diff options
Diffstat (limited to 'sound/soc')
32 files changed, 238 insertions, 274 deletions
diff --git a/sound/soc/au1x/Kconfig b/sound/soc/au1x/Kconfig index 410a893..4b67140 100644 --- a/sound/soc/au1x/Kconfig +++ b/sound/soc/au1x/Kconfig @@ -22,11 +22,13 @@ config SND_SOC_AU1XPSC_AC97 ## ## Boards ## -config SND_SOC_SAMPLE_PSC_AC97 - tristate "Sample Au12x0/Au1550 PSC AC97 sound machine" +config SND_SOC_DB1200 + tristate "DB1200 AC97+I2S audio support" depends on SND_SOC_AU1XPSC select SND_SOC_AU1XPSC_AC97 select SND_SOC_AC97_CODEC + select SND_SOC_AU1XPSC_I2S + select SND_SOC_WM8731 help - This is a sample AC97 sound machine for use in Au12x0/Au1550 - based systems which have audio on PSC1 (e.g. Db1200 demoboard). + Select this option to enable audio (AC97 or I2S) on the + Alchemy/AMD/RMI DB1200 demoboard. diff --git a/sound/soc/au1x/Makefile b/sound/soc/au1x/Makefile index 6c6950b..1687307 100644 --- a/sound/soc/au1x/Makefile +++ b/sound/soc/au1x/Makefile @@ -8,6 +8,6 @@ obj-$(CONFIG_SND_SOC_AU1XPSC_I2S) += snd-soc-au1xpsc-i2s.o obj-$(CONFIG_SND_SOC_AU1XPSC_AC97) += snd-soc-au1xpsc-ac97.o # Boards -snd-soc-sample-ac97-objs := sample-ac97.o +snd-soc-db1200-objs := db1200.o -obj-$(CONFIG_SND_SOC_SAMPLE_PSC_AC97) += snd-soc-sample-ac97.o +obj-$(CONFIG_SND_SOC_DB1200) += snd-soc-db1200.o diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c new file mode 100644 index 0000000..cdf7be1 --- /dev/null +++ b/sound/soc/au1x/db1200.c @@ -0,0 +1,141 @@ +/* + * DB1200 ASoC audio fabric support code. + * + * (c) 2008-9 Manuel Lauss <manuel.lauss@gmail.com> + * + */ + +#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 <asm/mach-au1x00/au1000.h> +#include <asm/mach-au1x00/au1xxx_psc.h> +#include <asm/mach-au1x00/au1xxx_dbdma.h> +#include <asm/mach-db1x00/bcsr.h> + +#include "../codecs/ac97.h" +#include "../codecs/wm8731.h" +#include "psc.h" + +/*------------------------- AC97 PART ---------------------------*/ + +static struct snd_soc_dai_link db1200_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &au1xpsc_ac97_dai, + .codec_dai = &ac97_dai, +}; + +static struct snd_soc_card db1200_ac97_machine = { + .name = "DB1200_AC97", + .dai_link = &db1200_ac97_dai, + .num_links = 1, + .platform = &au1xpsc_soc_platform, +}; + +static struct snd_soc_device db1200_ac97_devdata = { + .card = &db1200_ac97_machine, + .codec_dev = &soc_codec_dev_ac97, +}; + +/*------------------------- I2S PART ---------------------------*/ + +static int db1200_i2s_startup(struct snd_pcm_substream *substream) +{ + 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; + + /* WM8731 has its own 12MHz crystal */ + snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK, + 12000000, SND_SOC_CLOCK_IN); + + /* codec is bitclock and lrclk master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_LEFT_J | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + goto out; + + ret = 0; +out: + return ret; +} + +static struct snd_soc_ops db1200_i2s_wm8731_ops = { + .startup = db1200_i2s_startup, +}; + +static struct snd_soc_dai_link db1200_i2s_dai = { + .name = "WM8731", + .stream_name = "WM8731 PCM", + .cpu_dai = &au1xpsc_i2s_dai, + .codec_dai = &wm8731_dai, + .ops = &db1200_i2s_wm8731_ops, +}; + +static struct snd_soc_card db1200_i2s_machine = { + .name = "DB1200_I2S", + .dai_link = &db1200_i2s_dai, + .num_links = 1, + .platform = &au1xpsc_soc_platform, +}; + +static struct snd_soc_device db1200_i2s_devdata = { + .card = &db1200_i2s_machine, + .codec_dev = &soc_codec_dev_wm8731, +}; + +/*------------------------- COMMON PART ---------------------------*/ + +static struct platform_device *db1200_asoc_dev; + +static int __init db1200_audio_load(void) +{ + int ret; + + ret = -ENOMEM; + db1200_asoc_dev = platform_device_alloc("soc-audio", -1); + if (!db1200_asoc_dev) + goto out; + + /* DB1200 board setup set PSC1MUX to preferred audio device */ + if (bcsr_read(BCSR_RESETS) & BCSR_RESETS_PSC1MUX) + platform_set_drvdata(db1200_asoc_dev, &db1200_i2s_devdata); + else + platform_set_drvdata(db1200_asoc_dev, &db1200_ac97_devdata); + + db1200_ac97_devdata.dev = &db1200_asoc_dev->dev; + db1200_i2s_devdata.dev = &db1200_asoc_dev->dev; + ret = platform_device_add(db1200_asoc_dev); + + if (ret) { + platform_device_put(db1200_asoc_dev); + db1200_asoc_dev = NULL; + } +out: + return ret; +} + +static void __exit db1200_audio_unload(void) +{ + platform_device_unregister(db1200_asoc_dev); +} + +module_init(db1200_audio_load); +module_exit(db1200_audio_unload); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("DB1200 ASoC audio support"); +MODULE_AUTHOR("Manuel Lauss"); diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 19e4d37..6d9f4c6 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -51,8 +51,8 @@ struct au1xpsc_audio_dmadata { struct snd_pcm_substream *substream; unsigned long curr_period; /* current segment DDMA is working on */ unsigned long q_period; /* queue period(s) */ - unsigned long dma_area; /* address of queued DMA area */ - unsigned long dma_area_s; /* start address of DMA area */ + dma_addr_t dma_area; /* address of queued DMA area */ + dma_addr_t dma_area_s; /* start address of DMA area */ unsigned long pos; /* current byte position being played */ unsigned long periods; /* number of SG segments in total */ unsigned long period_bytes; /* size in bytes of one SG segment */ @@ -94,8 +94,7 @@ static const struct snd_pcm_hardware au1xpsc_pcm_hardware = { static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) { - au1xxx_dbdma_put_source_flags(cd->ddma_chan, - (void *)phys_to_virt(cd->dma_area), + au1xxx_dbdma_put_source(cd->ddma_chan, cd->dma_area, cd->period_bytes, DDMA_FLAGS_IE); /* update next-to-queue period */ @@ -109,9 +108,8 @@ static void au1x_pcm_queue_tx(struct au1xpsc_audio_dmadata *cd) static void au1x_pcm_queue_rx(struct au1xpsc_audio_dmadata *cd) { - au1xxx_dbdma_put_dest_flags(cd->ddma_chan, - (void *)phys_to_virt(cd->dma_area), - cd->period_bytes, DDMA_FLAGS_IE); + au1xxx_dbdma_put_dest(cd->ddma_chan, cd->dma_area, + cd->period_bytes, DDMA_FLAGS_IE); /* update next-to-queue period */ ++cd->q_period; @@ -233,7 +231,7 @@ static int au1xpsc_pcm_hw_params(struct snd_pcm_substream *substream, pcd->substream = substream; pcd->period_bytes = params_period_bytes(params); pcd->periods = params_periods(params); - pcd->dma_area_s = pcd->dma_area = (unsigned long)runtime->dma_addr; + pcd->dma_area_s = pcd->dma_area = runtime->dma_addr; pcd->q_period = 0; pcd->curr_period = 0; pcd->pos = 0; diff --git a/sound/soc/au1x/sample-ac97.c b/sound/soc/au1x/sample-ac97.c deleted file mode 100644 index 27683eb..0000000 --- a/sound/soc/au1x/sample-ac97.c +++ /dev/null @@ -1,144 +0,0 @@ -/* - * Sample Au12x0/Au1550 PSC AC97 sound machine. - * - * Copyright (c) 2007-2008 Manuel Lauss <mano@roarinelk.homelinux.net> - * - * This program is free software; you can redistribute it and/or modify - * it under the terms outlined in the file COPYING at the root of this - * source archive. - * - * This is a very generic AC97 sound machine driver for boards which - * have (AC97) audio at PSC1 (e.g. DB1200 demoboards). - */ - -#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 <asm/mach-au1x00/au1000.h> -#include <asm/mach-au1x00/au1xxx_psc.h> -#include <asm/mach-au1x00/au1xxx_dbdma.h> - -#include "../codecs/ac97.h" -#include "psc.h" - -static int au1xpsc_sample_ac97_init(struct snd_soc_codec *codec) -{ - snd_soc_dapm_sync(codec); - return 0; -} - -static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai = &au1xpsc_ac97_dai, /* see psc-ac97.c */ - .codec_dai = &ac97_dai, /* see codecs/ac97.c */ - .init = au1xpsc_sample_ac97_init, - .ops = NULL, -}; - -static struct snd_soc_card au1xpsc_sample_ac97_machine = { - .name = "Au1xxx PSC AC97 Audio", - .dai_link = &au1xpsc_sample_ac97_dai, - .num_links = 1, -}; - -static struct snd_soc_device au1xpsc_sample_ac97_devdata = { - .card = &au1xpsc_sample_ac97_machine, - .platform = &au1xpsc_soc_platform, /* see dbdma2.c */ - .codec_dev = &soc_codec_dev_ac97, -}; - -static struct resource au1xpsc_psc1_res[] = { - [0] = { - .start = CPHYSADDR(PSC1_BASE_ADDR), - .end = CPHYSADDR(PSC1_BASE_ADDR) + 0x000fffff, - .flags = IORESOURCE_MEM, - }, - [1] = { -#ifdef CONFIG_SOC_AU1200 - .start = AU1200_PSC1_INT, - .end = AU1200_PSC1_INT, -#elif defined(CONFIG_SOC_AU1550) - .start = AU1550_PSC1_INT, - .end = AU1550_PSC1_INT, -#endif - .flags = IORESOURCE_IRQ, - }, - [2] = { - .start = DSCR_CMD0_PSC1_TX, - .end = DSCR_CMD0_PSC1_TX, - .flags = IORESOURCE_DMA, - }, - [3] = { - .start = DSCR_CMD0_PSC1_RX, - .end = DSCR_CMD0_PSC1_RX, - .flags = IORESOURCE_DMA, - }, -}; - -static struct platform_device *au1xpsc_sample_ac97_dev; - -static int __init au1xpsc_sample_ac97_load(void) -{ - int ret; - -#ifdef CONFIG_SOC_AU1200 - unsigned long io; - - /* modify sys_pinfunc for AC97 on PSC1 */ - io = au_readl(SYS_PINFUNC); - io |= SYS_PINFUNC_P1C; - io &= ~(SYS_PINFUNC_P1A | SYS_PINFUNC_P1B); - au_writel(io, SYS_PINFUNC); - au_sync(); -#endif - - ret = -ENOMEM; - - /* setup PSC clock source for AC97 part: external clock provided - * by codec. The psc-ac97.c driver depends on this setting! - */ - au_writel(PSC_SEL_CLK_SERCLK, PSC1_BASE_ADDR + PSC_SEL_OFFSET); - au_sync(); - - au1xpsc_sample_ac97_dev = platform_device_alloc("soc-audio", -1); - if (!au1xpsc_sample_ac97_dev) - goto out; - - au1xpsc_sample_ac97_dev->resource = - kmemdup(au1xpsc_psc1_res, sizeof(struct resource) * - ARRAY_SIZE(au1xpsc_psc1_res), GFP_KERNEL); - au1xpsc_sample_ac97_dev->num_resources = ARRAY_SIZE(au1xpsc_psc1_res); - au1xpsc_sample_ac97_dev->id = 1; - - platform_set_drvdata(au1xpsc_sample_ac97_dev, - &au1xpsc_sample_ac97_devdata); - au1xpsc_sample_ac97_devdata.dev = &au1xpsc_sample_ac97_dev->dev; - ret = platform_device_add(au1xpsc_sample_ac97_dev); - - if (ret) { - platform_device_put(au1xpsc_sample_ac97_dev); - au1xpsc_sample_ac97_dev = NULL; - } - -out: - return ret; -} - -static void __exit au1xpsc_sample_ac97_exit(void) -{ - platform_device_unregister(au1xpsc_sample_ac97_dev); -} - -module_init(au1xpsc_sample_ac97_load); -module_exit(au1xpsc_sample_ac97_exit); - -MODULE_LICENSE("GPL"); -MODULE_DESCRIPTION("Au1xxx PSC sample AC97 machine"); -MODULE_AUTHOR("Manuel Lauss <mano@roarinelk.homelinux.net>"); diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 69bd0ac..a1bbe16 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -102,6 +102,12 @@ static int ac97_soc_probe(struct platform_device *pdev) INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); + ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); + if (ret < 0) { + printk(KERN_ERR "ASoC: failed to init gen ac97 glue\n"); + goto err; + } + /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index b69861d..3ef16bb 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -470,7 +470,7 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_ak4642); static int __init ak4642_modinit(void) { - int ret; + int ret = 0; #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) ret = i2c_add_driver(&ak4642_i2c_driver); #endif diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index bbc72c2..81b8c9d 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -191,6 +191,7 @@ static int ac97_analog_prepare(struct snd_pcm_substream *substream, vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); vra |= 0x1; /* enable variable rate audio */ + vra &= ~0x4; /* disable SPDIF output */ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); @@ -221,22 +222,6 @@ static int ac97_digital_prepare(struct snd_pcm_substream *substream, return stac9766_ac97_write(codec, reg, runtime->rate); } -static int ac97_digital_trigger(struct snd_pcm_substream *substream, - int cmd, struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - unsigned short vra; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_STOP: - vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); - vra &= !0x04; - stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra); - break; - } - return 0; -} - static int stac9766_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -315,7 +300,6 @@ static struct snd_soc_dai_ops stac9766_dai_ops_analog = { static struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, - .trigger = ac97_digital_trigger, }; struct snd_soc_dai stac9766_dai[] = { diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index a9dc5fb..da589d8 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -627,7 +627,7 @@ static int tlv320aic23_resume(struct platform_device *pdev) u16 reg; /* Sync reg_cache with the hardware */ - for (reg = 0; reg < TLV320AIC23_RESET; reg++) { + for (reg = 0; reg <= TLV320AIC23_ACTIVE; reg++) { u16 val = tlv320aic23_read_reg_cache(codec, reg); tlv320aic23_write(codec, reg, val); } diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 5f1681f..2a27f7b 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -26,7 +26,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> -#include <linux/i2c/twl4030.h> +#include <linux/i2c/twl.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -175,7 +175,7 @@ static int twl4030_write(struct snd_soc_codec *codec, { twl4030_write_reg_cache(codec, reg, value); if (likely(reg < TWL4030_REG_SW_SHADOW)) - return twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, + return twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, value, reg); else return 0; @@ -261,7 +261,7 @@ static void twl4030_power_up(struct snd_soc_codec *codec) do { /* this takes a little while, so don't slam i2c */ udelay(2000); - twl4030_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, + twl_i2c_read_u8(TWL4030_MODULE_AUDIO_VOICE, &byte, TWL4030_REG_ANAMICL); } while ((i++ < 100) && ((byte & TWL4030_CNCL_OFFSET_START) == @@ -542,7 +542,7 @@ static int pin_name##pga_event(struct snd_soc_dapm_widget *w, \ break; \ case SND_SOC_DAPM_POST_PMD: \ reg_val = twl4030_read_reg_cache(w->codec, reg); \ - twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, \ reg_val & (~mask), \ reg); \ break; \ @@ -679,7 +679,7 @@ static void headset_ramp(struct snd_soc_codec *codec, int ramp) mdelay((ramp_base[(hs_pop & TWL4030_RAMP_DELAY) >> 2] / twl4030->sysclk) + 1); /* Bypass the reg_cache to mute the headset */ - twl4030_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, + twl_i2c_write_u8(TWL4030_MODULE_AUDIO_VOICE, hs_gain & (~0x0f), TWL4030_REG_HS_GAIN_SET); diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index aa40d98..3e99fe5 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -101,7 +101,7 @@ static int uda134x_write(struct snd_soc_codec *codec, unsigned int reg, pr_debug("%s reg: %02X, value:%02X\n", __func__, reg, value); if (reg >= UDA134X_REGS_NUM) { - printk(KERN_ERR "%s unkown register: reg: %u", + printk(KERN_ERR "%s unknown register: reg: %u", __func__, reg); return -EINVAL; } @@ -552,7 +552,7 @@ static int uda134x_soc_probe(struct platform_device *pdev) ARRAY_SIZE(uda1341_snd_controls)); break; default: - printk(KERN_ERR "%s unkown codec type: %d", + printk(KERN_ERR "%s unknown codec type: %d", __func__, pd->model); return -EINVAL; } diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index f82125d..718ef91 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -925,7 +925,7 @@ static int wm8350_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) iface |= 0x3 << 8; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x3 << 8; /* lg not sure which mode */ + iface |= 0x3 << 8 | WM8350_AIF_LRCLK_INV; break; default: return -EINVAL; @@ -1340,9 +1340,10 @@ static int wm8350_resume(struct platform_device *pdev) return 0; } -static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data) +static irqreturn_t wm8350_hp_jack_handler(int irq, void *data) { struct wm8350_data *priv = data; + struct wm8350 *wm8350 = priv->codec.control_data; u16 reg; int report; int mask; @@ -1365,7 +1366,7 @@ static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data) if (!jack->jack) { dev_warn(wm8350->dev, "Jack interrupt called with no jack\n"); - return; + return IRQ_NONE; } /* Debounce */ @@ -1378,6 +1379,8 @@ static void wm8350_hp_jack_handler(struct wm8350 *wm8350, int irq, void *data) report = 0; snd_soc_jack_report(jack->jack, report, jack->report); + + return IRQ_HANDLED; } /** @@ -1421,9 +1424,7 @@ int wm8350_hp_jack_detect(struct snd_soc_codec *codec, enum wm8350_jack which, wm8350_set_bits(wm8350, WM8350_JACK_DETECT, ena); /* Sync status */ - wm8350_hp_jack_handler(wm8350, irq, priv); - - wm8350_unmask_irq(wm8350, irq); + wm8350_hp_jack_handler(irq, priv); return 0; } @@ -1482,12 +1483,16 @@ static int wm8350_probe(struct platform_device *pdev) wm8350_set_bits(wm8350, WM8350_ROUT2_VOLUME, WM8350_OUT2_VU | WM8350_OUT2R_MUTE); - wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); - wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); + /* Make sure jack detect is disabled to start off with */ + wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, + WM8350_JDL_ENA | WM8350_JDR_ENA); + wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, - wm8350_hp_jack_handler, priv); + wm8350_hp_jack_handler, 0, "Left jack detect", + priv); wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, - wm8350_hp_jack_handler, priv); + wm8350_hp_jack_handler, 0, "Right jack detect", + priv); ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) { @@ -1516,8 +1521,6 @@ static int wm8350_remove(struct platform_device *pdev) WM8350_JDL_ENA | WM8350_JDR_ENA); wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_4, WM8350_TOCLK_ENA); - wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); - wm8350_mask_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L); wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R); diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 265e68c..af8cb69 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -424,23 +424,23 @@ static int wm8510_pcm_hw_params(struct snd_pcm_substream *substream, /* filter coefficient */ switch (params_rate(params)) { - case SNDRV_PCM_RATE_8000: + case 8000: adn |= 0x5 << 1; break; - case SNDRV_PCM_RATE_11025: + case 11025: adn |= 0x4 << 1; break; - case SNDRV_PCM_RATE_16000: + case 16000: adn |= 0x3 << 1; break; - case SNDRV_PCM_RATE_22050: + case 22050: adn |= 0x2 << 1; break; - case SNDRV_PCM_RATE_32000: + case 32000: adn |= 0x1 << 1; break; - case SNDRV_PCM_RATE_44100: - case SNDRV_PCM_RATE_48000: + case 44100: + case 48000: break; } diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index c9438dd..dbc368c 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -199,7 +199,7 @@ static void wm8900_reset(struct snd_soc_codec *codec) snd_soc_write(codec, WM8900_REG_RESET, 0); memcpy(codec->reg_cache, wm8900_reg_defaults, - sizeof(codec->reg_cache)); + sizeof(wm8900_reg_defaults)); } static int wm8900_hp_event(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index b8cae17..3595bd5 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -607,7 +607,7 @@ SOC_SINGLE("Right Input PGA Common Mode Switch", WM8903_ANALOGUE_RIGHT_INPUT_1, SOC_SINGLE("DRC Switch", WM8903_DRC_0, 15, 1, 0), SOC_ENUM("DRC Compressor Slope R0", drc_slope_r0), SOC_ENUM("DRC Compressor Slope R1", drc_slope_r1), -SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8903_DRC_3, 5, 124, 1, +SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8903_DRC_3, 5, 124, 1, drc_tlv_thresh), SOC_SINGLE_TLV("DRC Volume", WM8903_DRC_3, 0, 30, 1, drc_tlv_amp), SOC_SINGLE_TLV("DRC Minimum Gain Volume", WM8903_DRC_1, 2, 3, 1, drc_tlv_min), @@ -617,11 +617,11 @@ SOC_ENUM("DRC Decay Rate", drc_decay), SOC_ENUM("DRC FF Delay", drc_ff_delay), SOC_SINGLE("DRC Anticlip Switch", WM8903_DRC_0, 1, 1, 0), SOC_SINGLE("DRC QR Switch", WM8903_DRC_0, 2, 1, 0), -SOC_SINGLE_TLV("DRC QR Threashold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max), +SOC_SINGLE_TLV("DRC QR Threshold Volume", WM8903_DRC_0, 6, 3, 0, drc_tlv_max), SOC_ENUM("DRC QR Decay Rate", drc_qr_decay), SOC_SINGLE("DRC Smoothing Switch", WM8903_DRC_0, 3, 1, 0), SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8903_DRC_0, 0, 1, 0), -SOC_ENUM("DRC Smoothing Threashold", drc_smoothing), +SOC_ENUM("DRC Smoothing Threshold", drc_smoothing), SOC_SINGLE_TLV("DRC Startup Volume", WM8903_DRC_0, 6, 18, 0, drc_tlv_startup), SOC_DOUBLE_R_TLV("Digital Capture Volume", WM8903_ADC_DIGITAL_VOLUME_LEFT, @@ -1504,7 +1504,7 @@ static int wm8903_resume(struct platform_device *pdev) struct i2c_client *i2c = codec->control_data; int i; u16 *reg_cache = codec->reg_cache; - u16 *tmp_cache = kmemdup(codec->reg_cache, sizeof(wm8903_reg_defaults), + u16 *tmp_cache = kmemdup(reg_cache, sizeof(wm8903_reg_defaults), GFP_KERNEL); /* Bring the codec back up to standby first to minimise pop/clicks */ @@ -1516,6 +1516,7 @@ static int wm8903_resume(struct platform_device *pdev) for (i = 2; i < ARRAY_SIZE(wm8903_reg_defaults); i++) if (tmp_cache[i] != reg_cache[i]) snd_soc_write(codec, i, tmp_cache[i]); + kfree(tmp_cache); } else { dev_err(&i2c->dev, "Failed to allocate temporary cache\n"); } diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 3d850b9..31e39ff 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -378,23 +378,23 @@ static int wm8940_i2s_hw_params(struct snd_pcm_substream *substream, iface |= (1 << 9); switch (params_rate(params)) { - case SNDRV_PCM_RATE_8000: + case 8000: addcntrl |= (0x5 << 1); break; - case SNDRV_PCM_RATE_11025: + case 11025: addcntrl |= (0x4 << 1); break; - case SNDRV_PCM_RATE_16000: + case 16000: addcntrl |= (0x3 << 1); break; - case SNDRV_PCM_RATE_22050: + case 22050: addcntrl |= (0x2 << 1); break; - case SNDRV_PCM_RATE_32000: + case 32000: addcntrl |= (0x1 << 1); break; - case SNDRV_PCM_RATE_44100: - case SNDRV_PCM_RATE_48000: + case 44100: + case 48000: break; } ret = snd_soc_write(codec, WM8940_ADDCNTRL, addcntrl); diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 81c57b5..8812751 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -47,7 +47,7 @@ static const u16 wm8974_reg[WM8974_CACHEREGNUM] = { }; #define WM8974_POWER1_BIASEN 0x08 -#define WM8974_POWER1_BUFIOEN 0x10 +#define WM8974_POWER1_BUFIOEN 0x04 struct wm8974_priv { struct snd_soc_codec codec; @@ -482,23 +482,23 @@ static int wm8974_pcm_hw_params(struct snd_pcm_substream *substream, /* filter coefficient */ switch (params_rate(params)) { - case SNDRV_PCM_RATE_8000: + case 8000: adn |= 0x5 << 1; break; - case SNDRV_PCM_RATE_11025: + case 11025: adn |= 0x4 << 1; break; - case SNDRV_PCM_RATE_16000: + case 16000: adn |= 0x3 << 1; break; - case SNDRV_PCM_RATE_22050: + case 22050: adn |= 0x2 << 1; break; - case SNDRV_PCM_RATE_32000: + case 32000: adn |= 0x1 << 1; break; - case SNDRV_PCM_RATE_44100: - case SNDRV_PCM_RATE_48000: + case 44100: + case 48000: break; } diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 5e32f2e..2981afa 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -689,7 +689,7 @@ SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE, SOC_SINGLE("DRC Switch", WM8993_DRC_CONTROL_1, 15, 1, 0), SOC_ENUM("DRC Path", drc_path), -SOC_SINGLE_TLV("DRC Compressor Threashold Volume", WM8993_DRC_CONTROL_2, +SOC_SINGLE_TLV("DRC Compressor Threshold Volume", WM8993_DRC_CONTROL_2, 2, 60, 1, drc_comp_threash), SOC_SINGLE_TLV("DRC Compressor Amplitude Volume", WM8993_DRC_CONTROL_3, 11, 30, 1, drc_comp_amp), @@ -709,7 +709,7 @@ SOC_SINGLE_TLV("DRC Quick Release Volume", WM8993_DRC_CONTROL_3, 2, 3, 0, SOC_ENUM("DRC Quick Release Rate", drc_qr_rate), SOC_SINGLE("DRC Smoothing Switch", WM8993_DRC_CONTROL_1, 11, 1, 0), SOC_SINGLE("DRC Smoothing Hysteresis Switch", WM8993_DRC_CONTROL_1, 8, 1, 0), -SOC_ENUM("DRC Smoothing Hysteresis Threashold", drc_smooth), +SOC_ENUM("DRC Smoothing Hysteresis Threshold", drc_smooth), SOC_SINGLE_TLV("DRC Startup Volume", WM8993_DRC_CONTROL_4, 8, 18, 0, drc_startup_tlv), diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 0ac1215..e237bf6 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -463,7 +463,8 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, { u16 *cache = codec->reg_cache; - soc_ac97_ops.write(codec->ac97, reg, val); + if (reg < 0x7c) + soc_ac97_ops.write(codec->ac97, reg, val); reg = reg >> 1; if (reg < (ARRAY_SIZE(wm9712_reg))) cache[reg] = val; diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index 3326e2a..1a5b8e0 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -55,7 +55,7 @@ static __init int efika_fabric_init(void) struct platform_device *pdev; int rc; - if (!machine_is_compatible("bplan,efika")) + if (!of_machine_is_compatible("bplan,efika")) return -ENODEV; card.platform = &mpc5200_audio_dma_platform; diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index b928ef7..6644cba 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -55,7 +55,7 @@ static __init int pcm030_fabric_init(void) struct platform_device *pdev; int rc; - if (!machine_is_compatible("phytec,pcm030")) + if (!of_machine_is_compatible("phytec,pcm030")) return -ENODEV; card.platform = &mpc5200_audio_dma_platform; diff --git a/sound/soc/imx/mx1_mx2-pcm.c b/sound/soc/imx/mx1_mx2-pcm.c index b838665..bffffcd5 100644 --- a/sound/soc/imx/mx1_mx2-pcm.c +++ b/sound/soc/imx/mx1_mx2-pcm.c @@ -322,12 +322,12 @@ static int mx1_mx2_pcm_open(struct snd_pcm_substream *substream) pr_debug("%s: Requesting dma channel (%s)\n", __func__, prtd->dma_params->name); - prtd->dma_ch = imx_dma_request_by_prio(prtd->dma_params->name, - DMA_PRIO_HIGH); - if (prtd->dma_ch < 0) { + ret = imx_dma_request_by_prio(prtd->dma_params->name, DMA_PRIO_HIGH); + if (ret < 0) { printk(KERN_ERR "Error %d requesting dma channel\n", ret); return ret; } + prtd->dma_ch = ret; imx_dma_config_burstlen(prtd->dma_ch, prtd->dma_params->watermark_level); diff --git a/sound/soc/imx/mx27vis_wm8974.c b/sound/soc/imx/mx27vis_wm8974.c index 0267d2d..07d2a24 100644 --- a/sound/soc/imx/mx27vis_wm8974.c +++ b/sound/soc/imx/mx27vis_wm8974.c @@ -180,7 +180,8 @@ static int mx27vis_hifi_hw_free(struct snd_pcm_substream *substream) struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; /* disable the PLL */ - return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, 0, 0); + return codec_dai->ops->set_pll(codec_dai, IGNORED_ARG, IGNORED_ARG, + 0, 0); } /* diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index d49458a..19283e5 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -23,9 +23,9 @@ obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o -obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o -obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o -obj-$(CONFIG_MACH_OMAP3517EVM) += snd-soc-am3517evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o +obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index 71b2c16..68980c1 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -145,6 +145,7 @@ static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { }; static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"PCM DAC", NULL, "APLL Enable"}, {"Headphone Amplifier", NULL, "PCM DAC"}, {"Line Out", NULL, "PCM DAC"}, {"Headphone Jack", NULL, "Headphone Amplifier"}, diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index c071f96..3c85c0f 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -24,7 +24,7 @@ #include <linux/clk.h> #include <linux/platform_device.h> -#include <linux/i2c/twl4030.h> +#include <linux/i2c/twl.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -321,11 +321,11 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, + twl_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux, TWL4030_INTBR_PMBR1); pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03); pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02); - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + twl_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c index 507b2ed..4984754 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.c +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c @@ -270,7 +270,7 @@ static int attach_gpio_amp(struct device *dev, gpio_direction_output(pd->amp_gain[1], 0); } - /* note, curently we assume GPA0 isn't valid amp */ + /* note, currently we assume GPA0 isn't valid amp */ if (pdata->amp_gpio > 0) { ret = gpio_request(pd->amp_gpio, "gpio-amp"); if (ret) { @@ -312,7 +312,7 @@ int simtec_audio_resume(struct device *dev) return 0; } -struct dev_pm_ops simtec_audio_pmops = { +const struct dev_pm_ops simtec_audio_pmops = { .resume = simtec_audio_resume, }; EXPORT_SYMBOL_GPL(simtec_audio_pmops); diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h index 2714203..e18faee 100644 --- a/sound/soc/s3c24xx/s3c24xx_simtec.h +++ b/sound/soc/s3c24xx/s3c24xx_simtec.h @@ -15,7 +15,7 @@ extern int simtec_audio_core_probe(struct platform_device *pdev, extern int simtec_audio_remove(struct platform_device *pdev); #ifdef CONFIG_PM -extern struct dev_pm_ops simtec_audio_pmops; +extern const struct dev_pm_ops simtec_audio_pmops; #define simtec_audio_pm &simtec_audio_pmops #else #define simtec_audio_pm NULL diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 0eb1722..1d61109 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -196,7 +196,7 @@ static int s6000_pcm_start(struct snd_pcm_substream *substream) 0 /* destination skip after chunk (impossible) */, 4 /* 16 byte burst size */, -1 /* don't conserve bandwidth */, - 0 /* low watermark irq descriptor theshold */, + 0 /* low watermark irq descriptor threshold */, 0 /* disable hardware timestamps */, 1 /* enable channel */); diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index c7af097..5263ab1 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -42,42 +42,12 @@ static struct snd_soc_device fsi_snd_devdata = { .codec_dev = &soc_codec_dev_ak4642, }; -#define AK4642_BUS 0 -#define AK4642_ADR 0x12 -static int ak4642_add_i2c_device(void) -{ - struct i2c_board_info info; - struct i2c_adapter *adapter; - struct i2c_client *client; - - memset(&info, 0, sizeof(struct i2c_board_info)); - info.addr = AK4642_ADR; - strlcpy(info.type, "ak4642", I2C_NAME_SIZE); - - adapter = i2c_get_adapter(AK4642_BUS); - if (!adapter) { - printk(KERN_DEBUG "can't get i2c adapter\n"); - return -ENODEV; - } - - client = i2c_new_device(adapter, &info); - i2c_put_adapter(adapter); - if (!client) { - printk(KERN_DEBUG "can't add i2c device\n"); - return -ENODEV; - } - - return 0; -} - static struct platform_device *fsi_snd_device; static int __init fsi_ak4642_init(void) { int ret = -ENOMEM; - ak4642_add_i2c_device(); - fsi_snd_device = platform_device_alloc("soc-audio", -1); if (!fsi_snd_device) goto out; diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 9c49c11..42813b8 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -876,7 +876,7 @@ static int fsi_probe(struct platform_device *pdev) res = platform_get_resource(pdev, IORESOURCE_MEM, 0); irq = platform_get_irq(pdev, 0); - if (!res || !irq) { + if (!res || (int)irq <= 0) { dev_err(&pdev->dev, "Not enough FSI platform resources.\n"); ret = -ENODEV; goto exit; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ef8f282..0a6440c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1236,7 +1236,7 @@ static int soc_poweroff(struct device *dev) return 0; } -static struct dev_pm_ops soc_pm_ops = { +static const struct dev_pm_ops soc_pm_ops = { .suspend = soc_suspend, .resume = soc_resume, .poweroff = soc_poweroff, |
