spi) +{ + struct wm8988_priv *wm8988 = spi->dev.driver_data; + + wm8988_unregister(wm8988); + + return 0; +} + +static struct spi_driver wm8988_spi_driver = { + .driver = { + .name = "wm8988", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8988_spi_probe, + .remove = __devexit_p(wm8988_spi_remove), +}; +#endif + +static int __init wm8988_modinit(void) +{ + int ret; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8988_i2c_driver); + if (ret != 0) + pr_err("WM8988: Unable to register I2C driver: %d\n", ret); +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8988_spi_driver); + if (ret != 0) + pr_err("WM8988: Unable to register SPI driver: %d\n", ret); +#endif + return ret; +} +module_init(wm8988_modinit); + +static void __exit wm8988_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8988_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8988_spi_driver); +#endif +} +module_exit(wm8988_exit); + + +MODULE_DESCRIPTION("ASoC WM8988 driver"); +MODULE_AUTHOR("Mark Brown <broonie@opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8988.h b/sound/soc/codecs/wm8988.h new file mode 100644 index 0000000..4552d37 --- /dev/null +++ b/sound/soc/codecs/wm8988.h @@ -0,0 +1,60 @@ +/* + * Copyright 2005 Openedhand Ltd. + * + * Author: Richard Purdie <richard@openedhand.com> + * + * Based on WM8753.h + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef _WM8988_H +#define _WM8988_H + +/* WM8988 register space */ + +#define WM8988_LINVOL 0x00 +#define WM8988_RINVOL 0x01 +#define WM8988_LOUT1V 0x02 +#define WM8988_ROUT1V 0x03 +#define WM8988_ADCDAC 0x05 +#define WM8988_IFACE 0x07 +#define WM8988_SRATE 0x08 +#define WM8988_LDAC 0x0a +#define WM8988_RDAC 0x0b +#define WM8988_BASS 0x0c +#define WM8988_TREBLE 0x0d +#define WM8988_RESET 0x0f +#define WM8988_3D 0x10 +#define WM8988_ALC1 0x11 +#define WM8988_ALC2 0x12 +#define WM8988_ALC3 0x13 +#define WM8988_NGATE 0x14 +#define WM8988_LADC 0x15 +#define WM8988_RADC 0x16 +#define WM8988_ADCTL1 0x17 +#define WM8988_ADCTL2 0x18 +#define WM8988_PWR1 0x19 +#define WM8988_PWR2 0x1a +#define WM8988_ADCTL3 0x1b +#define WM8988_ADCIN 0x1f +#define WM8988_LADCIN 0x20 +#define WM8988_RADCIN 0x21 +#define WM8988_LOUTM1 0x22 +#define WM8988_LOUTM2 0x23 +#define WM8988_ROUTM1 0x24 +#define WM8988_ROUTM2 0x25 +#define WM8988_LOUT2V 0x28 +#define WM8988_ROUT2V 0x29 +#define WM8988_LPPB 0x43 +#define WM8988_NUM_REG 0x44 + +#define WM8988_SYSCLK 0 + +extern struct snd_soc_dai wm8988_dai; +extern struct snd_soc_codec_device soc_codec_dev_wm8988; + +#endif diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 523bad0..a6feb784 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -189,6 +189,26 @@ SOC_SINGLE("3D Lower Cut-off Switch", AC97_REC_GAIN_MIC, 4, 1, 0), SOC_SINGLE("3D Depth", AC97_REC_GAIN_MIC, 0, 15, 1), }; +static int wm9713_voice_shutdown(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u16 status, rate; + + BUG_ON(event != SND_SOC_DAPM_PRE_PMD); + + /* Gracefully shut down the voice interface. */ + status = ac97_read(codec, AC97_EXTENDED_MID) | 0x1000; + rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; + ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); + schedule_timeout_interruptible(msecs_to_jiffies(1)); + ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); + ac97_write(codec, AC97_EXTENDED_MID, status); + + return 0; +} + + /* We have to create a fake left and right HP mixers because * the codec only has a single control that is shared by both channels. * This makes it impossible to determine the audio path using the current @@ -400,7 +420,8 @@ SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("HP Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Line Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("Capture Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), -SND_SOC_DAPM_DAC("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1), +SND_SOC_DAPM_DAC_E("Voice DAC", "Voice Playback", AC97_EXTENDED_MID, 12, 1, + wm9713_voice_shutdown, SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_DAC("Aux DAC", "Aux Playback", AC97_EXTENDED_MID, 11, 1), SND_SOC_DAPM_PGA("Left ADC", AC97_EXTENDED_MID, 5, 1, NULL, 0), SND_SOC_DAPM_PGA("Right ADC", AC97_EXTENDED_MID, 4, 1, NULL, 0), @@ -936,21 +957,6 @@ static int wm9713_pcm_hw_params(struct snd_pcm_substream *substream, return 0; } -static void wm9713_voiceshutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - u16 status, rate; - - /* Gracefully shut down the voice interface. */ - status = ac97_read(codec, AC97_EXTENDED_STATUS) | 0x1000; - rate = ac97_read(codec, AC97_HANDSET_RATE) & 0xF0FF; - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0200); - schedule_timeout_interruptible(msecs_to_jiffies(1)); - ac97_write(codec, AC97_HANDSET_RATE, rate | 0x0F00); - ac97_write(codec, AC97_EXTENDED_MID, status); -} - static int ac97_hifi_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -1019,7 +1025,6 @@ static struct snd_soc_dai_ops wm9713_dai_ops_aux = { static struct snd_soc_dai_ops wm9713_dai_ops_voice = { .hw_params = wm9713_pcm_hw_params, - .shutdown = wm9713_voiceshutdown, .set_clkdiv = wm9713_set_dai_clkdiv, .set_pll = wm9713_set_dai_pll, .set_fmt = wm9713_set_dai_fmt, @@ -1069,6 +1074,7 @@ struct snd_soc_dai wm9713_dai[] = { .rates = WM9713_PCM_RATES, .formats = WM9713_PCM_FORMATS,}, .ops = &wm9713_dai_ops_voice, + .symmetric_rates = 1, }, }; EXPORT_SYMBOL_GPL(wm9713_dai); diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 91ef179..b60b1df 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -383,10 +383,9 @@ static int __init n810_soc_init(void) clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); - if (gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) - BUG(); - if (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0) - BUG(); + BUG_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) || + (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0)); + gpio_direction_output(N810_HEADSET_AMP_GPIO, 0); gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 9126142..495192a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -215,8 +215,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; - int wlen, channels; + int wlen, channels, wpf; unsigned long port; + unsigned int format; if (cpu_class_is_omap1()) { dma = omap1_dma_reqs[bus_id][substream->stream]; @@ -244,18 +245,23 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, return 0; } - channels = params_channels(params); + format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK; + wpf = channels = params_channels(params); switch (channels) { case 2: - /* Use dual-phase frames */ - regs->rcr2 |= RPHASE; - regs->xcr2 |= XPHASE; + if (format == SND_SOC_DAIFMT_I2S) { + /* Use dual-phase frames */ + regs->rcr2 |= RPHASE; + regs->xcr2 |= XPHASE; + /* Set 1 word per (McBSP) frame for phase1 and phase2 */ + wpf--; + regs->rcr2 |= RFRLEN2(wpf - 1); + regs->xcr2 |= XFRLEN2(wpf - 1); + } case 1: - /* Set 1 word per (McBSP) frame */ - regs->rcr2 |= RFRLEN2(1 - 1); - regs->rcr1 |= RFRLEN1(1 - 1); - regs->xcr2 |= XFRLEN2(1 - 1); - regs->xcr1 |= XFRLEN1(1 - 1); + /* Set word per (McBSP) frame for phase1 */ + regs->rcr1 |= RFRLEN1(wpf - 1); + regs->xcr1 |= XFRLEN1(wpf - 1); break; default: /* Unsupported number of channels */ @@ -277,11 +283,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, } /* Set FS period and length in terms of bit clock periods */ - switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + switch (format) { case SND_SOC_DAIFMT_I2S: - regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr2 |= FPER(wlen * channels - 1); regs->srgr1 |= FWID(wlen - 1); break; + case SND_SOC_DAIFMT_DSP_A: case SND_SOC_DAIFMT_DSP_B: regs->srgr2 |= FPER(wlen * channels - 1); regs->srgr1 |= FWID(0); @@ -326,6 +333,13 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, regs->rcr2 |= RDATDLY(1); regs->xcr2 |= XDATDLY(1); break; + case SND_SOC_DAIFMT_DSP_A: + /* 1-bit data delay */ + regs->rcr2 |= RDATDLY(1); + regs->xcr2 |= XDATDLY(1); + /* Invert FS polarity configuration */ + temp_fmt ^= SND_SOC_DAIFMT_NB_IF; + break; case SND_SOC_DAIFMT_DSP_B: /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index b078ed5..07cf7f4 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -328,7 +328,7 @@ int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &omap_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = omap_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 0c2322d..027e1a4 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap2evm_ops, }; diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index fd24a4a..6aa428e 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link omap3beagle_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3beagle_ops, }; diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fe282d4..ad219aa 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .name = "PCM1773", .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_out_ops, .init = omap3pandora_out_init, }, { .name = "TWL4030", .stream_name = "Line/Mic In", .cpu_dai = &omap_mcbsp_dai[1], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_in_ops, .init = omap3pandora_in_init, } diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index a72dc4e..ec4f8fd 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &overo_ops, }; diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 10f1c86..1c79741 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -197,7 +197,7 @@ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, }; diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 02263e5..d5be2b3 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -26,8 +26,6 @@ #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <mach/pxa-regs.h> -#include <mach/hardware.h> #include <mach/corgi.h> #include <mach/audio.h> diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index fe4a729..949be9c 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -29,8 +29,6 @@ #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <mach/pxa-regs.h> -#include <mach/hardware.h> #include <mach/audio.h> #include "../codecs/wm9712.h" diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index ef7c6c8..a51058f 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -27,8 +27,6 @@ #include <asm/mach-types.h> #include <asm/hardware/locomo.h> -#include <mach/pxa-regs.h> -#include <mach/hardware.h> #include <mach/poodle.h> #include <mach/audio.h> diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 74ff69e..b9b61dd 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -30,7 +30,7 @@ #include <sound/pxa2xx-lib.h> #include <mach/hardware.h> -#include <mach/pxa-regs.h> +#include <mach/dma.h> #include <mach/regs-ssp.h> #include <mach/audio.h> #include <mach/ssp.h> @@ -589,7 +589,10 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_NB_IF: break; case SND_SOC_DAIFMT_IB_IF: - sspsp |= SSPSP_SCMODE(3); + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; break; default: return -EINVAL; @@ -606,7 +609,13 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, case SND_SOC_DAIFMT_NB_NF: sspsp |= SSPSP_SFRMP; break; + case SND_SOC_DAIFMT_NB_IF: + break; case SND_SOC_DAIFMT_IB_IF: + sspsp |= SSPSP_SCMODE(2); + break; + case SND_SOC_DAIFMT_IB_NF: + sspsp |= SSPSP_SCMODE(2) | SSPSP_SFRMP; break; default: return -EINVAL; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 01c21c6..d9c94d7 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -20,8 +20,8 @@ #include <sound/pxa2xx-lib.h> #include <mach/hardware.h> -#include <mach/pxa-regs.h> #include <mach/regs-ac97.h> +#include <mach/dma.h> #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index e6c2440..2f4b6e4 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -24,7 +24,7 @@ #include <sound/pxa2xx-lib.h> #include <mach/hardware.h> -#include <mach/pxa-regs.h> +#include <mach/dma.h> #include <mach/audio.h> #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 53b9fb1..d38e395 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -81,7 +81,7 @@ static struct snd_pcm_ops pxa2xx_pcm_ops = { .mmap = pxa2xx_pcm_mmap, }; -static u64 pxa2xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) @@ -91,7 +91,7 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->dma_mask) card->dev->dma_mask = &pxa2xx_pcm_dmamask; if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_32BIT_MASK; + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); if (dai->playback.channels_min) { ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 6ca9f53..c4cd2ac 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -26,8 +26,6 @@ #include <sound/soc-dapm.h> #include <asm/mach-types.h> -#include <mach/pxa-regs.h> -#include <mach/hardware.h> #include <mach/spitz.h> #include "../codecs/wm8750.h" #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index fc78137..dbbd3e9 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -30,8 +30,6 @@ #include <asm/mach-types.h> #include <mach/tosa.h> -#include <mach/pxa-regs.h> -#include <mach/hardware.h> #include <mach/audio.h> #include "../codecs/wm9712.h" diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index a9d68fa..169ddad 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -419,7 +419,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK; +static u64 s3c24xx_pcm_dmamask = DMA_BIT_MASK(32); static int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, struct snd_pcm *pcm) diff --git a/sound/soc/s6000/Kconfig b/sound/soc/s6000/Kconfig new file mode 100644 index 0000000..c74eb3d --- /dev/null +++ b/sound/soc/s6000/Kconfig @@ -0,0 +1,19 @@ +config SND_S6000_SOC + tristate "SoC Audio for the Stretch s6000 family" + depends on XTENSA_VARIANT_S6000 + help + Say Y or M if you want to add support for codecs attached to + s6000 family chips. You will also need to select the platform + to support below. + +config SND_S6000_SOC_I2S + tristate + +config SND_S6000_SOC_S6IPCAM + tristate "SoC Audio support for Stretch 6105 IP Camera" + depends on SND_S6000_SOC && XTENSA_PLATFORM_S6105 + select SND_S6000_SOC_I2S + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on the + Stretch s6105 IP Camera Reference Design. diff --git a/sound/soc/s6000/Makefile b/sound/soc/s6000/Makefile new file mode 100644 index 0000000..7a61361 --- /dev/null +++ b/sound/soc/s6000/Makefile @@ -0,0 +1,11 @@ +# s6000 Platform Support +snd-soc-s6000-objs := s6000-pcm.o +snd-soc-s6000-i2s-objs := s6000-i2s.o + +obj-$(CONFIG_SND_S6000_SOC) += snd-soc-s6000.o +obj-$(CONFIG_SND_S6000_SOC_I2S) += snd-soc-s6000-i2s.o + +# s6105 Machine Support +snd-soc-s6ipcam-objs := s6105-ipcam.o + +obj-$(CONFIG_SND_S6000_SOC_S6IPCAM) += snd-soc-s6ipcam.o diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c new file mode 100644 index 0000000..c5cda18 --- /dev/null +++ b/sound/soc/s6000/s6000-i2s.c @@ -0,0 +1,629 @@ +/* + * ALSA SoC I2S Audio Layer for the Stretch S6000 family + * + * Author: Daniel Gloeckner, <dg@emlix.com> + * Copyright: (C) 2009 emlix GmbH <info@emlix.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/interrupt.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "s6000-i2s.h" +#include "s6000-pcm.h" + +struct s6000_i2s_dev { + dma_addr_t sifbase; + u8 __iomem *scbbase; + unsigned int wide; + unsigned int channel_in; + unsigned int channel_out; + unsigned int lines_in; + unsigned int lines_out; + struct s6000_pcm_dma_params dma_params; +}; + +#define S6_I2S_INTERRUPT_STATUS 0x00 +#define S6_I2S_INT_OVERRUN 1 +#define S6_I2S_INT_UNDERRUN 2 +#define S6_I2S_INT_ALIGNMENT 4 +#define S6_I2S_INTERRUPT_ENABLE 0x04 +#define S6_I2S_INTERRUPT_RAW 0x08 +#define S6_I2S_INTERRUPT_CLEAR 0x0C +#define S6_I2S_INTERRUPT_SET 0x10 +#define S6_I2S_MODE 0x20 +#define S6_I2S_DUAL 0 +#define S6_I2S_WIDE 1 +#define S6_I2S_TX_DEFAULT 0x24 +#define S6_I2S_DATA_CFG(c) (0x40 + 0x10 * (c)) +#define S6_I2S_IN 0 +#define S6_I2S_OUT 1 +#define S6_I2S_UNUSED 2 +#define S6_I2S_INTERFACE_CFG(c) (0x44 + 0x10 * (c)) +#define S6_I2S_DIV_MASK 0x001fff +#define S6_I2S_16BIT 0x000000 +#define S6_I2S_20BIT 0x002000 +#define S6_I2S_24BIT 0x004000 +#define S6_I2S_32BIT 0x006000 +#define S6_I2S_BITS_MASK 0x006000 +#define S6_I2S_MEM_16BIT 0x000000 +#define S6_I2S_MEM_32BIT 0x008000 +#define S6_I2S_MEM_MASK 0x008000 +#define S6_I2S_CHANNELS_SHIFT 16 +#define S6_I2S_CHANNELS_MASK 0x030000 +#define S6_I2S_SCK_IN 0x000000 +#define S6_I2S_SCK_OUT 0x040000 +#define S6_I2S_SCK_DIR 0x040000 +#define S6_I2S_WS_IN 0x000000 +#define S6_I2S_WS_OUT 0x080000 +#define S6_I2S_WS_DIR 0x080000 +#define S6_I2S_LEFT_FIRST 0x000000 +#define S6_I2S_RIGHT_FIRST 0x100000 +#define S6_I2S_FIRST 0x100000 +#define S6_I2S_CUR_SCK 0x200000 +#define S6_I2S_CUR_WS 0x400000 +#define S6_I2S_ENABLE(c) (0x48 + 0x10 * (c)) +#define S6_I2S_DISABLE_IF 0x02 +#define S6_I2S_ENABLE_IF 0x03 +#define S6_I2S_IS_BUSY 0x04 +#define S6_I2S_DMA_ACTIVE 0x08 +#define S6_I2S_IS_ENABLED 0x10 + +#define S6_I2S_NUM_LINES 4 + +#define S6_I2S_SIF_PORT0 0x0000000 +#define S6_I2S_SIF_PORT1 0x0000080 /* docs say 0x0000010 */ + +static inline void s6_i2s_write_reg(struct s6000_i2s_dev *dev, int reg, u32 val) +{ + writel(val, dev->scbbase + reg); +} + +static inline u32 s6_i2s_read_reg(struct s6000_i2s_dev *dev, int reg) +{ + return readl(dev->scbbase + reg); +} + +static inline void s6_i2s_mod_reg(struct s6000_i2s_dev *dev, int reg, + u32 mask, u32 val) +{ + val ^= s6_i2s_read_reg(dev, reg) & ~mask; + s6_i2s_write_reg(dev, reg, val); +} + +static void s6000_i2s_start_channel(struct s6000_i2s_dev *dev, int channel) +{ + int i, j, cur, prev; + + /* + * Wait for WCLK to toggle 5 times before enabling the channel + * s6000 Family Datasheet 3.6.4: + * "At least two cycles of WS must occur between commands + * to disable or enable the interface" + */ + j = 0; + prev = ~S6_I2S_CUR_WS; + for (i = 1000000; --i && j < 6; ) { + cur = s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(channel)) + & S6_I2S_CUR_WS; + if (prev != cur) { + prev = cur; + j++; + } + } + if (j < 6) + printk(KERN_WARNING "s6000-i2s: timeout waiting for WCLK\n"); + + s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_ENABLE_IF); +} + +static void s6000_i2s_stop_channel(struct s6000_i2s_dev *dev, int channel) +{ + s6_i2s_write_reg(dev, S6_I2S_ENABLE(channel), S6_I2S_DISABLE_IF); +} + +static void s6000_i2s_start(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data; + int channel; + + channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + dev->channel_out : dev->channel_in; + + s6000_i2s_start_channel(dev, channel); +} + +static void s6000_i2s_stop(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s6000_i2s_dev *dev = rtd->dai->cpu_dai->private_data; + int channel; + + channel = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + dev->channel_out : dev->channel_in; + + s6000_i2s_stop_channel(dev, channel); +} + +static int s6000_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + int after) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if ((substream->stream == SNDRV_PCM_STREAM_CAPTURE) ^ !after) + s6000_i2s_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (!after) + s6000_i2s_stop(substream); + } + return 0; +} + +static unsigned int s6000_i2s_int_sources(struct s6000_i2s_dev *dev) +{ + unsigned int pending; + pending = s6_i2s_read_reg(dev, S6_I2S_INTERRUPT_RAW); + pending &= S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN; + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, pending); + + return pending; +} + +static unsigned int s6000_i2s_check_xrun(struct snd_soc_dai *cpu_dai) +{ + struct s6000_i2s_dev *dev = cpu_dai->private_data; + unsigned int errors; + unsigned int ret; + + errors = s6000_i2s_int_sources(dev); + if (likely(!errors)) + return 0; + + ret = 0; + if (errors & S6_I2S_INT_ALIGNMENT) + printk(KERN_ERR "s6000-i2s: WCLK misaligned\n"); + if (errors & S6_I2S_INT_UNDERRUN) + ret |= 1 << SNDRV_PCM_STREAM_PLAYBACK; + if (errors & S6_I2S_INT_OVERRUN) + ret |= 1 << SNDRV_PCM_STREAM_CAPTURE; + return ret; +} + +static void s6000_i2s_wait_disabled(struct s6000_i2s_dev *dev) +{ + int channel; + int n = 50; + for (channel = 0; channel < 2; channel++) { + while (--n >= 0) { + int v = s6_i2s_read_reg(dev, S6_I2S_ENABLE(channel)); + if ((v & S6_I2S_IS_ENABLED) + || !(v & (S6_I2S_DMA_ACTIVE | S6_I2S_IS_BUSY))) + break; + udelay(20); + } + } + if (n < 0) + printk(KERN_WARNING "s6000-i2s: timeout disabling interfaces"); +} + +static int s6000_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct s6000_i2s_dev *dev = cpu_dai->private_data; + u32 w; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + w = S6_I2S_SCK_IN | S6_I2S_WS_IN; + break; + case SND_SOC_DAIFMT_CBS_CFM: + w = S6_I2S_SCK_OUT | S6_I2S_WS_IN; + break; + case SND_SOC_DAIFMT_CBM_CFS: + w = S6_I2S_SCK_IN | S6_I2S_WS_OUT; + break; + case SND_SOC_DAIFMT_CBS_CFS: + w = S6_I2S_SCK_OUT | S6_I2S_WS_OUT; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + w |= S6_I2S_LEFT_FIRST; + break; + case SND_SOC_DAIFMT_NB_IF: + w |= S6_I2S_RIGHT_FIRST; + break; + default: + return -EINVAL; + } + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(0), + S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w); + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(1), + S6_I2S_FIRST | S6_I2S_WS_DIR | S6_I2S_SCK_DIR, w); + + return 0; +} + +static int s6000_i2s_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) +{ + struct s6000_i2s_dev *dev = dai->private_data; + + if (!div || (div & 1) || div > (S6_I2S_DIV_MASK + 1) * 2) + return -EINVAL; + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(div_id), + S6_I2S_DIV_MASK, div / 2 - 1); + return 0; +} + +static int s6000_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct s6000_i2s_dev *dev = dai->private_data; + int interf; + u32 w = 0; + + if (dev->wide) + interf = 0; + else { + w |= (((params_channels(params) - 2) / 2) + << S6_I2S_CHANNELS_SHIFT) & S6_I2S_CHANNELS_MASK; + interf = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + ? dev->channel_out : dev->channel_in; + } + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + w |= S6_I2S_16BIT | S6_I2S_MEM_16BIT; + break; + case SNDRV_PCM_FORMAT_S32_LE: + w |= S6_I2S_32BIT | S6_I2S_MEM_32BIT; + break; + default: + printk(KERN_WARNING "s6000-i2s: unsupported PCM format %x\n", + params_format(params)); + return -EINVAL; + } + + if (s6_i2s_read_reg(dev, S6_I2S_INTERFACE_CFG(interf)) + & S6_I2S_IS_ENABLED) { + printk(KERN_ERR "s6000-i2s: interface already enabled\n"); + return -EBUSY; + } + + s6_i2s_mod_reg(dev, S6_I2S_INTERFACE_CFG(interf), + S6_I2S_CHANNELS_MASK|S6_I2S_MEM_MASK|S6_I2S_BITS_MASK, + w); + + return 0; +} + +static int s6000_i2s_dai_probe(struct platform_device *pdev, + struct snd_soc_dai *dai) +{ + struct s6000_i2s_dev *dev = dai->private_data; + struct s6000_snd_platform_data *pdata = pdev->dev.platform_data; + + if (!pdata) + return -EINVAL; + + dev->wide = pdata->wide; + dev->channel_in = pdata->channel_in; + dev->channel_out = pdata->channel_out; + dev->lines_in = pdata->lines_in; + dev->lines_out = pdata->lines_out; + + s6_i2s_write_reg(dev, S6_I2S_MODE, + dev->wide ? S6_I2S_WIDE : S6_I2S_DUAL); + + if (dev->wide) { + int i; + + if (dev->lines_in + dev->lines_out > S6_I2S_NUM_LINES) + return -EINVAL; + + dev->channel_in = 0; + dev->channel_out = 1; + dai->capture.channels_min = 2 * dev->lines_in; + dai->capture.channels_max = dai->capture.channels_min; + dai->playback.channels_min = 2 * dev->lines_out; + dai->playback.channels_max = dai->playback.channels_min; + + for (i = 0; i < dev->lines_out; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_OUT); + + for (; i < S6_I2S_NUM_LINES - dev->lines_in; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), + S6_I2S_UNUSED); + + for (; i < S6_I2S_NUM_LINES; i++) + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(i), S6_I2S_IN); + } else { + unsigned int cfg[2] = {S6_I2S_UNUSED, S6_I2S_UNUSED}; + + if (dev->lines_in > 1 || dev->lines_out > 1) + return -EINVAL; + + dai->capture.channels_min = 2 * dev->lines_in; + dai->capture.channels_max = 8 * dev->lines_in; + dai->playback.channels_min = 2 * dev->lines_out; + dai->playback.channels_max = 8 * dev->lines_out; + + if (dev->lines_in) + cfg[dev->channel_in] = S6_I2S_IN; + if (dev->lines_out) + cfg[dev->channel_out] = S6_I2S_OUT; + + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(0), cfg[0]); + s6_i2s_write_reg(dev, S6_I2S_DATA_CFG(1), cfg[1]); + } + + if (dev->lines_out) { + if (dev->lines_in) { + if (!dev->dma_params.dma_out) + return -ENODEV; + } else { + dev->dma_params.dma_out = dev->dma_params.dma_in; + dev->dma_params.dma_in = 0; + } + } + dev->dma_params.sif_in = dev->sifbase + (dev->channel_in ? + S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0); + dev->dma_params.sif_out = dev->sifbase + (dev->channel_out ? + S6_I2S_SIF_PORT1 : S6_I2S_SIF_PORT0); + dev->dma_params.same_rate = pdata->same_rate | pdata->wide; + return 0; +} + +#define S6000_I2S_RATES (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000) +#define S6000_I2S_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops s6000_i2s_dai_ops = { + .set_fmt = s6000_i2s_set_dai_fmt, + .set_clkdiv = s6000_i2s_set_clkdiv, + .hw_params = s6000_i2s_hw_params, +}; + +struct snd_soc_dai s6000_i2s_dai = { + .name = "s6000-i2s", + .id = 0, + .probe = s6000_i2s_dai_probe, + .playback = { + .channels_min = 2, + .channels_max = 8, + .formats = S6000_I2S_FORMATS, + .rates = S6000_I2S_RATES, + .rate_min = 0, + .rate_max = 1562500, + }, + .capture = { + .channels_min = 2, + .channels_max = 8, + .formats = S6000_I2S_FORMATS, + .rates = S6000_I2S_RATES, + .rate_min = 0, + .rate_max = 1562500, + }, + .ops = &s6000_i2s_dai_ops, +} +EXPORT_SYMBOL_GPL(s6000_i2s_dai); + +static int __devinit s6000_i2s_probe(struct platform_device *pdev) +{ + struct s6000_i2s_dev *dev; + struct resource *scbmem, *sifmem, *region, *dma1, *dma2; + u8 __iomem *mmio; + int ret; + + scbmem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!scbmem) { + dev_err(&pdev->dev, "no mem resource?\n"); + ret = -ENODEV; + goto err_release_none; + } + + region = request_mem_region(scbmem->start, + scbmem->end - scbmem->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S SCB region already claimed\n"); + ret = -EBUSY; + goto err_release_none; + } + + mmio = ioremap(scbmem->start, scbmem->end - scbmem->start + 1); + if (!mmio) { + dev_err(&pdev->dev, "can't ioremap SCB region\n"); + ret = -ENOMEM; + goto err_release_scb; + } + + sifmem = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!sifmem) { + dev_err(&pdev->dev, "no second mem resource?\n"); + ret = -ENODEV; + goto err_release_map; + } + + region = request_mem_region(sifmem->start, + sifmem->end - sifmem->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S SIF region already claimed\n"); + ret = -EBUSY; + goto err_release_map; + } + + dma1 = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dma1) { + dev_err(&pdev->dev, "no dma resource?\n"); + ret = -ENODEV; + goto err_release_sif; + } + + region = request_mem_region(dma1->start, dma1->end - dma1->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, "I2S DMA region already claimed\n"); + ret = -EBUSY; + goto err_release_sif; + } + + dma2 = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (dma2) { + region = request_mem_region(dma2->start, + dma2->end - dma2->start + 1, + pdev->name); + if (!region) { + dev_err(&pdev->dev, + "I2S DMA region already claimed\n"); + ret = -EBUSY; + goto err_release_dma1; + } + } + + dev = kzalloc(sizeof(struct s6000_i2s_dev), GFP_KERNEL); + if (!dev) { + ret = -ENOMEM; + goto err_release_dma2; + } + + s6000_i2s_dai.dev = &pdev->dev; + s6000_i2s_dai.private_data = dev; + s6000_i2s_dai.dma_data = &dev->dma_params; + + dev->sifbase = sifmem->start; + dev->scbbase = mmio; + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0); + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_CLEAR, + S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN); + + s6000_i2s_stop_channel(dev, 0); + s6000_i2s_stop_channel(dev, 1); + s6000_i2s_wait_disabled(dev); + + dev->dma_params.check_xrun = s6000_i2s_check_xrun; + dev->dma_params.trigger = s6000_i2s_trigger; + dev->dma_params.dma_in = dma1->start; + dev->dma_params.dma_out = dma2 ? dma2->start : 0; + dev->dma_params.irq = platform_get_irq(pdev, 0); + if (dev->dma_params.irq < 0) { + dev_err(&pdev->dev, "no irq resource?\n"); + ret = -ENODEV; + goto err_release_dev; + } + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, + S6_I2S_INT_ALIGNMENT | + S6_I2S_INT_UNDERRUN | + S6_I2S_INT_OVERRUN); + + ret = snd_soc_register_dai(&s6000_i2s_dai); + if (ret) + goto err_release_dev; + + return 0; + +err_release_dev: + kfree(dev); +err_release_dma2: + if (dma2) + release_mem_region(dma2->start, dma2->end - dma2->start + 1); +err_release_dma1: + release_mem_region(dma1->start, dma1->end - dma1->start + 1); +err_release_sif: + release_mem_region(sifmem->start, (sifmem->end - sifmem->start) + 1); +err_release_map: + iounmap(mmio); +err_release_scb: + release_mem_region(scbmem->start, (scbmem->end - scbmem->start) + 1); +err_release_none: + return ret; +} + +static void __devexit s6000_i2s_remove(struct platform_device *pdev) +{ + struct s6000_i2s_dev *dev = s6000_i2s_dai.private_data; + struct resource *region; + void __iomem *mmio = dev->scbbase; + + snd_soc_unregister_dai(&s6000_i2s_dai); + + s6000_i2s_stop_channel(dev, 0); + s6000_i2s_stop_channel(dev, 1); + + s6_i2s_write_reg(dev, S6_I2S_INTERRUPT_ENABLE, 0); + s6000_i2s_dai.private_data = 0; + kfree(dev); + + region = platform_get_resource(pdev, IORESOURCE_DMA, 0); + release_mem_region(region->start, region->end - region->start + 1); + + region = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (region) + release_mem_region(region->start, + region->end - region->start + 1); + + region = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(region->start, (region->end - region->start) + 1); + + iounmap(mmio); + region = platform_get_resource(pdev, IORESOURCE_IO, 0); + release_mem_region(region->start, (region->end - region->start) + 1); +} + +static struct platform_driver s6000_i2s_driver = { + .probe = s6000_i2s_probe, + .remove = __devexit_p(s6000_i2s_remove), + .driver = { + .name = "s6000-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init s6000_i2s_init(void) +{ + return platform_driver_register(&s6000_i2s_driver); +} +module_init(s6000_i2s_init); + +static void __exit s6000_i2s_exit(void) +{ + platform_driver_unregister(&s6000_i2s_driver); +} +module_exit(s6000_i2s_exit); + +MODULE_AUTHOR("Daniel Gloeckner"); +MODULE_DESCRIPTION("Stretch s6000 family I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-i2s.h b/sound/soc/s6000/s6000-i2s.h new file mode 100644 index 0000000..2375fdf --- /dev/null +++ b/sound/soc/s6000/s6000-i2s.h @@ -0,0 +1,25 @@ +/* + * ALSA SoC I2S Audio Layer for the Stretch s6000 family + * + * Author: Daniel Gloeckner, <dg@emlix.com> + * Copyright: (C) 2009 emlix GmbH <info@emlix.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _S6000_I2S_H +#define _S6000_I2S_H + +extern struct snd_soc_dai s6000_i2s_dai; + +struct s6000_snd_platform_data { + int lines_in; + int lines_out; + int channel_in; + int channel_out; + int wide; + int same_rate; +}; +#endif diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c new file mode 100644 index 0000000..83b8028 --- /dev/null +++ b/sound/soc/s6000/s6000-pcm.c @@ -0,0 +1,497 @@ +/* + * ALSA PCM interface for the Stetch s6000 family + * + * Author: Daniel Gloeckner, <dg@emlix.com> + * Copyright: (C) 2009 emlix GmbH <info@emlix.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/interrupt.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/dma.h> +#include <variant/dmac.h> + +#include "s6000-pcm.h" + +#define S6_PCM_PREALLOCATE_SIZE (96 * 1024) +#define S6_PCM_PREALLOCATE_MAX (2048 * 1024) + +static struct snd_pcm_hardware s6000_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_JOINT_DUPLEX), + .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE), + .rates = (SNDRV_PCM_RATE_CONTINUOUS | SNDRV_PCM_RATE_5512 | \ + SNDRV_PCM_RATE_8000_192000), + .rate_min = 0, + .rate_max = 1562500, + .channels_min = 2, + .channels_max = 8, + .buffer_bytes_max = 0x7ffffff0, + .period_bytes_min = 16, + .period_bytes_max = 0xfffff0, + .periods_min = 2, + .periods_max = 1024, /* no limit */ + .fifo_size = 0, +}; + +struct s6000_runtime_data { + spinlock_t lock; + int period; /* current DMA period */ +}; + +static void s6000_pcm_enqueue_dma(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int channel; + unsigned int period_size; + unsigned int dma_offset; + dma_addr_t dma_pos; + dma_addr_t src, dst; + + period_size = snd_pcm_lib_period_bytes(substream); + dma_offset = prtd->period * period_size; + dma_pos = runtime->dma_addr + dma_offset; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + src = dma_pos; + dst = par->sif_out; + channel = par->dma_out; + } else { + src = par->sif_in; + dst = dma_pos; + channel = par->dma_in; + } + + if (!s6dmac_channel_enabled(DMA_MASK_DMAC(channel), + DMA_INDEX_CHNL(channel))) + return; + + if (s6dmac_fifo_full(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel))) { + printk(KERN_ERR "s6000-pcm: fifo full\n"); + return; + } + + BUG_ON(period_size & 15); + s6dmac_put_fifo(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel), + src, dst, period_size); + + prtd->period++; + if (unlikely(prtd->period >= runtime->periods)) + prtd->period = 0; +} + +static irqreturn_t s6000_pcm_irq(int irq, void *data) +{ + struct snd_pcm *pcm = data; + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + struct s6000_runtime_data *prtd; + unsigned int has_xrun; + int i, ret = IRQ_NONE; + u32 channel[2] = { + [SNDRV_PCM_STREAM_PLAYBACK] = params->dma_out, + [SNDRV_PCM_STREAM_CAPTURE] = params->dma_in + }; + + has_xrun = params->check_xrun(runtime->dai->cpu_dai); + + for (i = 0; i < ARRAY_SIZE(channel); ++i) { + struct snd_pcm_substream *substream = pcm->streams[i].substream; + unsigned int pending; + + if (!channel[i]) + continue; + + if (unlikely(has_xrun & (1 << i)) && + substream->runtime && + snd_pcm_running(substream)) { + dev_dbg(pcm->dev, "xrun\n"); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + ret = IRQ_HANDLED; + } + + pending = s6dmac_int_sources(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i])); + + if (pending & 1) { + ret = IRQ_HANDLED; + if (likely(substream->runtime && + snd_pcm_running(substream))) { + snd_pcm_period_elapsed(substream); + dev_dbg(pcm->dev, "period elapsed %x %x\n", + s6dmac_cur_src(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i])), + s6dmac_cur_dst(DMA_MASK_DMAC(channel[i]), + DMA_INDEX_CHNL(channel[i]))); + prtd = substream->runtime->private_data; + spin_lock(&prtd->lock); + s6000_pcm_enqueue_dma(substream); + spin_unlock(&prtd->lock); + } + } + + if (unlikely(pending & ~7)) { + if (pending & (1 << 3)) + printk(KERN_WARNING + "s6000-pcm: DMA %x Underflow\n", + channel[i]); + if (pending & (1 << 4)) + printk(KERN_WARNING + "s6000-pcm: DMA %x Overflow\n", + channel[i]); + if (pending & 0x1e0) + printk(KERN_WARNING + "s6000-pcm: DMA %x Master Error " + "(mask %x)\n", + channel[i], pending >> 5); + + } + } + + return ret; +} + +static int s6000_pcm_start(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + unsigned long flags; + int srcinc; + u32 dma; + + spin_lock_irqsave(&prtd->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + srcinc = 1; + dma = par->dma_out; + } else { + srcinc = 0; + dma = par->dma_in; + } + s6dmac_enable_chan(DMA_MASK_DMAC(dma), DMA_INDEX_CHNL(dma), + 1 /* priority 1 (0 is max) */, + 0 /* peripheral requests w/o xfer length mode */, + srcinc /* source address increment */, + srcinc^1 /* destination address increment */, + 0 /* chunksize 0 (skip impossible on this dma) */, + 0 /* source skip after chunk (impossible) */, + 0 /* destination skip after chunk (impossible) */, + 4 /* 16 byte burst size */, + -1 /* don't conserve bandwidth */, + 0 /* low watermark irq descriptor theshold */, + 0 /* disable hardware timestamps */, + 1 /* enable channel */); + + s6000_pcm_enqueue_dma(substream); + s6000_pcm_enqueue_dma(substream); + + spin_unlock_irqrestore(&prtd->lock, flags); + + return 0; +} + +static int s6000_pcm_stop(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + unsigned long flags; + u32 channel; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + channel = par->dma_out; + else + channel = par->dma_in; + + s6dmac_set_terminal_count(DMA_MASK_DMAC(channel), + DMA_INDEX_CHNL(channel), 0); + + spin_lock_irqsave(&prtd->lock, flags); + + s6dmac_disable_chan(DMA_MASK_DMAC(channel), DMA_INDEX_CHNL(channel)); + + spin_unlock_irqrestore(&prtd->lock, flags); + + return 0; +} + +static int s6000_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int ret; + + ret = par->trigger(substream, cmd, 0); + if (ret < 0) + return ret; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = s6000_pcm_start(substream); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = s6000_pcm_stop(substream); + break; + default: + ret = -EINVAL; + } + if (ret < 0) + return ret; + + return par->trigger(substream, cmd, 1); +} + +static int s6000_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct s6000_runtime_data *prtd = substream->runtime->private_data; + + prtd->period = 0; + + return 0; +} + +static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + unsigned long flags; + unsigned int offset; + dma_addr_t count; + + spin_lock_irqsave(&prtd->lock, flags); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + count = s6dmac_cur_src(DMA_MASK_DMAC(par->dma_out), + DMA_INDEX_CHNL(par->dma_out)); + else + count = s6dmac_cur_dst(DMA_MASK_DMAC(par->dma_in), + DMA_INDEX_CHNL(par->dma_in)); + + count -= runtime->dma_addr; + + spin_unlock_irqrestore(&prtd->lock, flags); + + offset = bytes_to_frames(runtime, count); + if (unlikely(offset >= runtime->buffer_size)) + offset = 0; + + return offset; +} + +static int s6000_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware); + + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 16); + if (ret < 0) + return ret; + ret = snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 16); + if (ret < 0) + return ret; + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + if (par->same_rate) { + int rate; + spin_lock(&par->lock); /* needed? */ + rate = par->rate; + spin_unlock(&par->lock); + if (rate != -1) { + ret = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_RATE, + rate, rate); + if (ret < 0) + return ret; + } + } + + prtd = kzalloc(sizeof(struct s6000_runtime_data), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int s6000_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct s6000_runtime_data *prtd = runtime->private_data; + + kfree(prtd); + + return 0; +} + +static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + int ret; + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) { + printk(KERN_WARNING "s6000-pcm: allocation of memory failed\n"); + return ret; + } + + if (par->same_rate) { + spin_lock(&par->lock); + if (par->rate == -1 || + !(par->in_use & ~(1 << substream->stream))) { + par->rate = params_rate(hw_params); + par->in_use |= 1 << substream->stream; + } else if (params_rate(hw_params) != par->rate) { + snd_pcm_lib_free_pages(substream); + par->in_use &= ~(1 << substream->stream); + ret = -EBUSY; + } + spin_unlock(&par->lock); + } + return ret; +} + +static int s6000_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *soc_runtime = substream->private_data; + struct s6000_pcm_dma_params *par = soc_runtime->dai->cpu_dai->dma_data; + + spin_lock(&par->lock); + par->in_use &= ~(1 << substream->stream); + if (!par->in_use) + par->rate = -1; + spin_unlock(&par->lock); + + return snd_pcm_lib_free_pages(substream); +} + +static struct snd_pcm_ops s6000_pcm_ops = { + .open = s6000_pcm_open, + .close = s6000_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = s6000_pcm_hw_params, + .hw_free = s6000_pcm_hw_free, + .trigger = s6000_pcm_trigger, + .prepare = s6000_pcm_prepare, + .pointer = s6000_pcm_pointer, +}; + +static void s6000_pcm_free(struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + + free_irq(params->irq, pcm); + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static u64 s6000_pcm_dmamask = DMA_32BIT_MASK; + +static int s6000_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct s6000_pcm_dma_params *params = runtime->dai->cpu_dai->dma_data; + int res; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &s6000_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_32BIT_MASK; + + if (params->dma_in) { + s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_in), + DMA_INDEX_CHNL(params->dma_in)); + s6dmac_int_sources(DMA_MASK_DMAC(params->dma_in), + DMA_INDEX_CHNL(params->dma_in)); + } + + if (params->dma_out) { + s6dmac_disable_chan(DMA_MASK_DMAC(params->dma_out), + DMA_INDEX_CHNL(params->dma_out)); + s6dmac_int_sources(DMA_MASK_DMAC(params->dma_out), + DMA_INDEX_CHNL(params->dma_out)); + } + + res = request_irq(params->irq, s6000_pcm_irq, IRQF_SHARED, + s6000_soc_platform.name, pcm); + if (res) { + printk(KERN_ERR "s6000-pcm couldn't get IRQ\n"); + return res; + } + + res = snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_DEV, + card->dev, + S6_PCM_PREALLOCATE_SIZE, + S6_PCM_PREALLOCATE_MAX); + if (res) + printk(KERN_WARNING "s6000-pcm: preallocation failed\n"); + + spin_lock_init(¶ms->lock); + params->in_use = 0; + params->rate = -1; + return 0; +} + +struct snd_soc_platform s6000_soc_platform = { + .name = "s6000-audio", + .pcm_ops = &s6000_pcm_ops, + .pcm_new = s6000_pcm_new, + .pcm_free = s6000_pcm_free, +}; +EXPORT_SYMBOL_GPL(s6000_soc_platform); + +static int __init s6000_pcm_init(void) +{ + return snd_soc_register_platform(&s6000_soc_platform); +} +module_init(s6000_pcm_init); + +static void __exit s6000_pcm_exit(void) +{ + snd_soc_unregister_platform(&s6000_soc_platform); +} +module_exit(s6000_pcm_exit); + +MODULE_AUTHOR("Daniel Gloeckner"); +MODULE_DESCRIPTION("Stretch s6000 family PCM DMA module"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/s6000/s6000-pcm.h b/sound/soc/s6000/s6000-pcm.h new file mode 100644 index 0000000..96f23f6 --- /dev/null +++ b/sound/soc/s6000/s6000-pcm.h @@ -0,0 +1,35 @@ +/* + * ALSA PCM interface for the Stretch s6000 family + * + * Author: Daniel Gloeckner, <dg@emlix.com> + * Copyright: (C) 2009 emlix GmbH <info@emlix.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _S6000_PCM_H +#define _S6000_PCM_H + +struct snd_soc_dai; +struct snd_pcm_substream; + +struct s6000_pcm_dma_params { + unsigned int (*check_xrun)(struct snd_soc_dai *cpu_dai); + int (*trigger)(struct snd_pcm_substream *substream, int cmd, int after); + dma_addr_t sif_in; + dma_addr_t sif_out; + u32 dma_in; + u32 dma_out; + int irq; + int same_rate; + + spinlock_t lock; + int in_use; + int rate; +}; + +extern struct snd_soc_platform s6000_soc_platform; + +#endif diff --git a/sound/soc/s6000/s6105-ipcam.c b/sound/soc/s6000/s6105-ipcam.c new file mode 100644 index 0000000..b5f95f9 --- /dev/null +++ b/sound/soc/s6000/s6105-ipcam.c @@ -0,0 +1,244 @@ +/* + * ASoC driver for Stretch s6105 IP camera platform + * + * Author: Daniel Gloeckner, <dg@emlix.com> + * Copyright: (C) 2009 emlix GmbH <info@emlix.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/timer.h> +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <variant/dmac.h> + +#include "../codecs/tlv320aic3x.h" +#include "s6000-pcm.h" +#include "s6000-i2s.h" + +#define S6105_CAM_CODEC_CLOCK 12288000 + +static int s6105_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret = 0; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF); + if (ret < 0) + return ret; + + /* set the codec system clock */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, S6105_CAM_CODEC_CLOCK, + SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops s6105_ops = { + .hw_params = s6105_hw_params, +}; + +/* s6105 machine dapm widgets */ +static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { + SND_SOC_DAPM_LINE("Audio Out Differential", NULL), + SND_SOC_DAPM_LINE("Audio Out Stereo", NULL), + SND_SOC_DAPM_LINE("Audio In", NULL), +}; + +/* s6105 machine audio_mapnections to the codec pins */ +static const struct snd_soc_dapm_route audio_map[] = { + /* Audio Out connected to HPLOUT, HPLCOM, HPROUT */ + {"Audio Out Differential", NULL, "HPLOUT"}, + {"Audio Out Differential", NULL, "HPLCOM"}, + {"Audio Out Stereo", NULL, "HPLOUT"}, + {"Audio Out Stereo", NULL, "HPROUT"}, + + /* Audio In connected to LINE1L, LINE1R */ + {"LINE1L", NULL, "Audio In"}, + {"LINE1R", NULL, "Audio In"}, +}; + +static int output_type_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + uinfo->count = 1; + uinfo->value.enumerated.items = 2; + if (uinfo->value.enumerated.item) { + uinfo->value.enumerated.item = 1; + strcpy(uinfo->value.enumerated.name, "HPLOUT/HPROUT"); + } else { + strcpy(uinfo->value.enumerated.name, "HPLOUT/HPLCOM"); + } + return 0; +} + +static int output_type_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = kcontrol->private_value; + return 0; +} + +static int output_type_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + unsigned int val = (ucontrol->value.enumerated.item[0] != 0); + char *differential = "Audio Out Differential"; + char *stereo = "Audio Out Stereo"; + + if (kcontrol->private_value == val) + return 0; + kcontrol->private_value = val; + snd_soc_dapm_disable_pin(codec, val ? differential : stereo); + snd_soc_dapm_sync(codec); + snd_soc_dapm_enable_pin(codec, val ? stereo : differential); + snd_soc_dapm_sync(codec); + + return 1; +} + +static const struct snd_kcontrol_new audio_out_mux = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Output Mux", + .index = 0, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = output_type_info, + .get = output_type_get, + .put = output_type_put, + .private_value = 1 /* default to stereo */ +}; + +/* Logic for a aic3x as connected on the s6105 ip camera ref design */ +static int s6105_aic3x_init(struct snd_soc_codec *codec) +{ + /* Add s6105 specific widgets */ + snd_soc_dapm_new_controls(codec, aic3x_dapm_widgets, + ARRAY_SIZE(aic3x_dapm_widgets)); + + /* Set up s6105 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* not present */ + snd_soc_dapm_nc_pin(codec, "MONO_LOUT"); + snd_soc_dapm_nc_pin(codec, "LINE2L"); + snd_soc_dapm_nc_pin(codec, "LINE2R"); + + /* not connected */ + snd_soc_dapm_nc_pin(codec, "MIC3L"); /* LINE2L on this chip */ + snd_soc_dapm_nc_pin(codec, "MIC3R"); /* LINE2R on this chip */ + snd_soc_dapm_nc_pin(codec, "LLOUT"); + snd_soc_dapm_nc_pin(codec, "RLOUT"); + snd_soc_dapm_nc_pin(codec, "HPRCOM"); + + /* always connected */ + snd_soc_dapm_enable_pin(codec, "Audio In"); + + /* must correspond to audio_out_mux.private_value initializer */ + snd_soc_dapm_disable_pin(codec, "Audio Out Differential"); + snd_soc_dapm_sync(codec); + snd_soc_dapm_enable_pin(codec, "Audio Out Stereo"); + + snd_soc_dapm_sync(codec); + + snd_ctl_add(codec->card, snd_ctl_new1(&audio_out_mux, codec)); + + return 0; +} + +/* s6105 digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link s6105_dai = { + .name = "TLV320AIC31", + .stream_name = "AIC31", + .cpu_dai = &s6000_i2s_dai, + .codec_dai = &aic3x_dai, + .init = s6105_aic3x_init, + .ops = &s6105_ops, +}; + +/* s6105 audio machine driver */ +static struct snd_soc_card snd_soc_card_s6105 = { + .name = "Stretch IP Camera", + .platform = &s6000_soc_platform, + .dai_link = &s6105_dai, + .num_links = 1, +}; + +/* s6105 audio private data */ +static struct aic3x_setup_data s6105_aic3x_setup = { + .i2c_bus = 0, + .i2c_address = 0x18, +}; + +/* s6105 audio subsystem */ +static struct snd_soc_device s6105_snd_devdata = { + .card = &snd_soc_card_s6105, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &s6105_aic3x_setup, +}; + +static struct s6000_snd_platform_data __initdata s6105_snd_data = { + .wide = 0, + .channel_in = 0, + .channel_out = 1, + .lines_in = 1, + .lines_out = 1, + .same_rate = 1, +}; + +static struct platform_device *s6105_snd_device; + +static int __init s6105_init(void) +{ + int ret; + + s6105_snd_device = platform_device_alloc("soc-audio", -1); + if (!s6105_snd_device) + return -ENOMEM; + + platform_set_drvdata(s6105_snd_device, &s6105_snd_devdata); + s6105_snd_devdata.dev = &s6105_snd_device->dev; + platform_device_add_data(s6105_snd_device, &s6105_snd_data, + sizeof(s6105_snd_data)); + + ret = platform_device_add(s6105_snd_device); + if (ret) + platform_device_put(s6105_snd_device); + + return ret; +} + +static void __exit s6105_exit(void) +{ + platform_device_unregister(s6105_snd_device); +} + +module_init(s6105_init); +module_exit(s6105_exit); + +MODULE_AUTHOR("Daniel Gloeckner"); +MODULE_DESCRIPTION("Stretch s6105 IP camera ASoC driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 99712f6..af11791 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -113,6 +113,35 @@ static int soc_ac97_dev_register(struct snd_soc_codec *codec) } #endif +static int soc_pcm_apply_symmetry(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_card *card = socdev->card; + struct snd_soc_dai_link *machine = rtd->dai; + struct snd_soc_dai *cpu_dai = machine->cpu_dai; + struct snd_soc_dai *codec_dai = machine->codec_dai; + int ret; + + if (codec_dai->symmetric_rates || cpu_dai->symmetric_rates || + machine->symmetric_rates) { + dev_dbg(card->dev, "Symmetry forces %dHz rate\n", + machine->rate); + + ret = snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + machine->rate, + machine->rate); + if (ret < 0) { + dev_err(card->dev, + "Unable to apply rate symmetry constraint: %d\n", ret); + return ret; + } + } + + return 0; +} + /* * Called by ALSA when a PCM substream is opened, the runtime->hw record is * then initialized and any private data can be allocated. This also calls @@ -221,6 +250,13 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) goto machine_err; } + /* Symmetry only applies if we've already got an active stream. */ + if (cpu_dai->active || codec_dai->active) { + ret = soc_pcm_apply_symmetry(substream); + if (ret != 0) + goto machine_err; + } + pr_debug("asoc: %s <-> %s info:\n", codec_dai->name, cpu_dai->name); pr_debug("asoc: rate mask 0x%x\n", runtime->hw.rates); pr_debug("asoc: min ch %d max ch %d\n", runtime->hw.channels_min, @@ -521,6 +557,8 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, } } + machine->rate = params_rate(params); + out: mutex_unlock(&pcm_mutex); return ret; @@ -1741,7 +1779,7 @@ int snd_soc_info_volsw_ext(struct snd_kcontrol *kcontrol, { int max = kcontrol->private_value; - if (max == 1) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -1771,7 +1809,7 @@ int snd_soc_info_volsw(struct snd_kcontrol *kcontrol, unsigned int shift = mc->shift; unsigned int rshift = mc->rshift; - if (max == 1) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -1878,7 +1916,7 @@ int snd_soc_info_volsw_2r(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; int max = mc->max; - if (max == 1) + if (max == 1 && !strstr(kcontrol->id.name, " Volume")) uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; else uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -2062,7 +2100,7 @@ EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->ops->set_sysclk) + if (dai->ops && dai->ops->set_sysclk) return dai->ops->set_sysclk(dai, clk_id, freq, dir); else return -EINVAL; @@ -2082,7 +2120,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_sysclk); int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->ops->set_clkdiv) + if (dai->ops && dai->ops->set_clkdiv) return dai->ops->set_clkdiv(dai, div_id, div); else return -EINVAL; @@ -2101,7 +2139,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_clkdiv); int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { - if (dai->ops->set_pll) + if (dai->ops && dai->ops->set_pll) return dai->ops->set_pll(dai, pll_id, freq_in, freq_out); else return -EINVAL; @@ -2117,7 +2155,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_pll); */ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { - if (dai->ops->set_fmt) + if (dai->ops && dai->ops->set_fmt) return dai->ops->set_fmt(dai, fmt); else return -EINVAL; @@ -2136,7 +2174,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int mask, int slots) { - if (dai->ops->set_sysclk) + if (dai->ops && dai->ops->set_tdm_slot) return dai->ops->set_tdm_slot(dai, mask, slots); else return -EINVAL; @@ -2152,7 +2190,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->ops->set_sysclk) + if (dai->ops && dai->ops->set_tristate) return dai->ops->set_tristate(dai, tristate); else return -EINVAL; @@ -2168,7 +2206,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_tristate); */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute) { - if (dai->ops->digital_mute) + if (dai->ops && dai->ops->digital_mute) return dai->ops->digital_mute(dai, mute); else return -EINVAL; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 735903a..a6d7337 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -357,8 +357,9 @@ static int dapm_new_mixer(struct snd_soc_codec *codec, path->long_name); ret = snd_ctl_add(codec->card, path->kcontrol); if (ret < 0) { - printk(KERN_ERR "asoc: failed to add dapm kcontrol %s\n", - path->long_name); + printk(KERN_ERR "asoc: failed to add dapm kcontrol %s: %d\n", + path->long_name, + ret); kfree(path->long_name); path->long_name = NULL; return ret; @@ -521,6 +522,65 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_reg_event); +/* Standard power change method, used to apply power changes to most + * widgets. + */ +static int dapm_generic_apply_power(struct snd_soc_dapm_widget *w) +{ + int ret; + + /* call any power change event handlers */ + if (w->event) + pr_debug("power %s event for %s flags %x\n", + w->power ? "on" : "off", + w->name, w->event_flags); + + /* power up pre event */ + if (w->power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); + if (ret < 0) + return ret; + } + + /* power down pre event */ + if (!w->power && w->event && + (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); + if (ret < 0) + return ret; + } + + /* Lower PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && !w->power) + dapm_set_pga(w, w->power); + + dapm_update_bits(w); + + /* Raise PGA volume to reduce pops */ + if (w->id == snd_soc_dapm_pga && w->power) + dapm_set_pga(w, w->power); + + /* power up post event */ + if (w->power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMU)) { + ret = w->event(w, + NULL, SND_SOC_DAPM_POST_PMU); + if (ret < 0) + return ret; + } + + /* power down post event */ + if (!w->power && w->event && + (w->event_flags & SND_SOC_DAPM_POST_PMD)) { + ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); + if (ret < 0) + return ret; + } + + return 0; +} + /* * Scan a single DAPM widget for a complete audio path and update the * power status appropriately. @@ -538,18 +598,22 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, if (w->id == snd_soc_dapm_adc && w->active) { in = is_connected_input_ep(w); dapm_clear_walk(w->codec); - w->power = (in != 0) ? 1 : 0; - dapm_update_bits(w); - return 0; + power = (in != 0) ? 1 : 0; + if (power == w->power) + return 0; + w->power = power; + return dapm_generic_apply_power(w); } /* active DAC */ if (w->id == snd_soc_dapm_dac && w->active) { out = is_connected_output_ep(w); dapm_clear_walk(w->codec); - w->power = (out != 0) ? 1 : 0; - dapm_update_bits(w); - return 0; + power = (out != 0) ? 1 : 0; + if (power == w->power) + return 0; + w->power = power; + return dapm_generic_apply_power(w); } /* pre and post event widgets */ @@ -600,56 +664,7 @@ static int dapm_power_widget(struct snd_soc_codec *codec, int event, if (!power_change) return 0; - /* call any power change event handlers */ - if (w->event) - pr_debug("power %s event for %s flags %x\n", - w->power ? "on" : "off", - w->name, w->event_flags); - - /* power up pre event */ - if (power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMU)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMU); - if (ret < 0) - return ret; - } - - /* power down pre event */ - if (!power && w->event && - (w->event_flags & SND_SOC_DAPM_PRE_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_PRE_PMD); - if (ret < 0) - return ret; - } - - /* Lower PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && !power) - dapm_set_pga(w, power); - - dapm_update_bits(w); - - /* Raise PGA volume to reduce pops */ - if (w->id == snd_soc_dapm_pga && power) - dapm_set_pga(w, power); - - /* power up post event */ - if (power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMU)) { - ret = w->event(w, - NULL, SND_SOC_DAPM_POST_PMU); - if (ret < 0) - return ret; - } - - /* power down post event */ - if (!power && w->event && - (w->event_flags & SND_SOC_DAPM_POST_PMD)) { - ret = w->event(w, NULL, SND_SOC_DAPM_POST_PMD); - if (ret < 0) - return ret; - } - - return 0; + return dapm_generic_apply_power(w); } /* diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index ba38912..574af56 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -954,7 +954,8 @@ static int __devinit snd_amd7930_create(struct snd_card *card, amd->regs = of_ioremap(&op->resource[0], 0, resource_size(&op->resource[0]), "amd7930"); if (!amd->regs) { - snd_printk("amd7930-%d: Unable to map chip registers.\n", dev); + snd_printk(KERN_ERR + "amd7930-%d: Unable to map chip registers.\n", dev); return -EIO; } @@ -962,7 +963,7 @@ static int __devinit snd_amd7930_create(struct snd_card *card, if (request_irq(irq, snd_amd7930_interrupt, IRQF_DISABLED | IRQF_SHARED, "amd7930", amd)) { - snd_printk("amd7930-%d: Unable to grab IRQ %d\n", + snd_printk(KERN_ERR "amd7930-%d: Unable to grab IRQ %d\n", dev, irq); snd_amd7930_free(amd); return -EBUSY; diff --git a/sound/synth/emux/emux_hwdep.c b/sound/synth/emux/emux_hwdep.c index 0a53914..ff0b2a8 100644 --- a/sound/synth/emux/emux_hwdep.c +++ b/sound/synth/emux/emux_hwdep.c @@ -24,25 +24,6 @@ #include <asm/uaccess.h> #include "emux_voice.h" -/* - * open the hwdep device - */ -static int -snd_emux_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - - -/* - * close the device - */ -static int -snd_emux_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - #define TMP_CLIENT_ID 0x1001 @@ -146,8 +127,6 @@ snd_emux_init_hwdep(struct snd_emux *emu) emu->hwdep = hw; strcpy(hw->name, SNDRV_EMUX_HWDEP_NAME); hw->iface = SNDRV_HWDEP_IFACE_EMUX_WAVETABLE; - hw->ops.open = snd_emux_hwdep_open; - hw->ops.release = snd_emux_hwdep_release; hw->ops.ioctl = snd_emux_hwdep_ioctl; hw->exclusive = 1; hw->private_data = emu; diff --git a/sound/synth/emux/emux_oss.c b/sound/synth/emux/emux_oss.c index 5c47b6c..87e4220 100644 --- a/sound/synth/emux/emux_oss.c +++ b/sound/synth/emux/emux_oss.c @@ -132,7 +132,7 @@ snd_emux_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure) p = snd_emux_create_port(emu, tmpname, 32, 1, &callback); if (p == NULL) { - snd_printk("can't create port\n"); + snd_printk(KERN_ERR "can't create port\n"); snd_emux_dec_count(emu); mutex_unlock(&emu->register_mutex); return -ENOMEM; diff --git a/sound/synth/emux/emux_seq.c b/sound/synth/emux/emux_seq.c index 335aa2c..ca5f7ef 100644 --- a/sound/synth/emux/emux_seq.c +++ b/sound/synth/emux/emux_seq.c @@ -74,15 +74,15 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index) emu->client = snd_seq_create_kernel_client(card, index, "%s WaveTable", emu->name); if (emu->client < 0) { - snd_printk("can't create client\n"); + snd_printk(KERN_ERR "can't create client\n"); return -ENODEV; } if (emu->num_ports < 0) { - snd_printk("seqports must be greater than zero\n"); + snd_printk(KERN_WARNING "seqports must be greater than zero\n"); emu->num_ports = 1; } else if (emu->num_ports >= SNDRV_EMUX_MAX_PORTS) { - snd_printk("too many ports." + snd_printk(KERN_WARNING "too many ports." "limited max. ports %d\n", SNDRV_EMUX_MAX_PORTS); emu->num_ports = SNDRV_EMUX_MAX_PORTS; } @@ -100,7 +100,7 @@ snd_emux_init_seq(struct snd_emux *emu, struct snd_card *card, int index) p = snd_emux_create_port(emu, tmpname, MIDI_CHANNELS, 0, &pinfo); if (p == NULL) { - snd_printk("can't create port\n"); + snd_printk(KERN_ERR "can't create port\n"); return -ENOMEM; } @@ -147,12 +147,12 @@ snd_emux_create_port(struct snd_emux *emu, char *name, /* Allocate structures for this channel */ if ((p = kzalloc(sizeof(*p), GFP_KERNEL)) == NULL) { - snd_printk("no memory\n"); + snd_printk(KERN_ERR "no memory\n"); return NULL; } p->chset.channels = kcalloc(max_channels, sizeof(struct snd_midi_channel), GFP_KERNEL); if (p->chset.channels == NULL) { - snd_printk("no memory\n"); + snd_printk(KERN_ERR "no memory\n"); kfree(p); return NULL; } @@ -376,12 +376,12 @@ int snd_emux_init_virmidi(struct snd_emux *emu, struct snd_card *card) goto __error; } emu->vmidi[i] = rmidi; - //snd_printk("virmidi %d ok\n", i); + /* snd_printk(KERN_DEBUG "virmidi %d ok\n", i); */ } return 0; __error: - //snd_printk("error init..\n"); + /* snd_printk(KERN_DEBUG "error init..\n"); */ snd_emux_delete_virmidi(emu); return -ENOMEM; } diff --git a/sound/synth/emux/emux_synth.c b/sound/synth/emux/emux_synth.c index 2cc6f6f..3e921b3 100644 --- a/sound/synth/emux/emux_synth.c +++ b/sound/synth/emux/emux_synth.c @@ -956,7 +956,8 @@ void snd_emux_lock_voice(struct snd_emux *emu, int voice) if (emu->voices[voice].state == SNDRV_EMUX_ST_OFF) emu->voices[voice].state = SNDRV_EMUX_ST_LOCKED; else - snd_printk("invalid voice for lock %d (state = %x)\n", + snd_printk(KERN_WARNING + "invalid voice for lock %d (state = %x)\n", voice, emu->voices[voice].state); spin_unlock_irqrestore(&emu->voice_lock, flags); } @@ -973,7 +974,8 @@ void snd_emux_unlock_voice(struct snd_emux *emu, int voice) if (emu->voices[voice].state == SNDRV_EMUX_ST_LOCKED) emu->voices[voice].state = SNDRV_EMUX_ST_OFF; else - snd_printk("invalid voice for unlock %d (state = %x)\n", + snd_printk(KERN_WARNING + "invalid voice for unlock %d (state = %x)\n", voice, emu->voices[voice].state); spin_unlock_irqrestore(&emu->voice_lock, flags); } diff --git a/sound/synth/emux/soundfont.c b/sound/synth/emux/soundfont.c index 36d53bd..63c8f45 100644 --- a/sound/synth/emux/soundfont.c +++ b/sound/synth/emux/soundfont.c @@ -133,7 +133,7 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, int rc; if (count < (long)sizeof(patch)) { - snd_printk("patch record too small %ld\n", count); + snd_printk(KERN_ERR "patch record too small %ld\n", count); return -EINVAL; } if (copy_from_user(&patch, data, sizeof(patch))) @@ -143,15 +143,16 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, data += sizeof(patch); if (patch.key != SNDRV_OSS_SOUNDFONT_PATCH) { - snd_printk("'The wrong kind of patch' %x\n", patch.key); + snd_printk(KERN_ERR "The wrong kind of patch %x\n", patch.key); return -EINVAL; } if (count < patch.len) { - snd_printk("Patch too short %ld, need %d\n", count, patch.len); + snd_printk(KERN_ERR "Patch too short %ld, need %d\n", + count, patch.len); return -EINVAL; } if (patch.len < 0) { - snd_printk("poor length %d\n", patch.len); + snd_printk(KERN_ERR "poor length %d\n", patch.len); return -EINVAL; } @@ -195,7 +196,8 @@ snd_soundfont_load(struct snd_sf_list *sflist, const void __user *data, case SNDRV_SFNT_REMOVE_INFO: /* patch must be opened */ if (!sflist->currsf) { - snd_printk("soundfont: remove_info: patch not opened\n"); + snd_printk(KERN_ERR "soundfont: remove_info: " + "patch not opened\n"); rc = -EINVAL; } else { int bank, instr; @@ -531,7 +533,7 @@ load_info(struct snd_sf_list *sflist, const void __user *data, long count) return -EINVAL; if (count < (long)sizeof(hdr)) { - printk("Soundfont error: invalid patch zone length\n"); + printk(KERN_ERR "Soundfont error: invalid patch zone length\n"); return -EINVAL; } if (copy_from_user((char*)&hdr, data, sizeof(hdr))) @@ -541,12 +543,14 @@ load_info(struct snd_sf_list *sflist, const void __user *data, long count) count -= sizeof(hdr); if (hdr.nvoices <= 0 || hdr.nvoices >= 100) { - printk("Soundfont error: Illegal voice number %d\n", hdr.nvoices); + printk(KERN_ERR "Soundfont error: Illegal voice number %d\n", + hdr.nvoices); return -EINVAL; } if (count < (long)sizeof(struct soundfont_voice_info) * hdr.nvoices) { - printk("Soundfont Error: patch length(%ld) is smaller than nvoices(%d)\n", + printk(KERN_ERR "Soundfont Error: " + "patch length(%ld) is smaller than nvoices(%d)\n", count, hdr.nvoices); return -EINVAL; } @@ -952,7 +956,7 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, int rc; if (count < (long)sizeof(patch)) { - snd_printk("patch record too small %ld\n", count); + snd_printk(KERN_ERR "patch record too small %ld\n", count); return -EINVAL; } if (copy_from_user(&patch, data, sizeof(patch))) @@ -1034,7 +1038,8 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, /* panning position; -128 - 127 => 0-127 */ zone->v.pan = (patch.panning + 128) / 2; #if 0 - snd_printk("gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n", + snd_printk(KERN_DEBUG + "gus: basefrq=%d (ofs=%d) root=%d,tune=%d, range:%d-%d\n", (int)patch.base_freq, zone->v.rate_offset, zone->v.root, zone->v.tune, zone->v.low, zone->v.high); #endif @@ -1068,7 +1073,8 @@ load_guspatch(struct snd_sf_list *sflist, const char __user *data, zone->v.parm.volrelease = 0x8000 | snd_sf_calc_parm_decay(release); zone->v.attenuation = calc_gus_attenuation(patch.env_offset[0]); #if 0 - snd_printk("gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n", + snd_printk(KERN_DEBUG + "gus: atkhld=%x, dcysus=%x, volrel=%x, att=%d\n", zone->v.parm.volatkhld, zone->v.parm.voldcysus, zone->v.parm.volrelease, diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 4f0eac9..523aec1 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -48,7 +48,10 @@ config SND_USB_CAIAQ * Native Instruments Kore Controller * Native Instruments Kore Controller 2 * Native Instruments Audio Kontrol 1 + * Native Instruments Audio 4 DJ * Native Instruments Audio 8 DJ + * Native Instruments Guitar Rig Session I/O + * Native Instruments Guitar Rig mobile To compile this driver as a module, choose M here: the module will be called snd-usb-caiaq. diff --git a/sound/usb/caiaq/caiaq-audio.c b/sound/usb/caiaq/caiaq-audio.c index b3a6033..08d51e0 100644 --- a/sound/usb/caiaq/caiaq-audio.c +++ b/sound/usb/caiaq/caiaq-audio.c @@ -114,6 +114,7 @@ static int stream_start(struct snd_usb_caiaqdev *dev) dev->output_panic = 0; dev->first_packet = 1; dev->streaming = 1; + dev->warned = 0; for (i = 0; i < N_URBS; i++) { ret = usb_submit_urb(dev->data_urbs_in[i], GFP_ATOMIC); @@ -376,6 +377,9 @@ static void read_in_urb_mode2(struct snd_usb_caiaqdev *dev, for (stream = 0; stream < dev->n_streams; stream++, i++) { sub = dev->sub_capture[stream]; + if (dev->input_panic) + usb_buf[i] = 0; + if (sub) { struct snd_pcm_runtime *rt = sub->runtime; char *audio_buf = rt->dma_area; @@ -397,6 +401,9 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, if (!dev->streaming) return; + if (iso->actual_length < dev->bpp) + return; + switch (dev->spec.data_alignment) { case 0: read_in_urb_mode0(dev, urb, iso); @@ -406,10 +413,11 @@ static void read_in_urb(struct snd_usb_caiaqdev *dev, break; } - if (dev->input_panic || dev->output_panic) { + if ((dev->input_panic || dev->output_panic) && !dev->warned) { debug("streaming error detected %s %s\n", dev->input_panic ? "(input)" : "", dev->output_panic ? "(output)" : ""); + dev->warned = 1; } } @@ -638,9 +646,10 @@ int snd_usb_caiaq_audio_init(struct snd_usb_caiaqdev *dev) case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AK1): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_RIGKONTROL3): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_SESSIONIO): - dev->samplerates |= SNDRV_PCM_RATE_88200; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_GUITARRIGMOBILE): dev->samplerates |= SNDRV_PCM_RATE_192000; - break; + /* fall thru */ + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ): dev->samplerates |= SNDRV_PCM_RATE_88200; break; diff --git a/sound/usb/caiaq/caiaq-control.c b/sound/usb/caiaq/caiaq-control.c index ccd763d..e92c2bb 100644 --- a/sound/usb/caiaq/caiaq-control.c +++ b/sound/usb/caiaq/caiaq-control.c @@ -39,12 +39,12 @@ static int control_info(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; int is_intval = pos & CNT_INTVAL; + unsigned int id = dev->chip.usb_id; uinfo->count = 1; pos &= ~CNT_INTVAL; - if (dev->chip.usb_id == - USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ) + if (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO8DJ) && (pos == 0)) { /* current input mode of A8DJ */ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; @@ -53,6 +53,15 @@ static int control_info(struct snd_kcontrol *kcontrol, return 0; } + if (id == USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ) + && (pos == 0)) { + /* current input mode of A4DJ */ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + return 0; + } + if (is_intval) { uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->value.integer.min = 0; @@ -73,6 +82,14 @@ static int control_get(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; + if (dev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) { + /* A4DJ has only one control */ + /* do not expose hardware input mode 0 */ + ucontrol->value.integer.value[0] = dev->control_state[0] - 1; + return 0; + } + if (pos & CNT_INTVAL) ucontrol->value.integer.value[0] = dev->control_state[pos & ~CNT_INTVAL]; @@ -90,10 +107,20 @@ static int control_put(struct snd_kcontrol *kcontrol, struct snd_usb_caiaqdev *dev = caiaqdev(chip->card); int pos = kcontrol->private_value; + if (dev->chip.usb_id == + USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ)) { + /* A4DJ has only one control */ + /* do not expose hardware input mode 0 */ + dev->control_state[0] = ucontrol->value.integer.value[0] + 1; + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, + dev->control_state, sizeof(dev->control_state)); + return 1; + } + if (pos & CNT_INTVAL) { dev->control_state[pos & ~CNT_INTVAL] = ucontrol->value.integer.value[0]; - snd_usb_caiaq_send_command(dev, EP1_CMD_DIMM_LEDS, + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, dev->control_state, sizeof(dev->control_state)); } else { if (ucontrol->value.integer.value[0]) @@ -243,10 +270,13 @@ static struct caiaq_controller a8dj_controller[] = { { "GND lift for TC Vinyl mode", 24 + 0 }, { "GND lift for TC CD/Line mode", 24 + 1 }, { "GND lift for phono mode", 24 + 2 }, - { "GND lift for TC Vinyl mode", 24 + 3 }, { "Software lock", 40 } }; +static struct caiaq_controller a4dj_controller[] = { + { "Current input mode", 0 | CNT_INTVAL } +}; + static int __devinit add_controls(struct caiaq_controller *c, int num, struct snd_usb_caiaqdev *dev) { @@ -295,6 +325,10 @@ int __devinit snd_usb_caiaq_control_init(struct snd_usb_caiaqdev *dev) ret = add_controls(a8dj_controller, ARRAY_SIZE(a8dj_controller), dev); break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): + ret = add_controls(a4dj_controller, + ARRAY_SIZE(a4dj_controller), dev); + break; } return ret; diff --git a/sound/usb/caiaq/caiaq-device.c b/sound/usb/caiaq/caiaq-device.c index 09aed23..cf573a9 100644 --- a/sound/usb/caiaq/caiaq-device.c +++ b/sound/usb/caiaq/caiaq-device.c @@ -42,15 +42,17 @@ #endif MODULE_AUTHOR("Daniel Mack <daniel@caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.10"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.13"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," "{Native Instruments, Kore Controller}," "{Native Instruments, Kore Controller 2}," "{Native Instruments, Audio Kontrol 1}," + "{Native Instruments, Audio 4 DJ}," "{Native Instruments, Audio 8 DJ}," - "{Native Instruments, Session I/O}}"); + "{Native Instruments, Session I/O}," + "{Native Instruments, GuitarRig mobile}"); static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ static char* id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for this card */ @@ -116,6 +118,16 @@ static struct usb_device_id snd_usb_id_table[] = { .idVendor = USB_VID_NATIVEINSTRUMENTS, .idProduct = USB_PID_SESSIONIO }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_GUITARRIGMOBILE + }, + { + .match_flags = USB_DEVICE_ID_MATCH_DEVICE, + .idVendor = USB_VID_NATIVEINSTRUMENTS, + .idProduct = USB_PID_AUDIO4DJ + }, { /* terminator */ } }; @@ -239,6 +251,8 @@ int snd_usb_caiaq_set_audio_params (struct snd_usb_caiaqdev *dev, if (dev->audio_parm_answer != 1) debug("unable to set the device's audio params\n"); + else + dev->bpp = bpp; return dev->audio_parm_answer == 1 ? 0 : -EINVAL; } @@ -300,6 +314,12 @@ static void __devinit setup_card(struct snd_usb_caiaqdev *dev) } break; + case USB_ID(USB_VID_NATIVEINSTRUMENTS, USB_PID_AUDIO4DJ): + /* Audio 4 DJ - default input mode to phono */ + dev->control_state[0] = 2; + snd_usb_caiaq_send_command(dev, EP1_CMD_WRITE_IO, + dev->control_state, 1); + break; } if (dev->spec.num_analog_audio_out + diff --git a/sound/usb/caiaq/caiaq-device.h b/sound/usb/caiaq/caiaq-device.h index ab56e73..4cce1ad 100644 --- a/sound/usb/caiaq/caiaq-device.h +++ b/sound/usb/caiaq/caiaq-device.h @@ -10,8 +10,10 @@ #define USB_PID_KORECONTROLLER 0x4711 #define USB_PID_KORECONTROLLER2 0x4712 #define USB_PID_AK1 0x0815 +#define USB_PID_AUDIO4DJ 0x0839 #define USB_PID_AUDIO8DJ 0x1978 #define USB_PID_SESSIONIO 0x1915 +#define USB_PID_GUITARRIGMOBILE 0x0d8d #define EP1_BUFSIZE 64 #define CAIAQ_USB_STR_LEN 0xff @@ -87,9 +89,9 @@ struct snd_usb_caiaqdev { int audio_out_buf_pos[MAX_STREAMS]; int period_in_count[MAX_STREAMS]; int period_out_count[MAX_STREAMS]; - int input_panic, output_panic; + int input_panic, output_panic, warned; char *audio_in_buf, *audio_out_buf; - unsigned int samplerates; + unsigned int samplerates, bpp; struct snd_pcm_substream *sub_playback[MAX_STREAMS]; struct snd_pcm_substream *sub_capture[MAX_STREAMS]; diff --git a/sound/usb/usbaudio.c b/sound/usb/usbaudio.c index eec32e1..823296d 100644 --- a/sound/usb/usbaudio.c +++ b/sound/usb/usbaudio.c @@ -107,7 +107,7 @@ MODULE_PARM_DESC(ignore_ctl_error, #define MAX_PACKS_HS (MAX_PACKS * 8) /* in high speed mode */ #define MAX_URBS 8 #define SYNC_URBS 4 /* always four urbs for sync */ -#define MIN_PACKS_URB 1 /* minimum 1 packet per urb */ +#define MAX_QUEUE 24 /* try not to exceed this queue length, in ms */ struct audioformat { struct list_head list; @@ -121,6 +121,7 @@ struct audioformat { unsigned char attributes; /* corresponding attributes of cs endpoint */ unsigned char endpoint; /* endpoint */ unsigned char ep_attr; /* endpoint attributes */ + unsigned char datainterval; /* log_2 of data packet interval */ unsigned int maxpacksize; /* max. packet size */ unsigned int rates; /* rate bitmasks */ unsigned int rate_min, rate_max; /* min/max rates */ @@ -170,7 +171,6 @@ struct snd_usb_substream { unsigned int curframesize; /* current packet size in frames (for capture) */ unsigned int fill_max: 1; /* fill max packet size always */ unsigned int fmt_type; /* USB audio format type (1-3) */ - unsigned int packs_per_ms; /* packets per millisecond (for playback) */ unsigned int running: 1; /* running status */ @@ -525,7 +525,7 @@ static int snd_usb_audio_next_packet_size(struct snd_usb_substream *subs) /* * Prepare urb for streaming before playback starts or when paused. * - * We don't have any data, so we send a frame of silence. + * We don't have any data, so we send silence. */ static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, struct snd_pcm_runtime *runtime, @@ -537,13 +537,13 @@ static int prepare_nodata_playback_urb(struct snd_usb_substream *subs, offs = 0; urb->dev = ctx->subs->dev; - urb->number_of_packets = subs->packs_per_ms; - for (i = 0; i < subs->packs_per_ms; ++i) { + for (i = 0; i < ctx->packets; ++i) { counts = snd_usb_audio_next_packet_size(subs); urb->iso_frame_desc[i].offset = offs * stride; urb->iso_frame_desc[i].length = counts * stride; offs += counts; } + urb->number_of_packets = ctx->packets; urb->transfer_buffer_length = offs * stride; memset(urb->transfer_buffer, subs->cur_audiofmt->format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0, @@ -607,9 +607,7 @@ static int prepare_playback_urb(struct snd_usb_substream *subs, break; } } - /* finish at the frame boundary at/after the period boundary */ - if (period_elapsed && - (i & (subs->packs_per_ms - 1)) == subs->packs_per_ms - 1) + if (period_elapsed) /* finish at the period boundary */ break; } if (subs->hwptr_done + offs > runtime->buffer_size) { @@ -1034,9 +1032,9 @@ static void release_substream_urbs(struct snd_usb_substream *subs, int force) static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int period_bytes, unsigned int rate, unsigned int frame_bits) { - unsigned int maxsize, n, i; + unsigned int maxsize, i; int is_playback = subs->direction == SNDRV_PCM_STREAM_PLAYBACK; - unsigned int npacks[MAX_URBS], urb_packs, total_packs, packs_per_ms; + unsigned int urb_packs, total_packs, packs_per_ms; /* calculate the frequency in 16.16 format */ if (snd_usb_get_speed(subs->dev) == USB_SPEED_FULL) @@ -1067,11 +1065,9 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri packs_per_ms = 8 >> subs->datainterval; else packs_per_ms = 1; - subs->packs_per_ms = packs_per_ms; if (is_playback) { - urb_packs = nrpacks; - urb_packs = max(urb_packs, (unsigned int)MIN_PACKS_URB); + urb_packs = max(nrpacks, 1); urb_packs = min(urb_packs, (unsigned int)MAX_PACKS); } else urb_packs = 1; @@ -1079,7 +1075,7 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri /* decide how many packets to be used */ if (is_playback) { - unsigned int minsize; + unsigned int minsize, maxpacks; /* determine how small a packet can be */ minsize = (subs->freqn >> (16 - subs->datainterval)) * (frame_bits >> 3); @@ -1088,13 +1084,17 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri minsize -= minsize >> 3; minsize = max(minsize, 1u); total_packs = (period_bytes + minsize - 1) / minsize; - /* round up to multiple of packs_per_ms */ - total_packs = (total_packs + packs_per_ms - 1) - & ~(packs_per_ms - 1); /* we need at least two URBs for queueing */ - if (total_packs < 2 * MIN_PACKS_URB * packs_per_ms) - total_packs = 2 * MIN_PACKS_URB * packs_per_ms; + if (total_packs < 2) { + total_packs = 2; + } else { + /* and we don't want too long a queue either */ + maxpacks = max(MAX_QUEUE * packs_per_ms, urb_packs * 2); + total_packs = min(total_packs, maxpacks); + } } else { + while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) + urb_packs >>= 1; total_packs = MAX_URBS * urb_packs; } subs->nurbs = (total_packs + urb_packs - 1) / urb_packs; @@ -1102,31 +1102,11 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri /* too much... */ subs->nurbs = MAX_URBS; total_packs = MAX_URBS * urb_packs; - } - n = total_packs; - for (i = 0; i < subs->nurbs; i++) { - npacks[i] = n > urb_packs ? urb_packs : n; - n -= urb_packs; - } - if (subs->nurbs <= 1) { + } else if (subs->nurbs < 2) { /* too little - we need at least two packets * to ensure contiguous playback/capture */ subs->nurbs = 2; - npacks[0] = (total_packs + 1) / 2; - npacks[1] = total_packs - npacks[0]; - } else if (npacks[subs->nurbs-1] < MIN_PACKS_URB * packs_per_ms) { - /* the last packet is too small.. */ - if (subs->nurbs > 2) { - /* merge to the first one */ - npacks[0] += npacks[subs->nurbs - 1]; - subs->nurbs--; - } else { - /* divide to two */ - subs->nurbs = 2; - npacks[0] = (total_packs + 1) / 2; - npacks[1] = total_packs - npacks[0]; - } } /* allocate and initialize data urbs */ @@ -1134,7 +1114,8 @@ static int init_substream_urbs(struct snd_usb_substream *subs, unsigned int peri struct snd_urb_ctx *u = &subs->dataurb[i]; u->index = i; u->subs = subs; - u->packets = npacks[i]; + u->packets = (i + 1) * total_packs / subs->nurbs + - i * total_packs / subs->nurbs; u->buffer_size = maxsize * u->packets; if (subs->fmt_type == USB_FORMAT_TYPE_II) u->packets++; /* for transfer delimiter */ @@ -1292,14 +1273,14 @@ static int init_usb_sample_rate(struct usb_device *dev, int iface, if ((err = snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), SET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_OUT, SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { - snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep 0x%x\n", + snd_printk(KERN_ERR "%d:%d:%d: cannot set freq %d to ep %#x\n", dev->devnum, iface, fmt->altsetting, rate, ep); return err; } if ((err = snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), GET_CUR, USB_TYPE_CLASS|USB_RECIP_ENDPOINT|USB_DIR_IN, SAMPLING_FREQ_CONTROL << 8, ep, data, 3, 1000)) < 0) { - snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep 0x%x\n", + snd_printk(KERN_WARNING "%d:%d:%d: cannot get freq at ep %#x\n", dev->devnum, iface, fmt->altsetting, ep); return 0; /* some devices don't support reading */ } @@ -1365,12 +1346,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->datapipe = usb_sndisocpipe(dev, ep); else subs->datapipe = usb_rcvisocpipe(dev, ep); - if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH && - get_endpoint(alts, 0)->bInterval >= 1 && - get_endpoint(alts, 0)->bInterval <= 4) - subs->datainterval = get_endpoint(alts, 0)->bInterval - 1; - else - subs->datainterval = 0; + subs->datainterval = fmt->datainterval; subs->syncpipe = subs->syncinterval = 0; subs->maxpacksize = fmt->maxpacksize; subs->fill_max = 0; @@ -1431,9 +1407,11 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) subs->cur_audiofmt = fmt; #if 0 - printk("setting done: format = %d, rate = %d..%d, channels = %d\n", + printk(KERN_DEBUG + "setting done: format = %d, rate = %d..%d, channels = %d\n", fmt->format, fmt->rate_min, fmt->rate_max, fmt->channels); - printk(" datapipe = 0x%0x, syncpipe = 0x%0x\n", + printk(KERN_DEBUG + " datapipe = 0x%0x, syncpipe = 0x%0x\n", subs->datapipe, subs->syncpipe); #endif @@ -1468,7 +1446,7 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, channels = params_channels(hw_params); fmt = find_format(subs, format, rate, channels); if (!fmt) { - snd_printd(KERN_DEBUG "cannot set format: format = 0x%x, rate = %d, channels = %d\n", + snd_printd(KERN_DEBUG "cannot set format: format = %#x, rate = %d, channels = %d\n", format, rate, channels); return -EINVAL; } @@ -1581,11 +1559,15 @@ static struct snd_pcm_hardware snd_usb_hardware = #define hwc_debug(fmt, args...) /**/ #endif -static int hw_check_valid_format(struct snd_pcm_hw_params *params, struct audioformat *fp) +static int hw_check_valid_format(struct snd_usb_substream *subs, + struct snd_pcm_hw_params *params, + struct audioformat *fp) { struct snd_interval *it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *ct = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); struct snd_mask *fmts = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + struct snd_interval *pt = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); + unsigned int ptime; /* check the format */ if (!snd_mask_test(fmts, fp->format)) { @@ -1606,6 +1588,14 @@ static int hw_check_valid_format(struct snd_pcm_hw_params *params, struct audiof hwc_debug(" > check: rate_max %d < min %d\n", fp->rate_max, it->min); return 0; } + /* check whether the period time is >= the data packet interval */ + if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) { + ptime = 125 * (1 << fp->datainterval); + if (ptime > pt->max || (ptime == pt->max && pt->openmax)) { + hwc_debug(" > check: ptime %u > max %u\n", ptime, pt->max); + return 0; + } + } return 1; } @@ -1624,7 +1614,7 @@ static int hw_rule_rate(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(subs, params, fp)) continue; if (changed++) { if (rmin > fp->rate_min) @@ -1678,7 +1668,7 @@ static int hw_rule_channels(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(subs, params, fp)) continue; if (changed++) { if (rmin > fp->channels) @@ -1731,7 +1721,7 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, list_for_each(p, &subs->fmt_list) { struct audioformat *fp; fp = list_entry(p, struct audioformat, list); - if (!hw_check_valid_format(params, fp)) + if (!hw_check_valid_format(subs, params, fp)) continue; fbits |= (1ULL << fp->format); } @@ -1749,95 +1739,42 @@ static int hw_rule_format(struct snd_pcm_hw_params *params, return changed; } -#define MAX_MASK 64 - -/* - * check whether the registered audio formats need special hw-constraints - */ -static int check_hw_params_convention(struct snd_usb_substream *subs) +static int hw_rule_period_time(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) { - int i; - u32 *channels; - u32 *rates; - u32 cmaster, rmaster; - u32 rate_min = 0, rate_max = 0; - struct list_head *p; - int err = 1; - - channels = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL); - rates = kcalloc(MAX_MASK, sizeof(u32), GFP_KERNEL); - if (!channels || !rates) { - err = -ENOMEM; - goto __out; - } + struct snd_usb_substream *subs = rule->private; + struct audioformat *fp; + struct snd_interval *it; + unsigned char min_datainterval; + unsigned int pmin; + int changed; - list_for_each(p, &subs->fmt_list) { - struct audioformat *f; - f = list_entry(p, struct audioformat, list); - /* unconventional channels? */ - if (f->channels > 32) - goto __out; - /* continuous rate min/max matches? */ - if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) { - if (rate_min && f->rate_min != rate_min) - goto __out; - if (rate_max && f->rate_max != rate_max) - goto __out; - rate_min = f->rate_min; - rate_max = f->rate_max; - } - /* combination of continuous rates and fixed rates? */ - if (rates[f->format] & SNDRV_PCM_RATE_CONTINUOUS) { - if (f->rates != rates[f->format]) - goto __out; - } - if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) { - if (rates[f->format] && rates[f->format] != f->rates) - goto __out; - } - channels[f->format] |= (1 << f->channels); - rates[f->format] |= f->rates; - /* needs knot? */ - if (f->rates & SNDRV_PCM_RATE_KNOT) - goto __out; - } - /* check whether channels and rates match for all formats */ - cmaster = rmaster = 0; - for (i = 0; i < MAX_MASK; i++) { - if (cmaster != channels[i] && cmaster && channels[i]) - goto __out; - if (rmaster != rates[i] && rmaster && rates[i]) - goto __out; - if (channels[i]) - cmaster = channels[i]; - if (rates[i]) - rmaster = rates[i]; - } - /* check whether channels match for all distinct rates */ - memset(channels, 0, MAX_MASK * sizeof(u32)); - list_for_each(p, &subs->fmt_list) { - struct audioformat *f; - f = list_entry(p, struct audioformat, list); - if (f->rates & SNDRV_PCM_RATE_CONTINUOUS) + it = hw_param_interval(params, SNDRV_PCM_HW_PARAM_PERIOD_TIME); + hwc_debug("hw_rule_period_time: (%u,%u)\n", it->min, it->max); + min_datainterval = 0xff; + list_for_each_entry(fp, &subs->fmt_list, list) { + if (!hw_check_valid_format(subs, params, fp)) continue; - for (i = 0; i < 32; i++) { - if (f->rates & (1 << i)) - channels[i] |= (1 << f->channels); - } + min_datainterval = min(min_datainterval, fp->datainterval); + } + if (min_datainterval == 0xff) { + hwc_debug(" --> get emtpy\n"); + it->empty = 1; + return -EINVAL; } - cmaster = 0; - for (i = 0; i < 32; i++) { - if (cmaster != channels[i] && cmaster && channels[i]) - goto __out; - if (channels[i]) - cmaster = channels[i]; + pmin = 125 * (1 << min_datainterval); + changed = 0; + if (it->min < pmin) { + it->min = pmin; + it->openmin = 0; + changed = 1; } - err = 0; - - __out: - kfree(channels); - kfree(rates); - return err; + if (snd_interval_checkempty(it)) { + it->empty = 1; + return -EINVAL; + } + hwc_debug(" --> (%u,%u) (changed = %d)\n", it->min, it->max, changed); + return changed; } /* @@ -1885,6 +1822,8 @@ static int snd_usb_pcm_check_knot(struct snd_pcm_runtime *runtime, static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substream *subs) { struct list_head *p; + unsigned int pt, ptmin; + int param_period_time_if_needed; int err; runtime->hw.formats = subs->formats; @@ -1894,6 +1833,7 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.channels_min = 256; runtime->hw.channels_max = 0; runtime->hw.rates = 0; + ptmin = UINT_MAX; /* check min/max rates and channels */ list_for_each(p, &subs->fmt_list) { struct audioformat *fp; @@ -1912,42 +1852,54 @@ static int setup_hw_info(struct snd_pcm_runtime *runtime, struct snd_usb_substre runtime->hw.period_bytes_min = runtime->hw.period_bytes_max = fp->frame_size; } + pt = 125 * (1 << fp->datainterval); + ptmin = min(ptmin, pt); } - /* set the period time minimum 1ms */ - /* FIXME: high-speed mode allows 125us minimum period, but many parts - * in the current code assume the 1ms period. - */ + param_period_time_if_needed = SNDRV_PCM_HW_PARAM_PERIOD_TIME; + if (snd_usb_get_speed(subs->dev) != USB_SPEED_HIGH) + /* full speed devices have fixed data packet interval */ + ptmin = 1000; + if (ptmin == 1000) + /* if period time doesn't go below 1 ms, no rules needed */ + param_period_time_if_needed = -1; snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_TIME, - 1000 * MIN_PACKS_URB, - /*(nrpacks * MAX_URBS) * 1000*/ UINT_MAX); - - err = check_hw_params_convention(subs); - if (err < 0) + ptmin, UINT_MAX); + + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + hw_rule_rate, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, + -1)) < 0) return err; - else if (err) { - hwc_debug("setting extra hw constraints...\n"); - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, - hw_rule_rate, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_CHANNELS, - -1)) < 0) - return err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, - hw_rule_channels, subs, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_RATE, - -1)) < 0) - return err; - if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, - hw_rule_format, subs, - SNDRV_PCM_HW_PARAM_RATE, - SNDRV_PCM_HW_PARAM_CHANNELS, - -1)) < 0) - return err; - if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + hw_rule_channels, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_RATE, + param_period_time_if_needed, + -1)) < 0) + return err; + if ((err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + hw_rule_format, subs, + SNDRV_PCM_HW_PARAM_RATE, + SNDRV_PCM_HW_PARAM_CHANNELS, + param_period_time_if_needed, + -1)) < 0) + return err; + if (param_period_time_if_needed >= 0) { + err = snd_pcm_hw_rule_add(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_TIME, + hw_rule_period_time, subs, + SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, + -1); + if (err < 0) return err; } + if ((err = snd_usb_pcm_check_knot(runtime, subs)) < 0) + return err; return 0; } @@ -2160,7 +2112,8 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s fp = list_entry(p, struct audioformat, list); snd_iprintf(buffer, " Interface %d\n", fp->iface); snd_iprintf(buffer, " Altset %d\n", fp->altsetting); - snd_iprintf(buffer, " Format: 0x%x\n", fp->format); + snd_iprintf(buffer, " Format: %#x (%d bits)\n", + fp->format, snd_pcm_format_width(fp->format)); snd_iprintf(buffer, " Channels: %d\n", fp->channels); snd_iprintf(buffer, " Endpoint: %d %s (%s)\n", fp->endpoint & USB_ENDPOINT_NUMBER_MASK, @@ -2179,8 +2132,11 @@ static void proc_dump_substream_formats(struct snd_usb_substream *subs, struct s } snd_iprintf(buffer, "\n"); } + if (snd_usb_get_speed(subs->dev) == USB_SPEED_HIGH) + snd_iprintf(buffer, " Data packet interval: %d us\n", + 125 * (1 << fp->datainterval)); // snd_iprintf(buffer, " Max Packet Size = %d\n", fp->maxpacksize); - // snd_iprintf(buffer, " EP Attribute = 0x%x\n", fp->attributes); + // snd_iprintf(buffer, " EP Attribute = %#x\n", fp->attributes); } } @@ -2524,7 +2480,6 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform * build the rate table and bitmap flags */ int r, idx; - unsigned int nonzero_rates = 0; fp->rate_table = kmalloc(sizeof(int) * nr_rates, GFP_KERNEL); if (fp->rate_table == NULL) { @@ -2532,24 +2487,27 @@ static int parse_audio_format_rates(struct snd_usb_audio *chip, struct audioform return -1; } - fp->nr_rates = nr_rates; - fp->rate_min = fp->rate_max = combine_triple(&fmt[8]); + fp->nr_rates = 0; + fp->rate_min = fp->rate_max = 0; for (r = 0, idx = offset + 1; r < nr_rates; r++, idx += 3) { unsigned int rate = combine_triple(&fmt[idx]); + if (!rate) + continue; /* C-Media CM6501 mislabels its 96 kHz altsetting */ if (rate == 48000 && nr_rates == 1 && - chip->usb_id == USB_ID(0x0d8c, 0x0201) && + (chip->usb_id == USB_ID(0x0d8c, 0x0201) || + chip->usb_id == USB_ID(0x0d8c, 0x0102)) && fp->altsetting == 5 && fp->maxpacksize == 392) rate = 96000; - fp->rate_table[r] = rate; - nonzero_rates |= rate; - if (rate < fp->rate_min) + fp->rate_table[fp->nr_rates] = rate; + if (!fp->rate_min || rate < fp->rate_min) fp->rate_min = rate; - else if (rate > fp->rate_max) + if (!fp->rate_max || rate > fp->rate_max) fp->rate_max = rate; fp->rates |= snd_pcm_rate_to_rate_bit(rate); + fp->nr_rates++; } - if (!nonzero_rates) { + if (!fp->nr_rates) { hwc_debug("All rates were zero. Skipping format!\n"); return -1; } @@ -2619,7 +2577,7 @@ static int parse_audio_format_ii(struct snd_usb_audio *chip, struct audioformat fp->format = SNDRV_PCM_FORMAT_MPEG; break; default: - snd_printd(KERN_INFO "%d:%u:%d : unknown format tag 0x%x is detected. processed as MPEG.\n", + snd_printd(KERN_INFO "%d:%u:%d : unknown format tag %#x is detected. processed as MPEG.\n", chip->dev->devnum, fp->iface, fp->altsetting, format); fp->format = SNDRV_PCM_FORMAT_MPEG; break; @@ -2670,6 +2628,17 @@ static int parse_audio_format(struct snd_usb_audio *chip, struct audioformat *fp return 0; } +static unsigned char parse_datainterval(struct snd_usb_audio *chip, + struct usb_host_interface *alts) +{ + if (snd_usb_get_speed(chip->dev) == USB_SPEED_HIGH && + get_endpoint(alts, 0)->bInterval >= 1 && + get_endpoint(alts, 0)->bInterval <= 4) + return get_endpoint(alts, 0)->bInterval - 1; + else + return 0; +} + static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) @@ -2775,6 +2744,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) fp->altset_idx = i; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); if (snd_usb_get_speed(dev) == USB_SPEED_HIGH) fp->maxpacksize = (((fp->maxpacksize >> 11) & 3) + 1) @@ -2817,7 +2787,7 @@ static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) continue; } - snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint 0x%x\n", dev->devnum, iface_no, altno, fp->endpoint); + snd_printdd(KERN_INFO "%d:%u:%d: add audio endpoint %#x\n", dev->devnum, iface_no, altno, fp->endpoint); err = add_audio_endpoint(chip, stream, fp); if (err < 0) { kfree(fp->rate_table); @@ -2966,6 +2936,8 @@ static int create_fixed_stream_quirk(struct snd_usb_audio *chip, return -EINVAL; } alts = &iface->altsetting[fp->altset_idx]; + fp->datainterval = parse_datainterval(chip, alts); + fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); usb_set_interface(chip->dev, fp->iface, 0); init_usb_pitch(chip->dev, fp->iface, alts, fp); init_usb_sample_rate(chip->dev, fp->iface, alts, fp, fp->rate_max); @@ -3059,6 +3031,7 @@ static int create_uaxx_quirk(struct snd_usb_audio *chip, fp->iface = altsd->bInterfaceNumber; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = 0; fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); switch (fp->maxpacksize) { @@ -3126,6 +3099,7 @@ static int create_ua1000_quirk(struct snd_usb_audio *chip, fp->iface = altsd->bInterfaceNumber; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); fp->rate_max = fp->rate_min = combine_triple(&alts->extra[8]); @@ -3178,6 +3152,7 @@ static int create_ua101_quirk(struct snd_usb_audio *chip, fp->iface = altsd->bInterfaceNumber; fp->endpoint = get_endpoint(alts, 0)->bEndpointAddress; fp->ep_attr = get_endpoint(alts, 0)->bmAttributes; + fp->datainterval = parse_datainterval(chip, alts); fp->maxpacksize = le16_to_cpu(get_endpoint(alts, 0)->wMaxPacketSize); fp->rate_max = fp->rate_min = combine_triple(&alts->extra[15]); @@ -3763,7 +3738,7 @@ static int usb_audio_resume(struct usb_interface *intf) static int __init snd_usb_audio_init(void) { - if (nrpacks < MIN_PACKS_URB || nrpacks > MAX_PACKS) { + if (nrpacks < 1 || nrpacks > MAX_PACKS) { printk(KERN_WARNING "invalid nrpacks value.\n"); return -EINVAL; } diff --git a/sound/usb/usbmidi.c b/sound/usb/usbmidi.c index 320641a..26bad37 100644 --- a/sound/usb/usbmidi.c +++ b/sound/usb/usbmidi.c @@ -1625,6 +1625,7 @@ static int snd_usbmidi_create_endpoints_midiman(struct snd_usb_midi* umidi, } ep_info.out_ep = get_endpoint(hostif, 2)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK; + ep_info.out_interval = 0; ep_info.out_cables = endpoint->out_cables & 0x5555; err = snd_usbmidi_out_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]); if (err < 0) diff --git a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c index 00397c8..ecb58e7 100644 --- a/sound/usb/usbmixer.c +++ b/sound/usb/usbmixer.c @@ -66,6 +66,7 @@ static const struct rc_config { { USB_ID(0x041e, 0x3000), 0, 1, 2, 1, 18, 0x0013 }, /* Extigy */ { USB_ID(0x041e, 0x3020), 2, 1, 6, 6, 18, 0x0013 }, /* Audigy 2 NX */ { USB_ID(0x041e, 0x3040), 2, 2, 6, 6, 2, 0x6e91 }, /* Live! 24-bit */ + { USB_ID(0x041e, 0x3048), 2, 2, 6, 6, 2, 0x6e91 }, /* Toshiba SB0500 */ }; struct usb_mixer_interface { @@ -78,7 +79,6 @@ struct usb_mixer_interface { /* Sound Blaster remote control stuff */ const struct rc_config *rc_cfg; - unsigned long rc_hwdep_open; u32 rc_code; wait_queue_head_t rc_waitq; struct urb *rc_urb; @@ -110,6 +110,8 @@ struct mixer_build { const struct usbmix_selector_map *selector_map; }; +#define MAX_CHANNELS 10 /* max logical channels */ + struct usb_mixer_elem_info { struct usb_mixer_interface *mixer; struct usb_mixer_elem_info *next_id_elem; /* list of controls with same id */ @@ -120,6 +122,8 @@ struct usb_mixer_elem_info { int channels; int val_type; int min, max, res; + int cached; + int cache_val[MAX_CHANNELS]; u8 initialized; }; @@ -181,8 +185,6 @@ enum { USB_PROC_DCR_RELEASE = 6, }; -#define MAX_CHANNELS 10 /* max logical channels */ - /* * manual mapping of mixer names @@ -219,7 +221,10 @@ static int check_ignored_ctl(struct mixer_build *state, int unitid, int control) for (p = state->map; p->id; p++) { if (p->id == unitid && ! p->name && (! control || ! p->control || control == p->control)) { - // printk("ignored control %d:%d\n", unitid, control); + /* + printk(KERN_DEBUG "ignored control %d:%d\n", + unitid, control); + */ return 1; } } @@ -376,11 +381,35 @@ static int get_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int * } /* channel = 0: master, 1 = first channel */ -static inline int get_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int *value) +static inline int get_cur_mix_raw(struct usb_mixer_elem_info *cval, + int channel, int *value) { return get_ctl_value(cval, GET_CUR, (cval->control << 8) | channel, value); } +static int get_cur_mix_value(struct usb_mixer_elem_info *cval, + int channel, int index, int *value) +{ + int err; + + if (cval->cached & (1 << channel)) { + *value = cval->cache_val[index]; + return 0; + } + err = get_cur_mix_raw(cval, channel, value); + if (err < 0) { + if (!cval->mixer->ignore_ctl_error) + snd_printd(KERN_ERR "cannot get current value for " + "control %d ch %d: err = %d\n", + cval->control, channel, err); + return err; + } + cval->cached |= 1 << channel; + cval->cache_val[index] = *value; + return 0; +} + + /* * set a mixer value */ @@ -412,9 +441,17 @@ static int set_cur_ctl_value(struct usb_mixer_elem_info *cval, int validx, int v return set_ctl_value(cval, SET_CUR, validx, value); } -static inline int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, int value) +static int set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel, + int index, int value) { - return set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, value); + int err; + err = set_ctl_value(cval, SET_CUR, (cval->control << 8) | channel, + value); + if (err < 0) + return err; + cval->cached |= 1 << channel; + cval->cache_val[index] = value; + return 0; } /* @@ -718,7 +755,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) if (cval->min + cval->res < cval->max) { int last_valid_res = cval->res; int saved, test, check; - get_cur_mix_value(cval, minchn, &saved); + get_cur_mix_raw(cval, minchn, &saved); for (;;) { test = saved; if (test < cval->max) @@ -726,8 +763,8 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) else test -= cval->res; if (test < cval->min || test > cval->max || - set_cur_mix_value(cval, minchn, test) || - get_cur_mix_value(cval, minchn, &check)) { + set_cur_mix_value(cval, minchn, 0, test) || + get_cur_mix_raw(cval, minchn, &check)) { cval->res = last_valid_res; break; } @@ -735,7 +772,7 @@ static int get_min_max(struct usb_mixer_elem_info *cval, int default_min) break; cval->res *= 2; } - set_cur_mix_value(cval, minchn, saved); + set_cur_mix_value(cval, minchn, 0, saved); } cval->initialized = 1; @@ -775,35 +812,25 @@ static int mixer_ctl_feature_get(struct snd_kcontrol *kcontrol, struct snd_ctl_e struct usb_mixer_elem_info *cval = kcontrol->private_data; int c, cnt, val, err; + ucontrol->value.integer.value[0] = cval->min; if (cval->cmask) { cnt = 0; for (c = 0; c < MAX_CHANNELS; c++) { - if (cval->cmask & (1 << c)) { - err = get_cur_mix_value(cval, c + 1, &val); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) { - ucontrol->value.integer.value[0] = cval->min; - return 0; - } - snd_printd(KERN_ERR "cannot get current value for control %d ch %d: err = %d\n", cval->control, c + 1, err); - return err; - } - val = get_relative_value(cval, val); - ucontrol->value.integer.value[cnt] = val; - cnt++; - } + if (!(cval->cmask & (1 << c))) + continue; + err = get_cur_mix_value(cval, c + 1, cnt, &val); + if (err < 0) + return cval->mixer->ignore_ctl_error ? 0 : err; + val = get_relative_value(cval, val); + ucontrol->value.integer.value[cnt] = val; + cnt++; } + return 0; } else { /* master channel */ - err = get_cur_mix_value(cval, 0, &val); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) { - ucontrol->value.integer.value[0] = cval->min; - return 0; - } - snd_printd(KERN_ERR "cannot get current value for control %d master ch: err = %d\n", cval->control, err); - return err; - } + err = get_cur_mix_value(cval, 0, 0, &val); + if (err < 0) + return cval->mixer->ignore_ctl_error ? 0 : err; val = get_relative_value(cval, val); ucontrol->value.integer.value[0] = val; } @@ -820,34 +847,28 @@ static int mixer_ctl_feature_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e if (cval->cmask) { cnt = 0; for (c = 0; c < MAX_CHANNELS; c++) { - if (cval->cmask & (1 << c)) { - err = get_cur_mix_value(cval, c + 1, &oval); - if (err < 0) { - if (cval->mixer->ignore_ctl_error) - return 0; - return err; - } - val = ucontrol->value.integer.value[cnt]; - val = get_abs_value(cval, val); - if (oval != val) { - set_cur_mix_value(cval, c + 1, val); - changed = 1; - } - get_cur_mix_value(cval, c + 1, &val); - cnt++; + if (!(cval->cmask & (1 << c))) + continue; + err = get_cur_mix_value(cval, c + 1, cnt, &oval); + if (err < 0) + return cval->mixer->ignore_ctl_error ? 0 : err; + val = ucontrol->value.integer.value[cnt]; + val = get_abs_value(cval, val); + if (oval != val) { + set_cur_mix_value(cval, c + 1, cnt, val); + changed = 1; } + cnt++; } } else { /* master channel */ - err = get_cur_mix_value(cval, 0, &oval); - if (err < 0 && cval->mixer->ignore_ctl_error) - return 0; + err = get_cur_mix_value(cval, 0, 0, &oval); if (err < 0) - return err; + return cval->mixer->ignore_ctl_error ? 0 : err; val = ucontrol->value.integer.value[0]; val = get_abs_value(cval, val); if (val != oval) { - set_cur_mix_value(cval, 0, val); + set_cur_mix_value(cval, 0, 0, val); changed = 1; } } @@ -1706,7 +1727,8 @@ static void snd_usb_mixer_memory_change(struct usb_mixer_interface *mixer, break; /* live24ext: 4 = line-in jack */ case 3: /* hp-out jack (may actuate Mute) */ - if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) snd_usb_mixer_notify_id(mixer, mixer->rc_cfg->mute_mixer_id); break; default: @@ -1797,24 +1819,6 @@ static void snd_usb_soundblaster_remote_complete(struct urb *urb) wake_up(&mixer->rc_waitq); } -static int snd_usb_sbrc_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - struct usb_mixer_interface *mixer = hw->private_data; - - if (test_and_set_bit(0, &mixer->rc_hwdep_open)) - return -EBUSY; - return 0; -} - -static int snd_usb_sbrc_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - struct usb_mixer_interface *mixer = hw->private_data; - - clear_bit(0, &mixer->rc_hwdep_open); - smp_mb__after_clear_bit(); - return 0; -} - static long snd_usb_sbrc_hwdep_read(struct snd_hwdep *hw, char __user *buf, long count, loff_t *offset) { @@ -1867,9 +1871,8 @@ static int snd_usb_soundblaster_remote_init(struct usb_mixer_interface *mixer) hwdep->iface = SNDRV_HWDEP_IFACE_SB_RC; hwdep->private_data = mixer; hwdep->ops.read = snd_usb_sbrc_hwdep_read; - hwdep->ops.open = snd_usb_sbrc_hwdep_open; - hwdep->ops.release = snd_usb_sbrc_hwdep_release; hwdep->ops.poll = snd_usb_sbrc_hwdep_poll; + hwdep->exclusive = 1; mixer->rc_urb = usb_alloc_urb(0, GFP_KERNEL); if (!mixer->rc_urb) @@ -1956,8 +1959,9 @@ static int snd_audigy2nx_controls_create(struct usb_mixer_interface *mixer) int i, err; for (i = 0; i < ARRAY_SIZE(snd_audigy2nx_controls); ++i) { - if (i > 1 && /* Live24ext has 2 LEDs only */ - mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + if (i > 1 && /* Live24ext has 2 LEDs only */ + (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048))) break; err = snd_ctl_add(mixer->chip->card, snd_ctl_new1(&snd_audigy2nx_controls[i], mixer)); @@ -1994,7 +1998,8 @@ static void snd_audigy2nx_proc_read(struct snd_info_entry *entry, snd_iprintf(buffer, "%s jacks\n\n", mixer->chip->card->shortname); if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020)) jacks = jacks_audigy2nx; - else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) + else if (mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) jacks = jacks_live24ext; else return; @@ -2044,7 +2049,8 @@ int snd_usb_create_mixer(struct snd_usb_audio *chip, int ctrlif, goto _error; if (mixer->chip->usb_id == USB_ID(0x041e, 0x3020) || - mixer->chip->usb_id == USB_ID(0x041e, 0x3040)) { + mixer->chip->usb_id == USB_ID(0x041e, 0x3040) || + mixer->chip->usb_id == USB_ID(0x041e, 0x3048)) { struct snd_info_entry *entry; if ((err = snd_audigy2nx_controls_create(mixer)) < 0) diff --git a/sound/usb/usbmixer_maps.c b/sound/usb/usbmixer_maps.c index d755be0..3e5d66c 100644 --- a/sound/usb/usbmixer_maps.c +++ b/sound/usb/usbmixer_maps.c @@ -261,6 +261,22 @@ static struct usbmix_name_map aureon_51_2_map[] = { {} /* terminator */ }; +static struct usbmix_name_map scratch_live_map[] = { + /* 1: IT Line 1 (USB streaming) */ + /* 2: OT Line 1 (Speaker) */ + /* 3: IT Line 1 (Line connector) */ + { 4, "Line 1 In" }, /* FU */ + /* 5: OT Line 1 (USB streaming) */ + /* 6: IT Line 2 (USB streaming) */ + /* 7: OT Line 2 (Speaker) */ + /* 8: IT Line 2 (Line connector) */ + { 9, "Line 2 In" }, /* FU */ + /* 10: OT Line 2 (USB streaming) */ + /* 11: IT Mic (Line connector) */ + /* 12: OT Mic (USB streaming) */ + { 0 } /* terminator */ +}; + /* * Control map entries */ @@ -285,6 +301,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = live24ext_map, }, { + .id = USB_ID(0x041e, 0x3048), + .map = audigy2nx_map, + .selector_map = audigy2nx_selectors, + }, + { /* Hercules DJ Console (Windows Edition) */ .id = USB_ID(0x06f8, 0xb000), .ignore_ctl_error = 1, @@ -311,6 +332,11 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x0ccd, 0x0028), .map = aureon_51_2_map, }, + { + .id = USB_ID(0x13e5, 0x0001), + .map = scratch_live_map, + .ignore_ctl_error = 1, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/usbquirks.h b/sound/usb/usbquirks.h index 9211575..647ef50 100644 --- a/sound/usb/usbquirks.h +++ b/sound/usb/usbquirks.h @@ -39,6 +39,16 @@ .idProduct = prod, \ .bInterfaceClass = USB_CLASS_VENDOR_SPEC +/* Creative/Toshiba Multimedia Center SB-0500 */ +{ + USB_DEVICE(0x041e, 0x3048), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Toshiba", + .product_name = "SB-0500", + .ifnum = QUIRK_NO_INTERFACE + } +}, + /* Creative/E-Mu devices */ { USB_DEVICE(0x041e, 0x3010), @@ -128,6 +138,14 @@ .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL }, +{ + USB_DEVICE(0x046d, 0x0990), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Logitech, Inc.", + .product_name = "QuickCam Pro 9000", + .ifnum = QUIRK_NO_INTERFACE + } +}, /* * Yamaha devices diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 1558a5c..4af8740 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -30,9 +30,6 @@ #include "usbusx2y.h" #include "usX2Yhwdep.h" -int usX2Y_hwdep_pcm_new(struct snd_card *card); - - static int snd_us428ctls_vm_fault(struct vm_area_struct *area, struct vm_fault *vmf) { @@ -106,16 +103,6 @@ static unsigned int snd_us428ctls_poll(struct snd_hwdep *hw, struct file *file, } -static int snd_usX2Y_hwdep_open(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - -static int snd_usX2Y_hwdep_release(struct snd_hwdep *hw, struct file *file) -{ - return 0; -} - static int snd_usX2Y_hwdep_dsp_status(struct snd_hwdep *hw, struct snd_hwdep_dsp_status *info) { @@ -267,8 +254,6 @@ int usX2Y_hwdep_new(struct snd_card *card, struct usb_device* device) hw->iface = SNDRV_HWDEP_IFACE_USX2Y; hw->private_data = usX2Y(card); - hw->ops.open = snd_usX2Y_hwdep_open; - hw->ops.release = snd_usX2Y_hwdep_release; hw->ops.dsp_status = snd_usX2Y_hwdep_dsp_status; hw->ops.dsp_load = snd_usX2Y_hwdep_dsp_load; hw->ops.mmap = snd_us428ctls_mmap; diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 70b9635..24393da 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -557,7 +557,7 @@ static void stream_start(struct usb_stream_kernel *sk, s->idle_insize -= max_diff - max_diff_0; s->idle_insize += urb_size - s->period_size; if (s->idle_insize < 0) { - snd_printk("%i %i %i\n", + snd_printk(KERN_WARNING "%i %i %i\n", s->idle_insize, urb_size, s->period_size); return; } else if (s->idle_insize == 0) { diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index af8b849..5ce0da2 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -227,9 +227,9 @@ static void i_usX2Y_In04Int(struct urb *urb) if (usX2Y->US04) { if (0 == usX2Y->US04->submitted) - do + do { err = usb_submit_urb(usX2Y->US04->urb[usX2Y->US04->submitted++], GFP_ATOMIC); - while (!err && usX2Y->US04->submitted < usX2Y->US04->len); + } while (!err && usX2Y->US04->submitted < usX2Y->US04->len); } else if (us428ctls && us428ctls->p4outLast >= 0 && us428ctls->p4outLast < N_us428_p4out_BUFS) { if (us428ctls->p4outLast != us428ctls->p4outSent) { diff --git a/sound/usb/usx2y/usx2yhwdeppcm.h b/sound/usb/usx2y/usx2yhwdeppcm.h index c3382fd..9c4fb84 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.h +++ b/sound/usb/usx2y/usx2yhwdeppcm.h @@ -18,3 +18,5 @@ struct snd_usX2Y_hwdep_pcm_shm { volatile unsigned captured_iso_frames; int capture_iso_start; }; + +int usX2Y_hwdep_pcm_new(struct snd_card *card); |
