From da3b062e306452ffb74cf5e9e5128f9f1e0502ab Mon Sep 17 00:00:00 2001 From: Guennadi Liakhovetski Date: Thu, 18 Mar 2010 09:39:59 +0100 Subject: ASoC: SIU driver shall select FW_LOADER The SIU ASoC driver must load firmware to program the DSP, therefore it has to select FW_LOADER in its Kconfig entry. Signed-off-by: Guennadi Liakhovetski Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 1066749..f07f6d8 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -32,6 +32,7 @@ config SND_SOC_SH4_SIU select DMA_ENGINE select DMADEVICES select SH_DMAE + select FW_LOADER ## ## Boards -- cgit v1.1 From 44f497b4e0bba6ce1b73a107cc13636393344252 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 19 Mar 2010 11:10:19 +0200 Subject: ASoC: tlv320dac33: Fix DSP modes To make DSP_A mode working correctly the data delay should be configured to 0. DSP_B mode thus can not be used with DAC33, so remove it. Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index f9f367d..00d6f36 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1038,11 +1038,7 @@ static int dac33_set_dai_fmt(struct snd_soc_dai *codec_dai, case SND_SOC_DAIFMT_DSP_A: aictrl_a |= DAC33_AFMT_DSP; aictrl_b &= ~DAC33_DATA_DELAY_MASK; - aictrl_b |= DAC33_DATA_DELAY(1); /* 1 bit delay */ - break; - case SND_SOC_DAIFMT_DSP_B: - aictrl_a |= DAC33_AFMT_DSP; - aictrl_b &= ~DAC33_DATA_DELAY_MASK; /* No delay */ + aictrl_b |= DAC33_DATA_DELAY(0); break; case SND_SOC_DAIFMT_RIGHT_J: aictrl_a |= DAC33_AFMT_RIGHT_J; -- cgit v1.1 From fdb6b1e195757a66670801702e4b5fcc66ed3d72 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 19 Mar 2010 11:10:20 +0200 Subject: ASoC: tlv320dac33: Internal clocking changes During validation of the internal clocking setup it has been found that the following settings were not configured in an optimal way: ASRC_CTRL_A: SRCLKDIV was incorrect, instad of divide ratio 3, ratio of 2 has to be used (as the comment stated) DAC_CTRL_A: Fs = Fsref is the desired configuration instead of Fs = Fsref / 1.5 Signed-off-by: Peter Ujfalusi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 00d6f36..d50f169 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -778,7 +778,7 @@ static int dac33_prepare_chip(struct snd_pcm_substream *substream) if (dac33->fifo_mode) { /* Generic for all FIFO modes */ /* 50-51 : ASRC Control registers */ - dac33_write(codec, DAC33_ASRC_CTRL_A, (1 << 4)); /* div=2 */ + dac33_write(codec, DAC33_ASRC_CTRL_A, DAC33_SRCLKDIV(1)); dac33_write(codec, DAC33_ASRC_CTRL_B, 1); /* ??? */ /* Write registers 0x34 and 0x35 (MSB, LSB) */ @@ -1062,7 +1062,7 @@ static void dac33_init_chip(struct snd_soc_codec *codec) { /* 44-46: DAC Control Registers */ /* A : DAC sample rate Fsref/1.5 */ - dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(1)); + dac33_write(codec, DAC33_DAC_CTRL_A, DAC33_DACRATE(0)); /* B : DAC src=normal, not muted */ dac33_write(codec, DAC33_DAC_CTRL_B, DAC33_DACSRCR_RIGHT | DAC33_DACSRCL_LEFT); -- cgit v1.1 From 6937c947d31186750f72c9f8c942bbcc6fe63585 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Mar 2010 12:25:35 +0000 Subject: ASoC: Bail out of wm_hubs DC servo if calibration fails We're keeping track of the number of times we've iterated but never actually using this to bail out if the chip looks stuck. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 0ad9f5d..486bdd2 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -74,7 +74,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) msleep(1); reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0); dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & WM8993_DCS_DATAPATH_BUSY); + } while (reg & WM8993_DCS_DATAPATH_BUSY && count < 400); if (reg & WM8993_DCS_DATAPATH_BUSY) dev_err(codec->dev, "Timed out waiting for DC Servo\n"); -- cgit v1.1 From 8727b909bb2348d29e62c599cd7a5d610da3760f Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sun, 28 Feb 2010 10:42:38 +0800 Subject: ASoC: pxa-pcm-lib: initialize DMA channel to -1 This fixes a warning ("pxa_free_dma: trying to free channel 0 which is already freed") when a device was opened but the hw_params() call failed. Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/arm/pxa2xx-pcm-lib.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 743ac6a..fd51fa8 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -205,6 +205,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) if (!rtd->dma_desc_array) goto err1; + rtd->dma_ch = -1; runtime->private_data = rtd; return 0; -- cgit v1.1 From fc8aa7b16a5fcfe9c6d0be9bb587f1fcedd9145f Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Thu, 18 Mar 2010 07:53:11 +0100 Subject: sound/oss/vidc.c: change the field used with DMA_ACTIVE The constant DMA_ACTIVE is defined with the dma_buffparams structure rather than with the audio_operations structure. Takashi Iwai suggested that the dmap_out field of the audio_operations structure should be used instead. This is not tested. Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/oss/vidc.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index 725fef0..a4127ba 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -363,13 +363,13 @@ static void vidc_audio_trigger(int dev, int enable_bits) struct audio_operations *adev = audio_devs[dev]; if (enable_bits & PCM_ENABLE_OUTPUT) { - if (!(adev->flags & DMA_ACTIVE)) { + if (!(adev->dmap_out->flags & DMA_ACTIVE)) { unsigned long flags; local_irq_save(flags); /* prevent recusion */ - adev->flags |= DMA_ACTIVE; + adev->dmap_out->flags |= DMA_ACTIVE; dma_interrupt = vidc_audio_dma_interrupt; vidc_sound_dma_irq(0, NULL); -- cgit v1.1 From e3d2530a6cea80987f77b75d8784a00f3aaf22ff Mon Sep 17 00:00:00 2001 From: Kunal Gangakhedkar Date: Sat, 20 Mar 2010 23:08:01 +0530 Subject: ALSA: hda - Add PCI quirk for HP dv6-1110ax. Adding this PCI quirk fixes the board config detection. This also fixes jack sensing by using "hp_detect=1" via properly detected board config. Signed-off-by: Kunal Gangakhedkar Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c416bb..c4be3fa 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1730,6 +1730,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP HDX", STAC_HP_HDX), /* HDX16 */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620, "HP dv6", STAC_HP_DV5), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x3061, + "HP dv6", STAC_HP_DV5), /* HP dv6-1110ax */ SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, -- cgit v1.1 From 025f206c9e0f96cc41567b01c07fb852d8900da1 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 21 Mar 2010 18:34:43 -0400 Subject: ALSA: hda: Fix 0 dB offset for HP laptops using CX20551 (Waikiki) BugLink: https://launchpad.net/bugs/420578 The OR has verified that his hardware distorts because of the 0 dB offset not corresponding to the highest PCM level. Fix this by capping said PCM level to 0 dB similarly to what we do for CX20549 (Venice). Reported-by: Mike Pontillo Tested-by: Mike Pontillo Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 194a28c..61682e1 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1591,6 +1591,21 @@ static int patch_cxt5047(struct hda_codec *codec) #endif } spec->vmaster_nid = 0x13; + + switch (codec->subsystem_id >> 16) { + case 0x103c: + /* HP laptops have really bad sound over 0 dB on NID 0x10. + * Fix max PCM level to 0 dB (originally it has 0x1e steps + * with 0 dB offset 0x17) + */ + snd_hda_override_amp_caps(codec, 0x10, 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; + } + return 0; } -- cgit v1.1 From e933e9e5238b79870b04718024416a6dcf602a27 Mon Sep 17 00:00:00 2001 From: Derek Kelly Date: Mon, 22 Mar 2010 08:04:19 +0100 Subject: ALSA: hda - Add support of Nvidia GT220 HDMI This patch adds the device id for Nvidia GT220 cards to the nvhdmi driver. I have tested it and confirmed it to be working. Original patch download link: https://gist.github.com/324070/ Signed-off-by: Derek Kelly Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 70669a2..9e47717 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -554,6 +554,8 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000b, .name = "GT21x HDMI", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de000a, .name = "GT220 HDMI", + .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000d, .name = "GT240 HDMI", .patch = patch_nvhdmi_8ch_89 }, {} /* terminator */ @@ -568,6 +570,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:10de000c"); MODULE_ALIAS("snd-hda-codec-id:10de000b"); +MODULE_ALIAS("snd-hda-codec-id:10de000a"); MODULE_ALIAS("snd-hda-codec-id:10de000d"); MODULE_LICENSE("GPL"); -- cgit v1.1 From ea823c08912cfb6d4af2fa8b6dd5d8deb2fb486a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 Mar 2010 08:07:55 +0100 Subject: ALSA: hda - Sort codec entry list of Nvidia HDMI Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_nvhdmi.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_nvhdmi.c b/sound/pci/hda/patch_nvhdmi.c index 9e47717..3c10c0b 100644 --- a/sound/pci/hda/patch_nvhdmi.c +++ b/sound/pci/hda/patch_nvhdmi.c @@ -538,8 +538,6 @@ static int patch_nvhdmi_2ch(struct hda_codec *codec) * patch entries */ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { - { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, - { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0003, .name = "MCP77/78 HDMI", @@ -550,14 +548,16 @@ static struct hda_codec_preset snd_hda_preset_nvhdmi[] = { .patch = patch_nvhdmi_8ch_7x }, { .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x }, - { .id = 0x10de000c, .name = "MCP89 HDMI", + { .id = 0x10de000a, .name = "GT220 HDMI", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000b, .name = "GT21x HDMI", .patch = patch_nvhdmi_8ch_89 }, - { .id = 0x10de000a, .name = "GT220 HDMI", + { .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi_8ch_89 }, { .id = 0x10de000d, .name = "GT240 HDMI", .patch = patch_nvhdmi_8ch_89 }, + { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, + { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, {} /* terminator */ }; @@ -566,12 +566,12 @@ MODULE_ALIAS("snd-hda-codec-id:10de0003"); MODULE_ALIAS("snd-hda-codec-id:10de0005"); 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"); -MODULE_ALIAS("snd-hda-codec-id:10de000c"); -MODULE_ALIAS("snd-hda-codec-id:10de000b"); MODULE_ALIAS("snd-hda-codec-id:10de000a"); +MODULE_ALIAS("snd-hda-codec-id:10de000b"); +MODULE_ALIAS("snd-hda-codec-id:10de000c"); MODULE_ALIAS("snd-hda-codec-id:10de000d"); +MODULE_ALIAS("snd-hda-codec-id:10de0067"); +MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("NVIDIA HDMI HD-audio codec"); -- cgit v1.1 From bae84e70d66fe46c12231082cf1c4848ea22f3ef Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 22 Mar 2010 08:30:20 +0100 Subject: ALSA: hda - Fix access-after-free in patch_realtek.c alc_free_kctls() has to be called after all jobs done in alc_build_controls(). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4ec5763..053d53d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2532,8 +2532,6 @@ static int alc_build_controls(struct hda_codec *codec) return err; } - alc_free_kctls(codec); /* no longer needed */ - /* assign Capture Source enums to NID */ kctl = snd_hda_find_mixer_ctl(codec, "Capture Source"); if (!kctl) @@ -2602,6 +2600,9 @@ static int alc_build_controls(struct hda_codec *codec) } } } + + alc_free_kctls(codec); /* no longer needed */ + return 0; } -- cgit v1.1 From 3cc4e53f86dab635166929bfa47cc68d59b28c26 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 12 Feb 2010 14:39:36 +0000 Subject: ASoC: Remove BROKEN from i.MX audio after dependencies merged Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/imx/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index c7d0fd9..7174b4c 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,6 +1,6 @@ config SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC && BROKEN + depends on ARCH_MXC select SND_PCM select FIQ select SND_SOC_AC97_BUS -- cgit v1.1 From 1c583063a5c769fe2ec604752e383972c69e6d9b Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Wed, 24 Mar 2010 07:10:54 +0100 Subject: ALSA: cmipci: work around invalid PCM pointer When the CMI8738 FRAME2 register is read, the chip sometimes (probably when wrapping around) returns an invalid value that would be outside the programmed DMA buffer. This leads to an inconsistent PCM pointer that is likely to result in an underrun. To work around this, read the register multiple times until we get a valid value; the error state seems to be very short-lived. Signed-off-by: Clemens Ladisch Reported-and-tested-by: Matija Nalis Cc: Signed-off-by: Takashi Iwai --- sound/pci/cmipci.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index 1ded64e..329968e 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -941,13 +941,21 @@ static snd_pcm_uframes_t snd_cmipci_pcm_pointer(struct cmipci *cm, struct cmipci struct snd_pcm_substream *substream) { size_t ptr; - unsigned int reg; + unsigned int reg, rem, tries; + if (!rec->running) return 0; #if 1 // this seems better.. reg = rec->ch ? CM_REG_CH1_FRAME2 : CM_REG_CH0_FRAME2; - ptr = rec->dma_size - (snd_cmipci_read_w(cm, reg) + 1); - ptr >>= rec->shift; + for (tries = 0; tries < 3; tries++) { + rem = snd_cmipci_read_w(cm, reg); + if (rem < rec->dma_size) + goto ok; + } + printk(KERN_ERR "cmipci: invalid PCM pointer: %#x\n", rem); + return SNDRV_PCM_POS_XRUN; +ok: + ptr = (rec->dma_size - (rem + 1)) >> rec->shift; #else reg = rec->ch ? CM_REG_CH1_FRAME1 : CM_REG_CH0_FRAME1; ptr = snd_cmipci_read(cm, reg) - rec->offset; -- cgit v1.1 From a8462bde78fdb77c8ede61e1af99617905a78ccf Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 24 Mar 2010 14:58:34 +0300 Subject: ASoC: wm8994: playback => capture Sparse caught that initialize "playback" two times instead of initializing "capture". Signed-off-by: Dan Carpenter Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 29f3771..d10d651 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3401,7 +3401,7 @@ struct snd_soc_dai wm8994_dai[] = { .rates = WM8994_RATES, .formats = WM8994_FORMATS, }, - .playback = { + .capture = { .stream_name = "AIF3 Capture", .channels_min = 2, .channels_max = 2, -- cgit v1.1 From 6a4f2ccb467e00281470cde2dee08fe5ecde62d1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 25 Mar 2010 15:00:15 +0100 Subject: ALSA: hda - Don't set invalid connection index in Realtek initialiaiton Skip initialization of connections of DAC widgets that aren't used, which resulted in invalid verb parameters. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 053d53d..9a23444 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10043,8 +10043,11 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, alc_set_pin_output(codec, nid, pin_type); if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; - else + else { + if (spec->multiout.num_dacs >= dac_idx) + return; idx = spec->multiout.dac_nids[dac_idx] - 2; + } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } -- cgit v1.1 From e1f7f02b45cf33a774d56e505ce1718af9392f5e Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Thu, 25 Mar 2010 22:38:15 -0700 Subject: ALSA: ac97: Add IBM ThinkPad R40e to Headphone/Line Jack Sense blacklist BugLink: https://launchpad.net/bugs/303789 This model needs both 'Headphone Jack Sense' and 'Line Jack Sense' muted for audible audio, so just add its SSID to the blacklist and don't enumerate the controls. Signed-off-by: Daniel T Chen Cc: Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 1caf5e3..1a59b71 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1852,6 +1852,7 @@ static unsigned int ad1981_jacks_blacklist[] = { 0x10140523, /* Thinkpad R40 */ 0x10140534, /* Thinkpad X31 */ 0x10140537, /* Thinkpad T41p */ + 0x1014053e, /* Thinkpad R40e */ 0x10140554, /* Thinkpad T42p/R50p */ 0x10140567, /* Thinkpad T43p 2668-G7U */ 0x10140581, /* Thinkpad X41-2527 */ -- cgit v1.1 From 0f17014b340b98465fcf0de4c0d6c84a002ec53b Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 26 Mar 2010 16:07:25 +0200 Subject: ALSA: pcm_lib - fix xrun functionality The commit 4d96eb255c53ab5e39b37fd4d484ea3dc39ab456 broke the interrupt time xrun functionality (stream stop etc.) if the CONFIG_SND_PCM_XRUN_DEBUG is not set. This is because the xrun() is null defined without it. Fix this by letting the function xrun() to be always defined as it was before. Signed-off-by: Jarkko Nikula Cc: Jaroslav Kysela Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index b546ac2..a2ff861 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -148,6 +148,9 @@ static void pcm_debug_name(struct snd_pcm_substream *substream, #define xrun_debug(substream, mask) \ ((substream)->pstr->xrun_debug & (mask)) +#else +#define xrun_debug(substream, mask) 0 +#endif #define dump_stack_on_xrun(substream) do { \ if (xrun_debug(substream, XRUN_DEBUG_STACK)) \ @@ -169,6 +172,7 @@ static void xrun(struct snd_pcm_substream *substream) } } +#ifdef CONFIG_SND_PCM_XRUN_DEBUG #define hw_ptr_error(substream, fmt, args...) \ do { \ if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { \ @@ -255,8 +259,6 @@ static void xrun_log_show(struct snd_pcm_substream *substream) #else /* ! CONFIG_SND_PCM_XRUN_DEBUG */ -#define xrun_debug(substream, mask) 0 -#define xrun(substream) do { } while (0) #define hw_ptr_error(substream, fmt, args...) do { } while (0) #define xrun_log(substream, pos) do { } while (0) #define xrun_log_show(substream) do { } while (0) -- cgit v1.1 From 5cd165e7057020884e430941c24454d3df9a799d Mon Sep 17 00:00:00 2001 From: Daniel Chen Date: Sun, 28 Mar 2010 13:32:34 -0700 Subject: ALSA: ac97: Add Toshiba P500 to ac97 jack sense blacklist BugLink: https://launchpad.net/bugs/481058 The OR has verified that both 'Headphone Jack Sense' and 'Line Jack Sense' need to be muted for sound to be audible, so just add the machine's SSID to the ac97 jack sense blacklist. Reported-by: Richard Gagne Tested-by: Richard Gagne Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/ac97/ac97_patch.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index 1a59b71..e68c98e 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -1859,6 +1859,7 @@ static unsigned int ad1981_jacks_blacklist[] = { 0x10280160, /* Dell Dimension 2400 */ 0x104380b0, /* Asus A7V8X-MX */ 0x11790241, /* Toshiba Satellite A-15 S127 */ + 0x1179ff10, /* Toshiba P500 */ 0x144dc01a, /* Samsung NP-X20C004/SEG */ 0 /* end */ }; -- cgit v1.1 From 9ec8ddad59fadd8021adfea4cb716a49b0e232e9 Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Sun, 28 Mar 2010 02:34:40 -0400 Subject: ALSA: hda: Use LPIB for ga-ma770-ud3 board BugLink: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=575669 The OR states that position_fix=1 is necessary to work around glitching during volume adjustments using PulseAudio. Reported-by: Carlos Laviola Tested-by: Carlos Laviola Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 8b29156..4bb9067 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2269,6 +2269,7 @@ static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x103c, 0x306d, "HP dv3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1106, 0x3288, "ASUS M2V-MX SE", POS_FIX_LPIB), SND_PCI_QUIRK(0x1043, 0x813d, "ASUS P5AD2", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1458, 0xa022, "ga-ma770-ud3", POS_FIX_LPIB), SND_PCI_QUIRK(0x1462, 0x1002, "MSI Wind U115", POS_FIX_LPIB), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar Microtech", POS_FIX_LPIB), SND_PCI_QUIRK(0x8086, 0xd601, "eMachines T5212", POS_FIX_LPIB), -- cgit v1.1 From 5dbd5ec6e1cf2e49128025d80813a275744a7ac5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Mar 2010 09:16:24 +0200 Subject: ALSA: hda - Fix invalid bit values passed to snd_hda_codec_amp_stereo() The mask and value parameters passed to snd_hda_codec_amp_stereo() should be 8-bit values for mute and volume. Passing AMP_IN_MUTE() is wrong, which is found in many places in patch_realtek.c as a left-over from the conversion to snd_hda_codec_amp_stereo(). Reported-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 52 +++++++++++++++++++++---------------------- 1 file changed, 26 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9a23444..bc55c1e 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12459,11 +12459,11 @@ static void alc268_aspire_one_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, 0x15); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0f, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc268_acer_lc_unsol_event(struct hda_codec *codec, @@ -13482,11 +13482,11 @@ static void alc269_quanta_fl1_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, 0x15); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); @@ -13511,11 +13511,11 @@ static void alc269_lifebook_speaker_automute(struct hda_codec *codec) /* Check port replicator headphone socket */ present |= snd_hda_jack_detect(codec, 0x1a); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_write(codec, 0x20, 0, AC_VERB_SET_COEF_INDEX, 0x0c); @@ -13646,11 +13646,11 @@ static void alc269_speaker_automute(struct hda_codec *codec) unsigned char bits; present = snd_hda_jack_detect(codec, nid); - bits = present ? AMP_IN_MUTE(0) : 0; + bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } /* unsolicited event for HP jack sensing */ @@ -17115,9 +17115,9 @@ static void alc663_m51va_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) @@ -17128,13 +17128,13 @@ static void alc663_21jd_two_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x21); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) @@ -17145,13 +17145,13 @@ static void alc663_15jd_two_speaker_automute(struct hda_codec *codec) present = snd_hda_jack_detect(codec, 0x15); bits = present ? HDA_AMP_MUTE : 0; snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 0, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); snd_hda_codec_amp_stereo(codec, 0x0e, HDA_INPUT, 1, - AMP_IN_MUTE(0), bits); + HDA_AMP_MUTE, bits); } static void alc662_f5z_speaker_automute(struct hda_codec *codec) @@ -17190,14 +17190,14 @@ static void alc663_two_hp_m2_speaker_automute(struct hda_codec *codec) if (present1 || present2) { snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), AMP_IN_MUTE(0)); + HDA_AMP_MUTE, HDA_AMP_MUTE); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), AMP_IN_MUTE(0)); + HDA_AMP_MUTE, HDA_AMP_MUTE); } else { snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 0, - AMP_IN_MUTE(0), 0); + HDA_AMP_MUTE, 0); snd_hda_codec_amp_stereo(codec, 0x0c, HDA_INPUT, 1, - AMP_IN_MUTE(0), 0); + HDA_AMP_MUTE, 0); } } -- cgit v1.1 From 6694635d3ae1b038d7a0e38b80637db867c7c8e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 29 Mar 2010 17:21:45 +0200 Subject: ALSA: hda - Fix ADC/MUX assignment of ALC269 codec ALC269 codec has a few different variants, and each of them may have different ADC and MUX widgets. For example, one model has ADC 0x08 with MUX 0x23 while others has ADC 0x09 or ADC 0x07 with MUX 022 or 0x24. The difference of ADC appears usually as the capability of the digital mic pin (0x12), and the current driver sometimes misses the internal mic pin due to the mismatching ADC. This patch adds a bit more clever way to find the matching ADC instead of the static list. Now the driver checks all active input pins and fills only the ADC/MUX's that contain all of them. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 95 ++++++++++++++++++++++++++++++++++++------- 1 file changed, 80 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bc55c1e..22aea7b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4984,6 +4984,69 @@ static void set_capture_mixer(struct hda_codec *codec) } } +/* fill adc_nids (and capsrc_nids) containing all active input pins */ +static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, + int num_nids) +{ + struct alc_spec *spec = codec->spec; + int n; + hda_nid_t fallback_adc = 0, fallback_cap = 0; + + for (n = 0; n < num_nids; n++) { + hda_nid_t adc, cap; + hda_nid_t conn[HDA_MAX_NUM_INPUTS]; + int nconns, i, j; + + adc = nids[n]; + if (get_wcaps_type(get_wcaps(codec, adc)) != AC_WID_AUD_IN) + continue; + cap = adc; + nconns = snd_hda_get_connections(codec, cap, conn, + ARRAY_SIZE(conn)); + if (nconns == 1) { + cap = conn[0]; + nconns = snd_hda_get_connections(codec, cap, conn, + ARRAY_SIZE(conn)); + } + if (nconns <= 0) + continue; + if (!fallback_adc) { + fallback_adc = adc; + fallback_cap = cap; + } + for (i = 0; i < AUTO_PIN_LAST; i++) { + hda_nid_t nid = spec->autocfg.input_pins[i]; + if (!nid) + continue; + for (j = 0; j < nconns; j++) { + if (conn[j] == nid) + break; + } + if (j >= nconns) + break; + } + if (i >= AUTO_PIN_LAST) { + int num_adcs = spec->num_adc_nids; + spec->private_adc_nids[num_adcs] = adc; + spec->private_capsrc_nids[num_adcs] = cap; + spec->num_adc_nids++; + spec->adc_nids = spec->private_adc_nids; + if (adc != cap) + spec->capsrc_nids = spec->private_capsrc_nids; + } + } + if (!spec->num_adc_nids) { + printk(KERN_WARNING "hda_codec: %s: no valid ADC found;" + " using fallback 0x%x\n", fallback_adc); + spec->private_adc_nids[0] = fallback_adc; + spec->adc_nids = spec->private_adc_nids; + if (fallback_adc != fallback_cap) { + spec->private_capsrc_nids[0] = fallback_cap; + spec->capsrc_nids = spec->private_adc_nids; + } + } +} + #ifdef CONFIG_SND_HDA_INPUT_BEEP #define set_beep_amp(spec, nid, idx, dir) \ ((spec)->beep_amp = HDA_COMPOSE_AMP_VAL(nid, 3, idx, dir)) @@ -13333,9 +13396,9 @@ static hda_nid_t alc269vb_capsrc_nids[1] = { 0x22, }; -/* NOTE: ADC2 (0x07) is connected from a recording *MIXER* (0x24), - * not a mux! - */ +static hda_nid_t alc269_adc_candidates[] = { + 0x08, 0x09, 0x07, +}; #define alc269_modes alc260_modes #define alc269_capture_source alc880_lg_lw_capture_source @@ -13842,7 +13905,6 @@ static int alc269_parse_auto_config(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int err; static hda_nid_t alc269_ignore[] = { 0x1d, 0 }; - hda_nid_t real_capsrc_nids; err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, alc269_ignore); @@ -13866,18 +13928,19 @@ static int alc269_parse_auto_config(struct hda_codec *codec) if ((alc_read_coef_idx(codec, 0) & 0x00f0) == 0x0010) { add_verb(spec, alc269vb_init_verbs); - real_capsrc_nids = alc269vb_capsrc_nids[0]; alc_ssid_check(codec, 0, 0x1b, 0x14, 0x21); } else { add_verb(spec, alc269_init_verbs); - real_capsrc_nids = alc269_capsrc_nids[0]; alc_ssid_check(codec, 0x15, 0x1b, 0x14, 0); } spec->num_mux_defs = 1; spec->input_mux = &spec->private_imux[0]; + fillup_priv_adc_nids(codec, alc269_adc_candidates, + sizeof(alc269_adc_candidates)); + /* set default input source */ - snd_hda_codec_write_cache(codec, real_capsrc_nids, + snd_hda_codec_write_cache(codec, spec->capsrc_nids[0], 0, AC_VERB_SET_CONNECT_SEL, spec->input_mux->items[0].index); @@ -14156,14 +14219,16 @@ static int patch_alc269(struct hda_codec *codec) spec->stream_digital_playback = &alc269_pcm_digital_playback; spec->stream_digital_capture = &alc269_pcm_digital_capture; - if (!is_alc269vb) { - spec->adc_nids = alc269_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); - spec->capsrc_nids = alc269_capsrc_nids; - } else { - spec->adc_nids = alc269vb_adc_nids; - spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids); - spec->capsrc_nids = alc269vb_capsrc_nids; + if (!spec->adc_nids) { /* wasn't filled automatically? use default */ + if (!is_alc269vb) { + spec->adc_nids = alc269_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269_adc_nids); + spec->capsrc_nids = alc269_capsrc_nids; + } else { + spec->adc_nids = alc269vb_adc_nids; + spec->num_adc_nids = ARRAY_SIZE(alc269vb_adc_nids); + spec->capsrc_nids = alc269vb_capsrc_nids; + } } if (!spec->cap_mixer) -- cgit v1.1 From fb48e3c6a4d8888aff61fbf567aadac7d206e973 Mon Sep 17 00:00:00 2001 From: Graham Gower Date: Thu, 25 Mar 2010 10:52:12 +1030 Subject: ASoC: Fix passing platform_data to ac97 bus users and fix a leak [The issue is an attempt to write the pdata without the AC97 device allocated when using ac97.c - also added a comment in soc-core.c for the special case for ac97. -- broonie] Signed-off-by: Graham Gower Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/codecs/ac97.c | 15 +++++++++------ sound/soc/soc-core.c | 3 ++- 2 files changed, 11 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index a1bbe16..bcfa532 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -80,9 +80,11 @@ static int ac97_write(struct snd_soc_codec *codec, unsigned int reg, static int ac97_soc_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_card *card = socdev->card; struct snd_soc_codec *codec; struct snd_ac97_bus *ac97_bus; struct snd_ac97_template ac97_template; + int i; int ret = 0; printk(KERN_INFO "AC97 SoC Audio Codec %s\n", AC97_VERSION); @@ -102,12 +104,6 @@ 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) @@ -123,6 +119,13 @@ static int ac97_soc_probe(struct platform_device *pdev) if (ret < 0) goto bus_err; + for (i = 0; i < card->num_links; i++) { + if (card->dai_link[i].codec_dai->ac97_control) { + snd_ac97_dev_add_pdata(codec->ac97, + card->dai_link[i].cpu_dai->ac97_pdata); + } + } + return 0; bus_err: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8b0556..d0efd5e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1548,7 +1548,8 @@ int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid) mutex_unlock(&codec->mutex); return ret; } - if (card->dai_link[i].codec_dai->ac97_control) { + /* Check for codec->ac97 to handle the ac97.c fun */ + if (card->dai_link[i].codec_dai->ac97_control && codec->ac97) { snd_ac97_dev_add_pdata(codec->ac97, card->dai_link[i].cpu_dai->ac97_pdata); } -- cgit v1.1 From 1f85d72d2c9c9a1d6d32cf325936bc224ad5d591 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 30 Mar 2010 07:48:05 +0200 Subject: ALSA: hda - Add missing printk argument in previous patch Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 22aea7b..ca93c4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5037,7 +5037,8 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, } if (!spec->num_adc_nids) { printk(KERN_WARNING "hda_codec: %s: no valid ADC found;" - " using fallback 0x%x\n", fallback_adc); + " using fallback 0x%x\n", + codec->chip_name, fallback_adc); spec->private_adc_nids[0] = fallback_adc; spec->adc_nids = spec->private_adc_nids; if (fallback_adc != fallback_cap) { -- cgit v1.1 From 5a0e3ad6af8660be21ca98a971cd00f331318c05 Mon Sep 17 00:00:00 2001 From: Tejun Heo Date: Wed, 24 Mar 2010 17:04:11 +0900 Subject: include cleanup: Update gfp.h and slab.h includes to prepare for breaking implicit slab.h inclusion from percpu.h percpu.h is included by sched.h and module.h and thus ends up being included when building most .c files. percpu.h includes slab.h which in turn includes gfp.h making everything defined by the two files universally available and complicating inclusion dependencies. percpu.h -> slab.h dependency is about to be removed. Prepare for this change by updating users of gfp and slab facilities include those headers directly instead of assuming availability. As this conversion needs to touch large number of source files, the following script is used as the basis of conversion. http://userweb.kernel.org/~tj/misc/slabh-sweep.py The script does the followings. * Scan files for gfp and slab usages and update includes such that only the necessary includes are there. ie. if only gfp is used, gfp.h, if slab is used, slab.h. * When the script inserts a new include, it looks at the include blocks and try to put the new include such that its order conforms to its surrounding. It's put in the include block which contains core kernel includes, in the same order that the rest are ordered - alphabetical, Christmas tree, rev-Xmas-tree or at the end if there doesn't seem to be any matching order. * If the script can't find a place to put a new include (mostly because the file doesn't have fitting include block), it prints out an error message indicating which .h file needs to be added to the file. The conversion was done in the following steps. 1. The initial automatic conversion of all .c files updated slightly over 4000 files, deleting around 700 includes and adding ~480 gfp.h and ~3000 slab.h inclusions. The script emitted errors for ~400 files. 2. Each error was manually checked. Some didn't need the inclusion, some needed manual addition while adding it to implementation .h or embedding .c file was more appropriate for others. This step added inclusions to around 150 files. 3. The script was run again and the output was compared to the edits from #2 to make sure no file was left behind. 4. Several build tests were done and a couple of problems were fixed. e.g. lib/decompress_*.c used malloc/free() wrappers around slab APIs requiring slab.h to be added manually. 5. The script was run on all .h files but without automatically editing them as sprinkling gfp.h and slab.h inclusions around .h files could easily lead to inclusion dependency hell. Most gfp.h inclusion directives were ignored as stuff from gfp.h was usually wildly available and often used in preprocessor macros. Each slab.h inclusion directive was examined and added manually as necessary. 6. percpu.h was updated not to include slab.h. 7. Build test were done on the following configurations and failures were fixed. CONFIG_GCOV_KERNEL was turned off for all tests (as my distributed build env didn't work with gcov compiles) and a few more options had to be turned off depending on archs to make things build (like ipr on powerpc/64 which failed due to missing writeq). * x86 and x86_64 UP and SMP allmodconfig and a custom test config. * powerpc and powerpc64 SMP allmodconfig * sparc and sparc64 SMP allmodconfig * ia64 SMP allmodconfig * s390 SMP allmodconfig * alpha SMP allmodconfig * um on x86_64 SMP allmodconfig 8. percpu.h modifications were reverted so that it could be applied as a separate patch and serve as bisection point. Given the fact that I had only a couple of failures from tests on step 6, I'm fairly confident about the coverage of this conversion patch. If there is a breakage, it's likely to be something in one of the arch headers which should be easily discoverable easily on most builds of the specific arch. Signed-off-by: Tejun Heo Guess-its-ok-by: Christoph Lameter Cc: Ingo Molnar Cc: Lee Schermerhorn --- sound/aoa/codecs/onyx.c | 1 + sound/aoa/codecs/tas.c | 1 + sound/aoa/codecs/toonie.c | 1 + sound/aoa/core/gpio-pmf.c | 1 + sound/aoa/fabrics/layout.c | 1 + sound/aoa/soundbus/i2sbus/control.c | 1 + sound/aoa/soundbus/i2sbus/core.c | 1 + sound/aoa/soundbus/i2sbus/pcm.c | 1 + sound/arm/pxa2xx-pcm-lib.c | 1 + sound/core/control_compat.c | 1 + sound/core/hrtimer.c | 1 + sound/core/info.c | 1 + sound/core/jack.c | 1 + sound/core/misc.c | 1 + sound/core/oss/route.c | 1 - sound/core/pcm_compat.c | 1 + sound/core/pcm_memory.c | 1 + sound/core/seq/oss/seq_oss_init.c | 1 + sound/core/seq/oss/seq_oss_midi.c | 1 + sound/core/seq/oss/seq_oss_readq.c | 1 + sound/core/seq/oss/seq_oss_synth.c | 1 + sound/core/seq/oss/seq_oss_timer.c | 1 + sound/core/seq/oss/seq_oss_writeq.c | 1 + sound/core/seq/seq_compat.c | 1 + sound/core/seq/seq_system.c | 1 + sound/drivers/ml403-ac97cr.c | 1 + sound/drivers/mtpav.c | 1 - sound/drivers/mts64.c | 1 + sound/drivers/opl3/opl3_oss.c | 1 - sound/drivers/opl3/opl3_synth.c | 1 + sound/drivers/opl4/opl4_lib.c | 1 + sound/drivers/pcsp/pcsp_lib.c | 1 + sound/drivers/portman2x4.c | 1 + sound/drivers/vx/vx_hwdep.c | 1 + sound/i2c/other/tea575x-tuner.c | 1 + sound/isa/cmi8330.c | 1 - sound/isa/cs423x/cs4236.c | 1 - sound/isa/es18xx.c | 1 - sound/isa/gus/interwave.c | 1 - sound/isa/msnd/msnd_midi.c | 1 + sound/isa/opl3sa2.c | 1 - sound/isa/opti9xx/miro.c | 1 - sound/isa/opti9xx/opti92x-ad1848.c | 1 - sound/isa/sb/emu8000_pcm.c | 1 + sound/isa/sb/sb16.c | 1 - sound/isa/sb/sb8.c | 1 - sound/isa/wavefront/wavefront.c | 1 - sound/isa/wavefront/wavefront_fx.c | 1 + sound/isa/wavefront/wavefront_synth.c | 1 + sound/mips/hal2.c | 1 + sound/mips/sgio2audio.c | 2 +- sound/oss/ad1848.c | 1 + sound/oss/dmabuf.c | 1 + sound/oss/kahlua.c | 1 + sound/oss/mpu401.c | 1 + sound/oss/msnd.c | 1 - sound/oss/msnd_pinnacle.c | 2 +- sound/oss/opl3.c | 1 + sound/oss/sb_card.c | 1 + sound/oss/sb_common.c | 1 + sound/oss/sb_midi.c | 1 + sound/oss/sb_mixer.c | 2 ++ sound/oss/soundcard.c | 1 - sound/oss/uart401.c | 1 + sound/oss/v_midi.c | 1 + sound/oss/vidc.c | 1 + sound/oss/vwsnd.c | 1 + sound/oss/waveartist.c | 1 + sound/pci/ac97/ac97_proc.c | 1 - sound/pci/als4000.c | 1 - sound/pci/aw2/aw2-saa7146.c | 1 - sound/pci/ca0106/ca0106_mixer.c | 1 - sound/pci/ca0106/ca0106_proc.c | 1 - sound/pci/cs5530.c | 1 + sound/pci/cs5535audio/cs5535audio_pcm.c | 1 - sound/pci/cs5535audio/cs5535audio_pm.c | 1 - sound/pci/ctxfi/ctatc.c | 1 + sound/pci/ctxfi/ctpcm.c | 1 + sound/pci/echoaudio/darla20.c | 2 +- sound/pci/echoaudio/darla24.c | 2 +- sound/pci/echoaudio/echo3g.c | 2 +- sound/pci/echoaudio/gina20.c | 2 +- sound/pci/echoaudio/gina24.c | 2 +- sound/pci/echoaudio/indigo.c | 2 +- sound/pci/echoaudio/indigodj.c | 2 +- sound/pci/echoaudio/indigodjx.c | 2 +- sound/pci/echoaudio/indigoio.c | 2 +- sound/pci/echoaudio/indigoiox.c | 2 +- sound/pci/echoaudio/layla20.c | 2 +- sound/pci/echoaudio/layla24.c | 2 +- sound/pci/echoaudio/mia.c | 2 +- sound/pci/echoaudio/mona.c | 2 +- sound/pci/emu10k1/memory.c | 1 + sound/pci/hda/hda_beep.c | 1 + sound/pci/hda/hda_eld.c | 1 + sound/pci/ice1712/ak4xxx.c | 1 + sound/pci/ice1712/amp.c | 1 - sound/pci/ice1712/vt1720_mobo.c | 1 - sound/pci/ice1712/wtm.c | 1 - sound/pci/lx6464es/lx6464es.c | 1 + sound/pci/mixart/mixart.c | 1 + sound/pci/mixart/mixart_hwdep.c | 1 + sound/pci/oxygen/oxygen_lib.c | 1 + sound/pci/rme32.c | 2 +- sound/pci/rme96.c | 1 - sound/pci/rme9652/hdsp.c | 1 - sound/pci/rme9652/rme9652.c | 1 - sound/pci/sis7019.c | 1 + sound/pcmcia/pdaudiocf/pdaudiocf_core.c | 1 + sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c | 1 - sound/pcmcia/vx/vxpocket.c | 1 + sound/ppc/burgundy.c | 1 - sound/ppc/keywest.c | 1 - sound/ppc/snd_ps3.c | 2 +- sound/sh/sh_dac_audio.c | 1 + sound/soc/au1x/psc-ac97.c | 1 + sound/soc/au1x/psc-i2s.c | 1 + sound/soc/blackfin/bf5xx-ac97-pcm.c | 2 +- sound/soc/blackfin/bf5xx-ac97.c | 1 + sound/soc/blackfin/bf5xx-i2s-pcm.c | 2 +- sound/soc/blackfin/bf5xx-tdm-pcm.c | 2 +- sound/soc/codecs/ac97.c | 1 + sound/soc/codecs/ad1836.c | 1 + sound/soc/codecs/ad1938.c | 1 + sound/soc/codecs/ad1980.c | 1 + sound/soc/codecs/ad73311.c | 1 + sound/soc/codecs/ads117x.c | 1 + sound/soc/codecs/ak4104.c | 1 + sound/soc/codecs/ak4535.c | 1 + sound/soc/codecs/ak4642.c | 1 + sound/soc/codecs/ak4671.c | 1 + sound/soc/codecs/cs4270.c | 1 + sound/soc/codecs/cx20442.c | 1 + sound/soc/codecs/da7210.c | 1 + sound/soc/codecs/pcm3008.c | 1 + sound/soc/codecs/ssm2602.c | 1 + sound/soc/codecs/stac9766.c | 1 + sound/soc/codecs/tlv320aic23.c | 1 + sound/soc/codecs/tlv320aic26.c | 1 + sound/soc/codecs/tlv320aic3x.c | 1 + sound/soc/codecs/tlv320dac33.c | 1 + sound/soc/codecs/tpa6130a2.c | 1 + sound/soc/codecs/twl4030.c | 1 + sound/soc/codecs/uda134x.c | 1 + sound/soc/codecs/wm2000.c | 1 + sound/soc/codecs/wm8350.c | 1 + sound/soc/codecs/wm8400.c | 1 + sound/soc/codecs/wm8510.c | 1 + sound/soc/codecs/wm8523.c | 1 + sound/soc/codecs/wm8580.c | 1 + sound/soc/codecs/wm8711.c | 1 + sound/soc/codecs/wm8727.c | 1 + sound/soc/codecs/wm8728.c | 1 + sound/soc/codecs/wm8731.c | 1 + sound/soc/codecs/wm8750.c | 1 + sound/soc/codecs/wm8753.c | 1 + sound/soc/codecs/wm8776.c | 1 + sound/soc/codecs/wm8900.c | 1 + sound/soc/codecs/wm8903.c | 1 + sound/soc/codecs/wm8904.c | 1 + sound/soc/codecs/wm8940.c | 1 + sound/soc/codecs/wm8955.c | 1 + sound/soc/codecs/wm8960.c | 1 + sound/soc/codecs/wm8961.c | 1 + sound/soc/codecs/wm8971.c | 1 + sound/soc/codecs/wm8974.c | 1 + sound/soc/codecs/wm8978.c | 1 + sound/soc/codecs/wm8988.c | 1 + sound/soc/codecs/wm8990.c | 1 + sound/soc/codecs/wm8993.c | 1 + sound/soc/codecs/wm8994.c | 1 + sound/soc/codecs/wm9081.c | 1 + sound/soc/codecs/wm9705.c | 1 + sound/soc/codecs/wm9712.c | 1 + sound/soc/codecs/wm9713.c | 1 + sound/soc/davinci/davinci-i2s.c | 1 + sound/soc/davinci/davinci-mcasp.c | 1 + sound/soc/fsl/fsl_dma.c | 1 + sound/soc/fsl/fsl_ssi.c | 1 + sound/soc/fsl/mpc5200_dma.c | 1 + sound/soc/fsl/mpc8610_hpcd.c | 1 + sound/soc/fsl/soc-of-simple.c | 1 + sound/soc/imx/imx-pcm-dma-mx2.c | 1 + sound/soc/imx/imx-pcm-fiq.c | 1 + sound/soc/imx/imx-ssi.c | 1 + sound/soc/omap/mcpdm.c | 1 + sound/soc/omap/omap-pcm.c | 1 + sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/s6000/s6000-i2s.c | 1 + sound/soc/sh/dma-sh7760.c | 1 + sound/soc/sh/fsi.c | 1 + sound/soc/sh/siu_dai.c | 1 + sound/soc/sh/siu_pcm.c | 1 - sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 1 + sound/soc/txx9/txx9aclc-ac97.c | 1 + sound/soc/txx9/txx9aclc.c | 1 + sound/sound_firmware.c | 1 - sound/sparc/cs4231.c | 1 - sound/sparc/dbri.c | 1 + sound/synth/emux/emux_proc.c | 1 - sound/usb/caiaq/audio.c | 1 + sound/usb/caiaq/device.c | 1 + sound/usb/caiaq/midi.c | 1 + sound/usb/usx2y/us122l.c | 1 + sound/usb/usx2y/usX2Yhwdep.c | 1 + sound/usb/usx2y/usb_stream.c | 1 + sound/usb/usx2y/usbusx2y.c | 1 + sound/usb/usx2y/usbusx2yaudio.c | 1 + sound/usb/usx2y/usx2yhwdeppcm.c | 1 + 210 files changed, 176 insertions(+), 56 deletions(-) (limited to 'sound') diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index 84bb07d..91852e4 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -33,6 +33,7 @@ */ #include #include +#include MODULE_AUTHOR("Johannes Berg "); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index 1dd66dd..fd2188c 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -66,6 +66,7 @@ #include #include #include +#include MODULE_AUTHOR("Johannes Berg "); MODULE_LICENSE("GPL"); diff --git a/sound/aoa/codecs/toonie.c b/sound/aoa/codecs/toonie.c index f13827e..69d2cb6 100644 --- a/sound/aoa/codecs/toonie.c +++ b/sound/aoa/codecs/toonie.c @@ -11,6 +11,7 @@ */ #include #include +#include MODULE_AUTHOR("Johannes Berg "); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("toonie codec driver for snd-aoa"); diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c index 1dd0c28..6776d1c 100644 --- a/sound/aoa/core/gpio-pmf.c +++ b/sound/aoa/core/gpio-pmf.c @@ -6,6 +6,7 @@ * GPL v2, can be found in COPYING. */ +#include #include #include #include "../aoa.h" diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c index 7a437da..1cd9b30 100644 --- a/sound/aoa/fabrics/layout.c +++ b/sound/aoa/fabrics/layout.c @@ -12,6 +12,7 @@ #include #include #include +#include #include "../aoa.h" #include "../soundbus/soundbus.h" diff --git a/sound/aoa/soundbus/i2sbus/control.c b/sound/aoa/soundbus/i2sbus/control.c index 87beb4a..47f854c 100644 --- a/sound/aoa/soundbus/i2sbus/control.c +++ b/sound/aoa/soundbus/i2sbus/control.c @@ -8,6 +8,7 @@ #include #include +#include #include #include diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c index 4e3b819..9d6f3b1 100644 --- a/sound/aoa/soundbus/i2sbus/core.c +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -7,6 +7,7 @@ */ #include +#include #include #include #include diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c index 59bacd3..be83899 100644 --- a/sound/aoa/soundbus/i2sbus/pcm.c +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -8,6 +8,7 @@ #include #include +#include #include #include #include diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index fd51fa8..8808b82 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -4,6 +4,7 @@ * published by the Free Software Foundation. */ +#include #include #include diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index 368dc9c..4268744 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -21,6 +21,7 @@ /* this file included from control.c */ #include +#include struct snd_ctl_elem_list32 { u32 offset; diff --git a/sound/core/hrtimer.c b/sound/core/hrtimer.c index 7f4d744..7730575 100644 --- a/sound/core/hrtimer.c +++ b/sound/core/hrtimer.c @@ -19,6 +19,7 @@ */ #include +#include #include #include #include diff --git a/sound/core/info.c b/sound/core/info.c index d749a0d..cc4a53d 100644 --- a/sound/core/info.c +++ b/sound/core/info.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/core/jack.c b/sound/core/jack.c index f705eec..14b8a4e 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -20,6 +20,7 @@ */ #include +#include #include #include diff --git a/sound/core/misc.c b/sound/core/misc.c index 3da4f92..2c41825 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -21,6 +21,7 @@ #include #include +#include #include #include diff --git a/sound/core/oss/route.c b/sound/core/oss/route.c index 0dcc287..bbe25d8 100644 --- a/sound/core/oss/route.c +++ b/sound/core/oss/route.c @@ -19,7 +19,6 @@ * */ -#include #include #include #include diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index 08bfed5..5fb2e28 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -21,6 +21,7 @@ /* This file included from pcm_native.c */ #include +#include static int snd_pcm_ioctl_delay_compat(struct snd_pcm_substream *substream, s32 __user *src) diff --git a/sound/core/pcm_memory.c b/sound/core/pcm_memory.c index d6d49d6..917e405 100644 --- a/sound/core/pcm_memory.c +++ b/sound/core/pcm_memory.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index d0d721c..6857122 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -29,6 +29,7 @@ #include "seq_oss_event.h" #include #include +#include /* * common variables diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 9dfb2f7..677dc84 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -28,6 +28,7 @@ #include #include "../seq_lock.h" #include +#include /* diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c index f5de79f..73661c4 100644 --- a/sound/core/seq/oss/seq_oss_readq.c +++ b/sound/core/seq/oss/seq_oss_readq.c @@ -25,6 +25,7 @@ #include #include "../seq_lock.h" #include +#include /* * constants diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 945a27c..ee44ab9 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -24,6 +24,7 @@ #include "seq_oss_midi.h" #include "../seq_lock.h" #include +#include /* * constants diff --git a/sound/core/seq/oss/seq_oss_timer.c b/sound/core/seq/oss/seq_oss_timer.c index c440fda..ab59cbf 100644 --- a/sound/core/seq/oss/seq_oss_timer.c +++ b/sound/core/seq/oss/seq_oss_timer.c @@ -23,6 +23,7 @@ #include "seq_oss_timer.h" #include "seq_oss_event.h" #include +#include /* */ diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c index 2174248..d50338b 100644 --- a/sound/core/seq/oss/seq_oss_writeq.c +++ b/sound/core/seq/oss/seq_oss_writeq.c @@ -27,6 +27,7 @@ #include "../seq_lock.h" #include "../seq_clientmgr.h" #include +#include /* diff --git a/sound/core/seq/seq_compat.c b/sound/core/seq/seq_compat.c index c956fe4..81f7c10 100644 --- a/sound/core/seq/seq_compat.c +++ b/sound/core/seq/seq_compat.c @@ -21,6 +21,7 @@ /* This file included from seq.c */ #include +#include struct snd_seq_port_info32 { struct snd_seq_addr addr; /* client/port numbers */ diff --git a/sound/core/seq/seq_system.c b/sound/core/seq/seq_system.c index 77884e6..c38b90c 100644 --- a/sound/core/seq/seq_system.c +++ b/sound/core/seq/seq_system.c @@ -20,6 +20,7 @@ */ #include +#include #include #include "seq_system.h" #include "seq_timer.h" diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 1950ffc..a1282c1 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -39,6 +39,7 @@ #include #include +#include #include #include diff --git a/sound/drivers/mtpav.c b/sound/drivers/mtpav.c index 2f8f295..da03597 100644 --- a/sound/drivers/mtpav.c +++ b/sound/drivers/mtpav.c @@ -54,7 +54,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/drivers/mts64.c b/sound/drivers/mts64.c index 9284829..8539ab0 100644 --- a/sound/drivers/mts64.c +++ b/sound/drivers/mts64.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/drivers/opl3/opl3_oss.c b/sound/drivers/opl3/opl3_oss.c index a54b1dc..ade3ca5 100644 --- a/sound/drivers/opl3/opl3_oss.c +++ b/sound/drivers/opl3/opl3_oss.c @@ -19,7 +19,6 @@ */ #include "opl3_voice.h" -#include static int snd_opl3_open_seq_oss(struct snd_seq_oss_arg *arg, void *closure); static int snd_opl3_close_seq_oss(struct snd_seq_oss_arg *arg); diff --git a/sound/drivers/opl3/opl3_synth.c b/sound/drivers/opl3/opl3_synth.c index 6d57b64..301acb6 100644 --- a/sound/drivers/opl3/opl3_synth.c +++ b/sound/drivers/opl3/opl3_synth.c @@ -19,6 +19,7 @@ * */ +#include #include #include diff --git a/sound/drivers/opl4/opl4_lib.c b/sound/drivers/opl4/opl4_lib.c index 01997f2..f07e38d 100644 --- a/sound/drivers/opl4/opl4_lib.c +++ b/sound/drivers/opl4/opl4_lib.c @@ -20,6 +20,7 @@ #include "opl4_local.h" #include #include +#include #include #include diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c index e1145ac..d77ffa9 100644 --- a/sound/drivers/pcsp/pcsp_lib.c +++ b/sound/drivers/pcsp/pcsp_lib.c @@ -7,6 +7,7 @@ */ #include +#include #include #include #include diff --git a/sound/drivers/portman2x4.c b/sound/drivers/portman2x4.c index 60158e2..f2b0ba2 100644 --- a/sound/drivers/portman2x4.c +++ b/sound/drivers/portman2x4.c @@ -42,6 +42,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/drivers/vx/vx_hwdep.c b/sound/drivers/vx/vx_hwdep.c index 46df881..f7a6fbd 100644 --- a/sound/drivers/vx/vx_hwdep.c +++ b/sound/drivers/vx/vx_hwdep.c @@ -22,6 +22,7 @@ #include #include +#include #include #include #include diff --git a/sound/i2c/other/tea575x-tuner.c b/sound/i2c/other/tea575x-tuner.c index c4c6ef7..ee538f1 100644 --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/isa/cmi8330.c b/sound/isa/cmi8330.c index 8246aae..fe79a16 100644 --- a/sound/isa/cmi8330.c +++ b/sound/isa/cmi8330.c @@ -46,7 +46,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/cs423x/cs4236.c b/sound/isa/cs423x/cs4236.c index cc15d1d..999dc1e 100644 --- a/sound/isa/cs423x/cs4236.c +++ b/sound/isa/cs423x/cs4236.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 9a43baa..fb4d6b3 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -80,7 +80,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c index 534a6ec..c7b80e4 100644 --- a/sound/isa/gus/interwave.c +++ b/sound/isa/gus/interwave.c @@ -26,7 +26,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/msnd/msnd_midi.c b/sound/isa/msnd/msnd_midi.c index 4be562b..7874956 100644 --- a/sound/isa/msnd/msnd_midi.c +++ b/sound/isa/msnd/msnd_midi.c @@ -25,6 +25,7 @@ */ #include +#include #include #include #include diff --git a/sound/isa/opl3sa2.c b/sound/isa/opl3sa2.c index 0481a55..265abcc 100644 --- a/sound/isa/opl3sa2.c +++ b/sound/isa/opl3sa2.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index 5913717..8c24102 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index 4d2d040..c35dc68 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/sb/emu8000_pcm.c b/sound/isa/sb/emu8000_pcm.c index 91dc3d8..ccedbfe 100644 --- a/sound/isa/sb/emu8000_pcm.c +++ b/sound/isa/sb/emu8000_pcm.c @@ -20,6 +20,7 @@ #include "emu8000_local.h" #include +#include #include #include diff --git a/sound/isa/sb/sb16.c b/sound/isa/sb/sb16.c index 519c363..4d1c5a3 100644 --- a/sound/isa/sb/sb16.c +++ b/sound/isa/sb/sb16.c @@ -21,7 +21,6 @@ #include #include -#include #include #include #include diff --git a/sound/isa/sb/sb8.c b/sound/isa/sb/sb8.c index 3cd57ee..81284a8 100644 --- a/sound/isa/sb/sb8.c +++ b/sound/isa/sb/sb8.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/isa/wavefront/wavefront.c b/sound/isa/wavefront/wavefront.c index a34ae7b..711670e 100644 --- a/sound/isa/wavefront/wavefront.c +++ b/sound/isa/wavefront/wavefront.c @@ -21,7 +21,6 @@ #include #include -#include #include #include #include diff --git a/sound/isa/wavefront/wavefront_fx.c b/sound/isa/wavefront/wavefront_fx.c index 2bb1cee..657e2d6 100644 --- a/sound/isa/wavefront/wavefront_fx.c +++ b/sound/isa/wavefront/wavefront_fx.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 5d4ff48..4fb7b19 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c index 9a88cdf..453d343 100644 --- a/sound/mips/hal2.c +++ b/sound/mips/hal2.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c index 6aff217..717604c 100644 --- a/sound/mips/sgio2audio.c +++ b/sound/mips/sgio2audio.c @@ -25,11 +25,11 @@ #include #include #include -#include #include #include #include #include +#include #include #include diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index d12bd98..24793c5 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -45,6 +45,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/oss/dmabuf.c b/sound/oss/dmabuf.c index 1bfcf7e..bcc3e8e 100644 --- a/sound/oss/dmabuf.c +++ b/sound/oss/dmabuf.c @@ -26,6 +26,7 @@ #define SAMPLE_ROUNDUP 0 #include +#include #include "sound_config.h" #define DMAP_FREE_ON_CLOSE 0 diff --git a/sound/oss/kahlua.c b/sound/oss/kahlua.c index 24d152c..52d06a3 100644 --- a/sound/oss/kahlua.c +++ b/sound/oss/kahlua.c @@ -31,6 +31,7 @@ #include #include #include +#include #include "sound_config.h" diff --git a/sound/oss/mpu401.c b/sound/oss/mpu401.c index 0af9d24..25e4609 100644 --- a/sound/oss/mpu401.c +++ b/sound/oss/mpu401.c @@ -19,6 +19,7 @@ */ #include +#include #include #include #include diff --git a/sound/oss/msnd.c b/sound/oss/msnd.c index 21eb6dc..c0cc951 100644 --- a/sound/oss/msnd.c +++ b/sound/oss/msnd.c @@ -24,7 +24,6 @@ #include #include -#include #include #include #include diff --git a/sound/oss/msnd_pinnacle.c b/sound/oss/msnd_pinnacle.c index bf27e00..a1e3f96 100644 --- a/sound/oss/msnd_pinnacle.c +++ b/sound/oss/msnd_pinnacle.c @@ -35,12 +35,12 @@ #include #include -#include #include #include #include #include #include +#include #include #include #include "sound_config.h" diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index 7781c13..938c48c 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -24,6 +24,7 @@ */ #include +#include #include #include diff --git a/sound/oss/sb_card.c b/sound/oss/sb_card.c index 7de18b5..84ef4d0 100644 --- a/sound/oss/sb_card.c +++ b/sound/oss/sb_card.c @@ -24,6 +24,7 @@ #include #include +#include #include #include "sound_config.h" #include "sb_mixer.h" diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c index ce4db49..7d42c54 100644 --- a/sound/oss/sb_common.c +++ b/sound/oss/sb_common.c @@ -31,6 +31,7 @@ #include #include #include +#include #include "sound_config.h" #include "sound_firmware.h" diff --git a/sound/oss/sb_midi.c b/sound/oss/sb_midi.c index 8b79670..f139028 100644 --- a/sound/oss/sb_midi.c +++ b/sound/oss/sb_midi.c @@ -12,6 +12,7 @@ */ #include +#include #include "sound_config.h" diff --git a/sound/oss/sb_mixer.c b/sound/oss/sb_mixer.c index fad1a4f..2039d31 100644 --- a/sound/oss/sb_mixer.c +++ b/sound/oss/sb_mixer.c @@ -16,6 +16,8 @@ * Stanislav Voronyi : Support for AWE 3DSE device (Jun 7 1999) */ +#include + #include "sound_config.h" #define __SB_MIXER_C__ diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index fde7c12..2d9c513 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -36,7 +36,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index a446b82..8e514a6 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include "sound_config.h" diff --git a/sound/oss/v_midi.c b/sound/oss/v_midi.c index 103940f..f0b4151 100644 --- a/sound/oss/v_midi.c +++ b/sound/oss/v_midi.c @@ -21,6 +21,7 @@ #include #include +#include #include #include "sound_config.h" diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index a4127ba..ac39a53 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -17,6 +17,7 @@ * We currently support a mixer device, but it is currently non-functional. */ +#include #include #include #include diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 6713110..20b3b32 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -149,6 +149,7 @@ #include #include #include +#include #include diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index 2c63bb9..e688dde 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -35,6 +35,7 @@ #include #include +#include #include #include #include diff --git a/sound/pci/ac97/ac97_proc.c b/sound/pci/ac97/ac97_proc.c index 73b17d5..6320bf0 100644 --- a/sound/pci/ac97/ac97_proc.c +++ b/sound/pci/ac97/ac97_proc.c @@ -22,7 +22,6 @@ * */ -#include #include #include diff --git a/sound/pci/als4000.c b/sound/pci/als4000.c index d75cf7b..6cf1de8 100644 --- a/sound/pci/als4000.c +++ b/sound/pci/als4000.c @@ -68,7 +68,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/aw2/aw2-saa7146.c b/sound/pci/aw2/aw2-saa7146.c index 296123a..8afd8b5 100644 --- a/sound/pci/aw2/aw2-saa7146.c +++ b/sound/pci/aw2/aw2-saa7146.c @@ -25,7 +25,6 @@ #include #include -#include #include #include #include diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 8f443a9..85fd315 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -63,7 +63,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index 0470461..ba96428 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -63,7 +63,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/cs5530.c b/sound/pci/cs5530.c index 207479a..bc07e27 100644 --- a/sound/pci/cs5530.c +++ b/sound/pci/cs5530.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index 0f48a87..f16bc8a 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -23,7 +23,6 @@ */ #include -#include #include #include #include diff --git a/sound/pci/cs5535audio/cs5535audio_pm.c b/sound/pci/cs5535audio/cs5535audio_pm.c index 564c33b..a3301cc 100644 --- a/sound/pci/cs5535audio/cs5535audio_pm.c +++ b/sound/pci/cs5535audio/cs5535audio_pm.c @@ -19,7 +19,6 @@ */ #include -#include #include #include #include diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index 480cb1e..1bff80c 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -24,6 +24,7 @@ #include "ctdaio.h" #include "cttimer.h" #include +#include #include #include #include diff --git a/sound/pci/ctxfi/ctpcm.c b/sound/pci/ctxfi/ctpcm.c index d0dc227..85ab43e 100644 --- a/sound/pci/ctxfi/ctpcm.c +++ b/sound/pci/ctxfi/ctpcm.c @@ -17,6 +17,7 @@ #include "ctpcm.h" #include "cttimer.h" +#include #include /* Hardware descriptions for playback */ diff --git a/sound/pci/echoaudio/darla20.c b/sound/pci/echoaudio/darla20.c index a65bafe..fe7ad64 100644 --- a/sound/pci/echoaudio/darla20.c +++ b/sound/pci/echoaudio/darla20.c @@ -40,9 +40,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/darla24.c b/sound/pci/echoaudio/darla24.c index 0a6c50b..d1fd34b 100644 --- a/sound/pci/echoaudio/darla24.c +++ b/sound/pci/echoaudio/darla24.c @@ -44,9 +44,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/echo3g.c b/sound/pci/echoaudio/echo3g.c index f514279..1dffdc5 100644 --- a/sound/pci/echoaudio/echo3g.c +++ b/sound/pci/echoaudio/echo3g.c @@ -51,9 +51,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/gina20.c b/sound/pci/echoaudio/gina20.c index 2364f8a..050e54a 100644 --- a/sound/pci/echoaudio/gina20.c +++ b/sound/pci/echoaudio/gina20.c @@ -44,9 +44,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/gina24.c b/sound/pci/echoaudio/gina24.c index 616b558..5748fc6 100644 --- a/sound/pci/echoaudio/gina24.c +++ b/sound/pci/echoaudio/gina24.c @@ -50,9 +50,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/indigo.c b/sound/pci/echoaudio/indigo.c index 776175c..4ae5e35 100644 --- a/sound/pci/echoaudio/indigo.c +++ b/sound/pci/echoaudio/indigo.c @@ -42,9 +42,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/indigodj.c b/sound/pci/echoaudio/indigodj.c index 8816b0b..3550715 100644 --- a/sound/pci/echoaudio/indigodj.c +++ b/sound/pci/echoaudio/indigodj.c @@ -42,9 +42,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/indigodjx.c b/sound/pci/echoaudio/indigodjx.c index b1e3652..19b191f 100644 --- a/sound/pci/echoaudio/indigodjx.c +++ b/sound/pci/echoaudio/indigodjx.c @@ -42,10 +42,10 @@ #include #include #include -#include #include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/indigoio.c b/sound/pci/echoaudio/indigoio.c index 1035125..a9fcedf 100644 --- a/sound/pci/echoaudio/indigoio.c +++ b/sound/pci/echoaudio/indigoio.c @@ -43,9 +43,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/indigoiox.c b/sound/pci/echoaudio/indigoiox.c index 60b7cb2..bcdfac6 100644 --- a/sound/pci/echoaudio/indigoiox.c +++ b/sound/pci/echoaudio/indigoiox.c @@ -43,10 +43,10 @@ #include #include #include -#include #include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/layla20.c b/sound/pci/echoaudio/layla20.c index 8c3f5c5..d3a98c5 100644 --- a/sound/pci/echoaudio/layla20.c +++ b/sound/pci/echoaudio/layla20.c @@ -49,9 +49,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/layla24.c b/sound/pci/echoaudio/layla24.c index ed1cc0a..2a1dca6d 100644 --- a/sound/pci/echoaudio/layla24.c +++ b/sound/pci/echoaudio/layla24.c @@ -51,9 +51,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/mia.c b/sound/pci/echoaudio/mia.c index cc2bbfc..9cdf14c 100644 --- a/sound/pci/echoaudio/mia.c +++ b/sound/pci/echoaudio/mia.c @@ -50,9 +50,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/echoaudio/mona.c b/sound/pci/echoaudio/mona.c index 3e7e0182..1047be4 100644 --- a/sound/pci/echoaudio/mona.c +++ b/sound/pci/echoaudio/mona.c @@ -48,9 +48,9 @@ #include #include #include -#include #include #include +#include #include #include #include diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 6a47672..ffb1ddb 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -22,6 +22,7 @@ */ #include +#include #include #include diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index e4581a42..29714c8 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -21,6 +21,7 @@ #include #include +#include #include #include #include "hda_beep.h" diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index dcd2244..d8da18a 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -22,6 +22,7 @@ */ #include +#include #include #include #include "hda_codec.h" diff --git a/sound/pci/ice1712/ak4xxx.c b/sound/pci/ice1712/ak4xxx.c index 03391da..90d560c 100644 --- a/sound/pci/ice1712/ak4xxx.c +++ b/sound/pci/ice1712/ak4xxx.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/pci/ice1712/amp.c b/sound/pci/ice1712/amp.c index 6da21a2..e328cfb 100644 --- a/sound/pci/ice1712/amp.c +++ b/sound/pci/ice1712/amp.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include "ice1712.h" diff --git a/sound/pci/ice1712/vt1720_mobo.c b/sound/pci/ice1712/vt1720_mobo.c index 7f9674b..4c551e1 100644 --- a/sound/pci/ice1712/vt1720_mobo.c +++ b/sound/pci/ice1712/vt1720_mobo.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include "ice1712.h" diff --git a/sound/pci/ice1712/wtm.c b/sound/pci/ice1712/wtm.c index 5af9e84..e618f78 100644 --- a/sound/pci/ice1712/wtm.c +++ b/sound/pci/ice1712/wtm.c @@ -29,7 +29,6 @@ #include #include #include -#include #include #include "ice1712.h" diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 0cca560..ef9af3f 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 7e8e7da..55e9315 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include diff --git a/sound/pci/mixart/mixart_hwdep.c b/sound/pci/mixart/mixart_hwdep.c index 4cf4cd8..bf2696a 100644 --- a/sound/pci/mixart/mixart_hwdep.c +++ b/sound/pci/mixart/mixart_hwdep.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include "mixart.h" diff --git a/sound/pci/oxygen/oxygen_lib.c b/sound/pci/oxygen/oxygen_lib.c index 9c5e645..fad03d6 100644 --- a/sound/pci/oxygen/oxygen_lib.c +++ b/sound/pci/oxygen/oxygen_lib.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c index d5e1c6e..3c04524 100644 --- a/sound/pci/rme32.c +++ b/sound/pci/rme32.c @@ -70,10 +70,10 @@ #include +#include #include #include #include -#include #include #include diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index 9d5252b..d19dc05 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -27,7 +27,6 @@ #include #include #include -#include #include #include diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 52c6eb5..b92adef 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/pci/rme9652/rme9652.c b/sound/pci/rme9652/rme9652.c index 44a3e2d..c492af5 100644 --- a/sound/pci/rme9652/rme9652.c +++ b/sound/pci/rme9652/rme9652.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 7e3e8fb..9cc1b5a 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c index 5d2afa0..9dce0bd 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_core.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_core.c @@ -19,6 +19,7 @@ */ #include +#include #include #include #include "pdaudiocf.h" diff --git a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c index 0d668f4..43f995a 100644 --- a/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c +++ b/sound/pcmcia/pdaudiocf/pdaudiocf_pcm.c @@ -20,7 +20,6 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ -#include #include #include #include diff --git a/sound/pcmcia/vx/vxpocket.c b/sound/pcmcia/vx/vxpocket.c index 7be3b33..cfd1438 100644 --- a/sound/pcmcia/vx/vxpocket.c +++ b/sound/pcmcia/vx/vxpocket.c @@ -21,6 +21,7 @@ #include #include +#include #include #include "vxpocket.h" #include diff --git a/sound/ppc/burgundy.c b/sound/ppc/burgundy.c index 1f72e1c..00e2d51 100644 --- a/sound/ppc/burgundy.c +++ b/sound/ppc/burgundy.c @@ -21,7 +21,6 @@ #include #include -#include #include #include #include "pmac.h" diff --git a/sound/ppc/keywest.c b/sound/ppc/keywest.c index d06f780..8f064c7 100644 --- a/sound/ppc/keywest.c +++ b/sound/ppc/keywest.c @@ -22,7 +22,6 @@ #include #include #include -#include #include #include "pmac.h" diff --git a/sound/ppc/snd_ps3.c b/sound/ppc/snd_ps3.c index 53c81a5..2f12da4 100644 --- a/sound/ppc/snd_ps3.c +++ b/sound/ppc/snd_ps3.c @@ -20,10 +20,10 @@ #include #include +#include #include #include #include -#include #include #include diff --git a/sound/sh/sh_dac_audio.c b/sound/sh/sh_dac_audio.c index 76d9ad2..68e0dee 100644 --- a/sound/sh/sh_dac_audio.c +++ b/sound/sh/sh_dac_audio.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 340311d..a61ccd2 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -17,6 +17,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 0cf2ca6..495be6e 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -18,6 +18,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 67cbfe7..5e7aacf 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -29,8 +29,8 @@ #include #include #include -#include #include +#include #include #include diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index e693229..523b7fc 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index c6c6a4a7..1d2a1ad 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -29,8 +29,8 @@ #include #include #include -#include #include +#include #include #include diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 5e03bb2..6bac1ac 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -29,8 +29,8 @@ #include #include #include -#include #include +#include #include #include diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index a1bbe16..fd101d4 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -13,6 +13,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 3c80137..11b62de 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -17,6 +17,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index c233810..240cd15 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -27,6 +27,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 39c0f758..0420727 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -12,6 +12,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ad73311.c b/sound/soc/codecs/ad73311.c index d2fcc60..475807b 100644 --- a/sound/soc/codecs/ad73311.c +++ b/sound/soc/codecs/ad73311.c @@ -11,6 +11,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ads117x.c b/sound/soc/codecs/ads117x.c index cc96411..f8e75ed 100644 --- a/sound/soc/codecs/ads117x.c +++ b/sound/soc/codecs/ads117x.c @@ -11,6 +11,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index b68d99f..bdeb10d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -10,6 +10,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index ff96656..352d1d0 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 3ef16bb..729859c 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -29,6 +29,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 82fca28..926797a 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index dfbeb2d..81a62d1 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -23,6 +23,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index e000cdf..9f169c4 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -14,6 +14,7 @@ */ #include +#include #include #include diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index cf2975a..366daf1 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/pcm3008.c b/sound/soc/codecs/pcm3008.c index 2afcd0a..5a5f187 100644 --- a/sound/soc/codecs/pcm3008.c +++ b/sound/soc/codecs/pcm3008.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index d2ff1cd..29d0906 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -33,6 +33,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index 81b8c9d..3293629 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -15,6 +15,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index da589d8..776b79c 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 357b609..b5b7d6a 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index e4b946a..4a6d56c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index d50f169..d1e0e81 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -31,6 +31,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 958d49c..569ad87 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -26,6 +26,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 6f5d4af..520ffd6 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -27,6 +27,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index 3e99fe5..a8dcd5a 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -15,6 +15,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 217b026..a34cbcf 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -32,6 +32,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index df2c6d9..2e0772f 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index b432f4d..6acc885 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index af8cb69..9000b1d 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index d3a61d7..19cd472 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index d077df6..8cc9042 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index 24a3560..8ca3812 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8727.c b/sound/soc/codecs/wm8727.c index 63a254e..1072621 100644 --- a/sound/soc/codecs/wm8727.c +++ b/sound/soc/codecs/wm8727.c @@ -13,6 +13,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index 3fb653b..07adc37 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5a2619d..e7c6bf1 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 475c67a..2916ed4 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index c2444e7..613199a 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -40,6 +40,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index 44e7d9d..60b1b3e 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index dbc368c..b7fd96a 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -24,6 +24,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 3595bd5..fa5f99f 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 593e47d..c6f0abc 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index 31e39ff..0c04b47 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -30,6 +30,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 615dab2..c8d7a80 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index d07bcc1..f1e63e0 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index d2342c5..50634ab7 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index d9540d5..a65b781 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -20,6 +20,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index ee637af..69708c4 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 28bb59e..526f56b 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index 2862e4d..bb18c3e 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 056b787..831f473 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index bf022f6..03e8b1a 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 29f3771..8d1c637 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index c468497..3a184fc 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index ec54c6d..8793341 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -10,6 +10,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index e237bf6..2f48a8a 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -11,6 +11,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index ceb86b4..2fca514 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -16,6 +16,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 6362ca0..62af7e0 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -12,6 +12,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ab6518d..6c80cc3 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -18,6 +18,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index b1a3a27..410c749 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 93f0f38..762c1b8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 30ed568..d639e55 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -8,6 +8,7 @@ #include #include +#include #include diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index ef67d1c..83de1c8 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -9,6 +9,7 @@ * express or implied. */ +#include #include #include #include diff --git a/sound/soc/fsl/soc-of-simple.c b/sound/soc/fsl/soc-of-simple.c index 8bc5cd9..3bc13fd 100644 --- a/sound/soc/fsl/soc-of-simple.c +++ b/sound/soc/fsl/soc-of-simple.c @@ -12,6 +12,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 19452e4..86668ab 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index d9cb984..f96a373 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 56f46a7..6546b06 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -39,6 +39,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/omap/mcpdm.c b/sound/soc/omap/mcpdm.c index ad8df6c..1dab4c1 100644 --- a/sound/soc/omap/mcpdm.c +++ b/sound/soc/omap/mcpdm.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 825db38..ba8acbb 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -23,6 +23,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 9e95e51..d5fc52d 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -16,6 +16,7 @@ #include #include +#include #include #include #include diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index c5cda18..0664fac 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index baddb12..0d8bdf0 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -13,6 +13,7 @@ */ #include +#include #include #include #include diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 993abb7..8dc966f 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -19,6 +19,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index 5452d19..d86ee1b 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index ba7f8d0..8f85719 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -24,7 +24,6 @@ #include #include #include -#include #include #include diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c8b0556..2320153 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -28,6 +28,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 6c33510..7c28f40 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -38,6 +38,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/txx9/txx9aclc-ac97.c b/sound/soc/txx9/txx9aclc-ac97.c index 0f83bdb..612e18b 100644 --- a/sound/soc/txx9/txx9aclc-ac97.c +++ b/sound/soc/txx9/txx9aclc-ac97.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index efed64b..49cc7ea 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/sound_firmware.c b/sound/sound_firmware.c index 96deaef..340a0bc 100644 --- a/sound/sound_firmware.c +++ b/sound/sound_firmware.c @@ -2,7 +2,6 @@ #include #include #include -#include #include #include #include "oss/sound_firmware.h" diff --git a/sound/sparc/cs4231.c b/sound/sparc/cs4231.c index 8d13d93..7dcc0651 100644 --- a/sound/sparc/cs4231.c +++ b/sound/sparc/cs4231.c @@ -10,7 +10,6 @@ #include #include -#include #include #include #include diff --git a/sound/sparc/dbri.c b/sound/sparc/dbri.c index 1d2e51b..2eab6ce 100644 --- a/sound/sparc/dbri.c +++ b/sound/sparc/dbri.c @@ -58,6 +58,7 @@ #include #include #include +#include #include #include diff --git a/sound/synth/emux/emux_proc.c b/sound/synth/emux/emux_proc.c index 687e6a1..58a32a1 100644 --- a/sound/synth/emux/emux_proc.c +++ b/sound/synth/emux/emux_proc.c @@ -19,7 +19,6 @@ */ #include -#include #include #include #include diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index 86b2c3b..4328cad 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -17,6 +17,7 @@ */ #include +#include #include #include #include diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index a3f02dd..afc5aeb 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -23,6 +23,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/usb/caiaq/midi.c b/sound/usb/caiaq/midi.c index 538e8c0..2f218c7 100644 --- a/sound/usb/caiaq/midi.c +++ b/sound/usb/caiaq/midi.c @@ -17,6 +17,7 @@ */ #include +#include #include #include #include diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 44deb21..9ca9a13 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -16,6 +16,7 @@ * Inc., 675 Mass Ave, Cambridge, MA 02139, USA. */ +#include #include #include #include diff --git a/sound/usb/usx2y/usX2Yhwdep.c b/sound/usb/usx2y/usX2Yhwdep.c index 1879b72..04aafb4 100644 --- a/sound/usb/usx2y/usX2Yhwdep.c +++ b/sound/usb/usx2y/usX2Yhwdep.c @@ -21,6 +21,7 @@ */ #include +#include #include #include #include diff --git a/sound/usb/usx2y/usb_stream.c b/sound/usb/usx2y/usb_stream.c index 12ae034..c400ade 100644 --- a/sound/usb/usx2y/usb_stream.c +++ b/sound/usb/usx2y/usb_stream.c @@ -17,6 +17,7 @@ */ #include +#include #include "usb_stream.h" diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index c42350e..cbd37f2 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -133,6 +133,7 @@ #include #include #include +#include #include #include #include diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 74a67a8..5d37d1c 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -32,6 +32,7 @@ #include +#include #include #include #include diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index 9ed6c39..2a528e5 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -51,6 +51,7 @@ */ #include +#include #include "usbusx2yaudio.c" #if defined(USX2Y_NRPACKS_VARIABLE) || (!defined(USX2Y_NRPACKS_VARIABLE) && USX2Y_NRPACKS == 1) -- cgit v1.1 From b8e80cf386419453678b01bef830f53445ebb15d Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Tue, 30 Mar 2010 13:29:28 -0400 Subject: ALSA: hda: Fix 0 dB offset for Lenovo Thinkpad models using AD1981 BugLink: https://launchpad.net/bugs/551606 The OR's hardware distorts at PCM 100% because it does not correspond to 0 dB. Fix this in patch_ad1981() for all models using the Thinkpad quirk. Reported-by: Jane Silber Cc: Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_analog.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index e6d1bdf..af34606 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -1896,6 +1896,14 @@ static int patch_ad1981(struct hda_codec *codec) case AD1981_THINKPAD: spec->mixers[0] = ad1981_thinkpad_mixers; spec->input_mux = &ad1981_thinkpad_capture_source; + /* set the upper-limit for mixer amp to 0dB for avoiding the + * possible damage by overloading + */ + snd_hda_override_amp_caps(codec, 0x11, 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 AD1981_TOSHIBA: spec->mixers[0] = ad1981_hp_mixers; -- cgit v1.1 From b5442a75deee293d10c2ab8f4a77013973c4c9e0 Mon Sep 17 00:00:00 2001 From: Janusz Krzysztofik Date: Sun, 28 Mar 2010 22:29:29 +0200 Subject: ASoC: OMAP: Fix capture pointer handling for OMAP1510 to work correctly with recent ALSA PCM code With recent (2.6.34) chnages in PCM handling, capture stopped working on my OMAP1510 based Amstrad Delta videophone. Using 2.6.34-rc2, I was able to correct the problem in 3 different ways: 1. reverting commit 7b3a177b0d4f92b3431b8dca777313a07533a710, 2. enabling additional jiffies check with echo 4 >/proc/asound/card0/pcm0c0/xrun_debug 3. applying the patch below. Since I wasn't able to reproduce the problem on my i686 PC, I guess the problem is probably machine specific. The patch reuses the method for software emulation of missing hardware pointer, already implemented for playback on OMAP1510. It's possible that event if a hardware pointer is available for capture on this machine, its behaviour may be not compatible with what upper layer expects. If you think the problem may be more general and should be solved differently, on a higher level, I can try to work more on it if you give me a hint. If the patch gets accepted, I suggest it goes as a fix in the current release cycle. Created and tested against linux-2.6.34-rc2. Signed-off-by: Janusz Krzysztofik Acked-by: Jarkko Nikula Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index 825db38..bdd1097 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -60,12 +60,11 @@ static void omap_pcm_dma_irq(int ch, u16 stat, void *data) struct omap_runtime_data *prtd = runtime->private_data; unsigned long flags; - if ((cpu_is_omap1510()) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) { + if ((cpu_is_omap1510())) { /* * OMAP1510 doesn't fully support DMA progress counter * and there is no software emulation implemented yet, - * so have to maintain our own playback progress counter + * so have to maintain our own progress counters * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); @@ -189,8 +188,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream) dma_params.frame_count = runtime->periods; omap_set_dma_params(prtd->dma_ch, &dma_params); - if ((cpu_is_omap1510()) && - (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + if ((cpu_is_omap1510())) omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ | OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ); else @@ -248,14 +246,15 @@ static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream) dma_addr_t ptr; snd_pcm_uframes_t offset; - if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + if (cpu_is_omap1510()) { + offset = prtd->period_index * runtime->period_size; + } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { ptr = omap_get_dma_dst_pos(prtd->dma_ch); offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); - } else if (!(cpu_is_omap1510())) { + } else { ptr = omap_get_dma_src_pos(prtd->dma_ch); offset = bytes_to_frames(runtime, ptr - runtime->dma_addr); - } else - offset = prtd->period_index * runtime->period_size; + } if (offset >= runtime->buffer_size) offset = 0; -- cgit v1.1 From 3815595e78d2baae6feb866e737f92d8ef48b337 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sun, 4 Apr 2010 12:14:03 +0200 Subject: ALSA: hda - Add MSI blacklist for Aopen MZ915-M The device needs MSI disablement. Added to the quirk list. Reported-by: Harald Dunkel Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4bb9067..f8fd586 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2362,6 +2362,7 @@ static struct snd_pci_quirk msi_black_list[] __devinitdata = { SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ SND_PCI_QUIRK(0x1849, 0x0888, "ASRock", 0), /* Athlon64 X2 + nvidia */ + SND_PCI_QUIRK(0xa0a0, 0x0575, "Aopen MZ915-M", 0), /* ICH6 */ {} }; -- cgit v1.1 From f11947c7c5b8abffd328739996dfdffef2b3e03f Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Fri, 2 Apr 2010 14:29:23 +0300 Subject: ALSA: i2c: cleanup: change parameter to pointer We actually pass an array of 7 chars not 5. This silences a smatch warning. Signed-off-by: Dan Carpenter Signed-off-by: Takashi Iwai --- sound/i2c/other/ak4113.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/i2c/other/ak4113.c b/sound/i2c/other/ak4113.c index fff62cc..971a84a 100644 --- a/sound/i2c/other/ak4113.c +++ b/sound/i2c/other/ak4113.c @@ -70,7 +70,7 @@ static int snd_ak4113_dev_free(struct snd_device *device) } int snd_ak4113_create(struct snd_card *card, ak4113_read_t *read, - ak4113_write_t *write, const unsigned char pgm[5], + ak4113_write_t *write, const unsigned char *pgm, void *private_data, struct ak4113 **r_ak4113) { struct ak4113 *chip; -- cgit v1.1 From a0fd4345f928d72a56e27b23e4cd28c94bf36be5 Mon Sep 17 00:00:00 2001 From: Julia Lawall Date: Fri, 2 Apr 2010 14:47:59 +0200 Subject: ALSA: echoaudio - Eliminate use after free Use the call to snd_card_free in the error handling code at the end of the function, as in the other error cases. A simplified version of the semantic patch that finds this problem is as follows: (http://coccinelle.lip6.fr/) // @@ expression E,E2; @@ snd_card_free(E) ... ( E = E2 | * E ) // Signed-off-by: Julia Lawall Signed-off-by: Takashi Iwai --- sound/pci/echoaudio/echoaudio.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index 8dab82d..668a5ec 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -2184,10 +2184,9 @@ static int __devinit snd_echo_probe(struct pci_dev *pci, goto ctl_error; #endif - if ((err = snd_card_register(card)) < 0) { - snd_card_free(card); + err = snd_card_register(card); + if (err < 0) goto ctl_error; - } snd_printk(KERN_INFO "Card registered: %s\n", card->longname); pci_set_drvdata(pci, chip); -- cgit v1.1 From 3fa49e3ad9ac20b15edfb0c51bbad36e45a84b17 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Mar 2010 15:24:40 +0100 Subject: ASoC: Avoid wraparound in wm_hubs DC servo correction If the correction wraps around then a substantial offset would be introduced. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 486bdd2..3729a12 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -113,13 +113,15 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* HPOUT1L */ reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) & WM8993_DCS_INTEG_CHAN_0_MASK;; - reg += hubs->dcs_codes; + if (reg + hubs->dcs_codes > 0 && reg + hubs->dcs_codes < 0xff) + reg += hubs->dcs_codes; dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1R */ reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) & WM8993_DCS_INTEG_CHAN_1_MASK; - reg += hubs->dcs_codes; + if (reg + hubs->dcs_codes > 0 && reg + hubs->dcs_codes < 0xff) + reg += hubs->dcs_codes; dcs_cfg |= reg; /* Do it */ -- cgit v1.1 From 8437f7006b9cfa249791e2fd57596683d4561843 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Mar 2010 17:09:45 +0100 Subject: ASoC: Support second DC servo readback method for wm_hubs More recent Wolfson hubs devices add the ability to read back the DC servo calibration information from the register used to write offsets, and later still ones remove the old readback registers. Add support for the new scheme, and use it for WM8994 device revisions that support it. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 3 ++- sound/soc/codecs/wm_hubs.c | 41 ++++++++++++++++++++++++++++++----------- sound/soc/codecs/wm_hubs.h | 1 + 3 files changed, 33 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d10d651..c80218f 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3730,11 +3730,12 @@ static int wm8994_codec_probe(struct platform_device *pdev) case 3: wm8994->hubs.dcs_codes = -5; wm8994->hubs.hp_startup_mode = 1; + wm8994->hubs.dcs_readback_mode = 1; break; default: + wm8994->hubs.dcs_readback_mode = 1; break; } - /* Remember if AIFnLRCLK is configured as a GPIO. This should be * configured on init - if a system wants to do this dynamically diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 3729a12..2b5c092 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -86,7 +86,7 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec) static void calibrate_dc_servo(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = codec->private_data; - u16 reg, dcs_cfg; + u16 reg, reg_l, reg_r, dcs_cfg; /* Set for 32 series updates */ snd_soc_update_bits(codec, WM8993_DC_SERVO_1, @@ -110,19 +110,38 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Applying %d code DC servo correction\n", hubs->dcs_codes); + /* Different chips in the family support different + * readback methods. + */ + switch (hubs->dcs_readback_mode) { + case 0: + reg_l = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) + & WM8993_DCS_INTEG_CHAN_0_MASK;; + reg_r = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) + & WM8993_DCS_INTEG_CHAN_1_MASK; + break; + case 1: + reg = snd_soc_read(codec, WM8993_DC_SERVO_3); + reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) + >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; + reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; + break; + default: + WARN(1, "Unknown DCS readback method"); + break; + } + /* HPOUT1L */ - reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_1) & - WM8993_DCS_INTEG_CHAN_0_MASK;; - if (reg + hubs->dcs_codes > 0 && reg + hubs->dcs_codes < 0xff) - reg += hubs->dcs_codes; - dcs_cfg = reg << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + if (reg_l + hubs->dcs_codes > 0 && + reg_l + hubs->dcs_codes < 0xff) + reg_l += hubs->dcs_codes; + dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; /* HPOUT1R */ - reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_2) & - WM8993_DCS_INTEG_CHAN_1_MASK; - if (reg + hubs->dcs_codes > 0 && reg + hubs->dcs_codes < 0xff) - reg += hubs->dcs_codes; - dcs_cfg |= reg; + if (reg_r + hubs->dcs_codes > 0 && + reg_r + hubs->dcs_codes < 0xff) + reg_r += hubs->dcs_codes; + dcs_cfg |= reg_r; /* Do it */ snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 420104f..e51c166 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -21,6 +21,7 @@ extern const unsigned int wm_hubs_spkmix_tlv[]; /* This *must* be the first element of the codec->private_data struct */ struct wm_hubs_data { int dcs_codes; + int dcs_readback_mode; int hp_startup_mode; }; -- cgit v1.1 From ae9d8607fe24253efc9f14b696f51cfd683801be Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Mar 2010 16:34:42 +0100 Subject: ASoC: Don't do runtime wm_hubs DC servo updates if using offset correction If we need to offset correct the DC servo then don't use runtime recalibration since that is likely to introduce further offsets which will be evident on powerdown. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2b5c092..e81ba6d 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -162,10 +162,16 @@ static int wm8993_put_dc_servo(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct wm_hubs_data *hubs = codec->private_data; int ret; ret = snd_soc_put_volsw_2r(kcontrol, ucontrol); + /* If we're applying an offset correction then updating the + * callibration would be likely to introduce further offsets. */ + if (hubs->dcs_codes) + return ret; + /* Only need to do this if the outputs are active */ if (snd_soc_read(codec, WM8993_POWER_MANAGEMENT_1) & (WM8993_HPOUT1L_ENA | WM8993_HPOUT1R_ENA)) -- cgit v1.1 From 4dcc93d0ede49fae32dd0ee41c685db1be14c529 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Mar 2010 17:18:41 +0100 Subject: ASoC: Don't use DCS_DATAPATH_BUSY for WM hubs devices The DCS_DATAPATH_BUSY bit used to monitor the completion of DC servo operations has been deprecated and with some more recente revisions may perform incorrectly, especially when only analogue bypass paths are in use. Switch to using readback from the DC servo command register instead, which is supported for all devices. Without this unacceptably long timeouts may be observed in some circumstances. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm_hubs.c | 38 +++++++++++++++----------------------- 1 file changed, 15 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index e81ba6d..e1f225a 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -62,21 +62,27 @@ static const char *speaker_mode_text[] = { static const struct soc_enum speaker_mode = SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text); -static void wait_for_dc_servo(struct snd_soc_codec *codec) +static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) { unsigned int reg; int count = 0; + unsigned int val; + + val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1; + + /* Trigger the command */ + snd_soc_write(codec, WM8993_DC_SERVO_0, val); dev_dbg(codec->dev, "Waiting for DC servo...\n"); do { count++; msleep(1); - reg = snd_soc_read(codec, WM8993_DC_SERVO_READBACK_0); + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & WM8993_DCS_DATAPATH_BUSY && count < 400); + } while (reg & op && count < 400); - if (reg & WM8993_DCS_DATAPATH_BUSY) + if (reg & op) dev_err(codec->dev, "Timed out waiting for DC Servo\n"); } @@ -92,18 +98,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8993_DC_SERVO_1, WM8993_DCS_SERIES_NO_01_MASK, 32 << WM8993_DCS_SERIES_NO_01_SHIFT); - - /* Enable the DC servo. Write all bits to avoid triggering startup - * or write calibration. - */ - snd_soc_update_bits(codec, WM8993_DC_SERVO_0, - 0xFFFF, - WM8993_DCS_ENA_CHAN_0 | - WM8993_DCS_ENA_CHAN_1 | - WM8993_DCS_TRIG_SERIES_1 | - WM8993_DCS_TRIG_SERIES_0); - - wait_for_dc_servo(codec); + wait_for_dc_servo(codec, + WM8993_DCS_TRIG_SERIES_0 | WM8993_DCS_TRIG_SERIES_1); /* Apply correction to DC servo result */ if (hubs->dcs_codes) { @@ -145,13 +141,9 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) /* Do it */ snd_soc_write(codec, WM8993_DC_SERVO_3, dcs_cfg); - snd_soc_update_bits(codec, WM8993_DC_SERVO_0, - WM8993_DCS_TRIG_DAC_WR_0 | - WM8993_DCS_TRIG_DAC_WR_1, - WM8993_DCS_TRIG_DAC_WR_0 | - WM8993_DCS_TRIG_DAC_WR_1); - - wait_for_dc_servo(codec); + wait_for_dc_servo(codec, + WM8993_DCS_TRIG_DAC_WR_0 | + WM8993_DCS_TRIG_DAC_WR_1); } } -- cgit v1.1 From d522ffbfb9fccf6eca283cd2e8b03cf3d21fb616 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 30 Mar 2010 14:29:14 +0100 Subject: ASoC: Only do WM8994 bias off transition from standby Otherwise we may try to power down multiple times when the using idle bias off and the driver is removed. Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8994.c | 53 ++++++++++++++++++++++++++--------------------- 1 file changed, 29 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c80218f..f8355ac 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3007,34 +3007,39 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - /* Switch over to startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, - WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - (1 << WM8994_VMID_RAMP_SHIFT)); - - /* Disable main biases */ - snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, - WM8994_BIAS_ENA | WM8994_VMID_SEL_MASK, 0); + if (codec->bias_level == SND_SOC_BIAS_STANDBY) { + /* Switch over to startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + (1 << WM8994_VMID_RAMP_SHIFT)); - /* Discharge line */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_1, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH, - WM8994_LINEOUT1_DISCH | - WM8994_LINEOUT2_DISCH); + /* Disable main biases */ + snd_soc_update_bits(codec, WM8994_POWER_MANAGEMENT_1, + WM8994_BIAS_ENA | + WM8994_VMID_SEL_MASK, 0); - msleep(5); + /* Discharge line */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_1, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH, + WM8994_LINEOUT1_DISCH | + WM8994_LINEOUT2_DISCH); - /* Switch off startup biases */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM8994_BIAS_SRC | WM8994_STARTUP_BIAS_ENA | - WM8994_VMID_BUF_ENA | - WM8994_VMID_RAMP_MASK, 0); + msleep(5); + /* Switch off startup biases */ + snd_soc_update_bits(codec, WM8994_ANTIPOP_2, + WM8994_BIAS_SRC | + WM8994_STARTUP_BIAS_ENA | + WM8994_VMID_BUF_ENA | + WM8994_VMID_RAMP_MASK, 0); + } break; } codec->bias_level = level; -- cgit v1.1 From d12841827a6de120199609dadb6ff4ec99bd90ea Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Mon, 5 Apr 2010 16:30:43 +0100 Subject: ALSA: hda - Enable amplifiers on Acer Inspire 6530G After more tests it appears that EAPD needs to be enabled on both the 0x14 and 0x15 NIDs to enable the main speaker and headphone amplifiers. The maximum volume setting is now equal to what the machine achieves under other operating systems. Disabling Front or LFE playback triggers EAPD and disables the amplifier. As such, these two playback switches have been removed from the mixer. Signed-off-by: Tony Vroon Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ca93c4c..5472062 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1621,6 +1621,11 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { */ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { +/* Route to built-in subwoofer as well as speakers */ + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Bias voltage on for external mic port */ {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN | PIN_VREF80}, /* Front Mic: set to PIN_IN (empty by default) */ @@ -1632,10 +1637,12 @@ static struct hda_verb alc888_acer_aspire_6530g_verbs[] = { /* Enable speaker output */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_EAPD_BTLENABLE, 2}, /* Enable headphone output */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT | PIN_HP}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 2}, { } }; @@ -8462,9 +8469,7 @@ static struct snd_kcontrol_new alc883_acer_aspire_mixer[] = { static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), - HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, 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("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), -- cgit v1.1 From 5f712b2b73a9fc87fcc52124cfe8adefaa0c92f5 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Mon, 22 Mar 2010 10:11:15 +0100 Subject: ALSA: ASoC: move dma_data from snd_soc_dai to snd_soc_pcm_stream This fixes a memory corruption when ASoC devices are used in full-duplex mode. Specifically for pxa-ssp code, where this pointer is dynamically allocated for each direction and destroyed upon each stream start. All other platforms are fixed blindly, I couldn't even compile-test them. Sorry for any breakage I may have caused. [Note that this is a backported version for 2.6.34. Upstream commit is fd23b7dee] Signed-off-by: Daniel Mack Reported-by: Sven Neumann Reported-by: Michael Hirsch Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/atmel/atmel-pcm.c | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 6 +++--- sound/soc/davinci/davinci-i2s.c | 3 ++- sound/soc/davinci/davinci-mcasp.c | 3 ++- sound/soc/davinci/davinci-pcm.c | 4 +++- sound/soc/imx/imx-pcm-dma-mx2.c | 8 ++++++-- sound/soc/imx/imx-ssi.c | 7 +++++-- sound/soc/omap/omap-mcbsp.c | 4 +++- sound/soc/omap/omap-mcpdm.c | 3 ++- sound/soc/omap/omap-pcm.c | 4 +++- sound/soc/pxa/pxa-ssp.c | 23 +++++++++++----------- sound/soc/pxa/pxa2xx-ac97.c | 17 ++++++++++++----- sound/soc/pxa/pxa2xx-i2s.c | 7 +++++-- sound/soc/pxa/pxa2xx-pcm.c | 4 +++- sound/soc/s3c24xx/s3c-ac97.c | 21 +++++++++++--------- sound/soc/s3c24xx/s3c-dma.c | 4 +++- sound/soc/s3c24xx/s3c-i2s-v2.c | 13 ++++++++----- sound/soc/s3c24xx/s3c-pcm.c | 7 +++++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 19 ++++++++++--------- sound/soc/s6000/s6000-i2s.c | 3 ++- sound/soc/s6000/s6000-pcm.c | 40 ++++++++++++++++++++++++++++----------- 21 files changed, 131 insertions(+), 71 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 9ef6b96..3e6628c8 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -180,7 +180,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); runtime->dma_bytes = params_buffer_bytes(params); - prtd->params = rtd->dai->cpu_dai->dma_data; + prtd->params = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); prtd->params->dma_intr_handler = atmel_pcm_dma_irq; prtd->dma_buffer = runtime->dma_addr; diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index e588e63..0b59806 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -363,12 +363,12 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, ssc_p->dma_params[dir] = dma_params; /* - * The cpu_dai->dma_data field is only used to communicate the - * appropriate DMA parameters to the pcm driver hw_params() + * The snd_soc_pcm_stream->dma_data field is only used to communicate + * the appropriate DMA parameters to the pcm driver hw_params() * function. It should not be used for other purposes * as it is common to all substreams. */ - rtd->dai->cpu_dai->dma_data = dma_params; + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_params); channels = params_channels(params); diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 6362ca0..4aad7ecc 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -585,7 +585,8 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->dma_params[SNDRV_PCM_STREAM_CAPTURE].channel = res->start; davinci_i2s_dai.private_data = dev; - davinci_i2s_dai.dma_data = dev->dma_params; + davinci_i2s_dai.capture.dma_data = dev->dma_params; + davinci_i2s_dai.playback.dma_data = dev->dma_params; ret = snd_soc_register_dai(&davinci_i2s_dai); if (ret != 0) goto err_free_mem; diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ab6518d..c056bfb 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -917,7 +917,8 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dma_data->channel = res->start; davinci_mcasp_dai[pdata->op_mode].private_data = dev; - davinci_mcasp_dai[pdata->op_mode].dma_data = dev->dma_params; + davinci_mcasp_dai[pdata->op_mode].capture.dma_data = dev->dma_params; + davinci_mcasp_dai[pdata->op_mode].playback.dma_data = dev->dma_params; davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 80c7fdf..2dc406f 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -649,8 +649,10 @@ static int davinci_pcm_open(struct snd_pcm_substream *substream) struct snd_pcm_hardware *ppcm; int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *pa = rtd->dai->cpu_dai->dma_data; + struct davinci_pcm_dma_params *pa; struct davinci_pcm_dma_params *params; + + pa = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); if (!pa) return -ENODEV; params = &pa[substream->stream]; diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 19452e4..c78c000 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -83,11 +83,13 @@ static void snd_imx_dma_err_callback(int channel, void *data, int err) static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_dma_params *dma_params; struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; int ret; + dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream); + iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH); if (iprtd->dma < 0) { pr_err("Failed to claim the audio DMA\n"); @@ -192,10 +194,12 @@ static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_dma_params *dma_params; struct imx_pcm_runtime_data *iprtd = runtime->private_data; int err; + dma_params = snd_soc_get_dma_data(rtd->dai->cpu_dai, substream); + iprtd->substream = substream; iprtd->buf = (unsigned int *)substream->dma_buffer.area; iprtd->period_cnt = 0; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 56f46a7..28e55c7 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -234,17 +234,20 @@ static int imx_ssi_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct imx_ssi *ssi = cpu_dai->private_data; + struct imx_pcm_dma_params *dma_data; u32 reg, sccr; /* Tx/Rx config */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { reg = SSI_STCCR; - cpu_dai->dma_data = &ssi->dma_params_tx; + dma_data = &ssi->dma_params_tx; } else { reg = SSI_SRCCR; - cpu_dai->dma_data = &ssi->dma_params_rx; + dma_data = &ssi->dma_params_rx; } + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); + sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; /* DAI data (word) size */ diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index e814a95..8ad9dc9 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -297,7 +297,9 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; omap_mcbsp_dai_dma_params[id][substream->stream].data_type = OMAP_DMA_DATA_TYPE_S16; - cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream]; + + snd_soc_dai_set_dma_data(cpu_dai, substream, + &omap_mcbsp_dai_dma_params[id][substream->stream]); if (mcbsp_data->configured) { /* McBSP already configured by another stream */ diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 25f19e4..b7f4f7e 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -150,7 +150,8 @@ static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream, int stream = substream->stream; int channels, err, link_mask = 0; - cpu_dai->dma_data = &omap_mcpdm_dai_dma_params[stream]; + snd_soc_dai_set_dma_data(cpu_dai, substream, + &omap_mcpdm_dai_dma_params[stream]); channels = params_channels(params); switch (channels) { diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index bdd1097..3945644 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -99,9 +99,11 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct omap_runtime_data *prtd = runtime->private_data; - struct omap_pcm_dma_data *dma_data = rtd->dai->cpu_dai->dma_data; + struct omap_pcm_dma_data *dma_data; int err = 0; + dma_data = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma_data) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 9e95e51..6959c51 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -121,10 +121,9 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, ssp_disable(ssp); } - if (cpu_dai->dma_data) { - kfree(cpu_dai->dma_data); - cpu_dai->dma_data = NULL; - } + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); + snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); + return ret; } @@ -141,10 +140,8 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, clk_disable(ssp->clk); } - if (cpu_dai->dma_data) { - kfree(cpu_dai->dma_data); - cpu_dai->dma_data = NULL; - } + kfree(snd_soc_dai_get_dma_data(cpu_dai, substream)); + snd_soc_dai_set_dma_data(cpu_dai, substream, NULL); } #ifdef CONFIG_PM @@ -569,19 +566,23 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, u32 sspsp; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; + struct pxa2xx_pcm_dma_params *dma_data; + + dma_data = snd_soc_dai_get_dma_data(dai, substream); /* generate correct DMA params */ - if (cpu_dai->dma_data) - kfree(cpu_dai->dma_data); + kfree(dma_data); /* Network mode with one active slot (ttsa == 1) can be used * to force 16-bit frame width on the wire (for S16_LE), even * with two channels. Use 16-bit DMA transfers for this case. */ - cpu_dai->dma_data = ssp_get_dma_params(ssp, + dma_data = ssp_get_dma_params(ssp, ((chn == 2) && (ttsa != 1)) || (width == 32), substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + snd_soc_dai_set_dma_data(dai, substream, dma_data); + /* we can only change the settings if the port is not in use */ if (ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) return 0; diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index e9ae7b3..d314115 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -122,11 +122,14 @@ static int pxa2xx_ac97_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_out; + dma_data = &pxa2xx_ac97_pcm_stereo_out; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_stereo_in; + dma_data = &pxa2xx_ac97_pcm_stereo_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -137,11 +140,14 @@ static int pxa2xx_ac97_hw_aux_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_out; + dma_data = &pxa2xx_ac97_pcm_aux_mono_out; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_aux_mono_in; + dma_data = &pxa2xx_ac97_pcm_aux_mono_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -156,7 +162,8 @@ static int pxa2xx_ac97_hw_mic_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; else - cpu_dai->dma_data = &pxa2xx_ac97_pcm_mic_mono_in; + snd_soc_dai_set_dma_data(cpu_dai, substream, + &pxa2xx_ac97_pcm_mic_mono_in); return 0; } diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 6b8f655..c1a52757 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -164,6 +164,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct pxa2xx_pcm_dma_params *dma_data; BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); @@ -171,9 +172,11 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, pxa_i2s_wait(); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_out; + dma_data = &pxa2xx_i2s_pcm_stereo_out; else - cpu_dai->dma_data = &pxa2xx_i2s_pcm_stereo_in; + dma_data = &pxa2xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); /* is port used by another stream */ if (!(SACR0 & SACR0_ENB)) { diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index d38e395..adc7e6f 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -25,9 +25,11 @@ static int pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct pxa2xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct pxa2xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data; + struct pxa2xx_pcm_dma_params *dma; int ret; + dma = snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); + /* return if this is a bufferless transfer e.g. * codec <--> BT codec or GSM modem -- lg FIXME */ if (!dma) diff --git a/sound/soc/s3c24xx/s3c-ac97.c b/sound/soc/s3c24xx/s3c-ac97.c index ee8ed9d7..ecf4fd0 100644 --- a/sound/soc/s3c24xx/s3c-ac97.c +++ b/sound/soc/s3c24xx/s3c-ac97.c @@ -224,11 +224,14 @@ static int s3c_ac97_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct s3c_dma_params *dma_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - cpu_dai->dma_data = &s3c_ac97_pcm_out; + dma_data = &s3c_ac97_pcm_out; else - cpu_dai->dma_data = &s3c_ac97_pcm_in; + dma_data = &s3c_ac97_pcm_in; + + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); return 0; } @@ -238,8 +241,8 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) @@ -265,7 +268,7 @@ static int s3c_ac97_trigger(struct snd_pcm_substream *substream, int cmd, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); return 0; } @@ -280,7 +283,7 @@ static int s3c_ac97_hw_mic_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; else - cpu_dai->dma_data = &s3c_ac97_mic_in; + snd_soc_dai_set_dma_data(cpu_dai, substream, &s3c_ac97_mic_in); return 0; } @@ -290,8 +293,8 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, { u32 ac_glbctrl; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); ac_glbctrl = readl(s3c_ac97.regs + S3C_AC97_GLBCTRL); ac_glbctrl &= ~S3C_AC97_GLBCTRL_MICINTM_MASK; @@ -311,7 +314,7 @@ static int s3c_ac97_mic_trigger(struct snd_pcm_substream *substream, writel(ac_glbctrl, s3c_ac97.regs + S3C_AC97_GLBCTRL); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); return 0; } diff --git a/sound/soc/s3c24xx/s3c-dma.c b/sound/soc/s3c24xx/s3c-dma.c index 7725e26..1b61c23 100644 --- a/sound/soc/s3c24xx/s3c-dma.c +++ b/sound/soc/s3c24xx/s3c-dma.c @@ -145,10 +145,12 @@ static int s3c_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct s3c24xx_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct s3c_dma_params *dma = rtd->dai->cpu_dai->dma_data; unsigned long totbytes = params_buffer_bytes(params); + struct s3c_dma_params *dma = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); int ret = 0; + pr_debug("Entered %s\n", __func__); /* return if this is a bufferless transfer e.g. diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index e994d83..8851594 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -339,14 +339,17 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai = rtd->dai; struct s3c_i2sv2_info *i2s = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->cpu_dai->dma_data = i2s->dma_playback; + dma_data = i2s->dma_playback; else - dai->cpu_dai->dma_data = i2s->dma_capture; + dma_data = i2s->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); /* Working copies of register */ iismod = readl(i2s->regs + S3C2412_IISMOD); @@ -394,8 +397,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -431,7 +434,7 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, * of the auto reload mechanism of S3C24XX. * This call won't bother S3C64XX. */ - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; diff --git a/sound/soc/s3c24xx/s3c-pcm.c b/sound/soc/s3c24xx/s3c-pcm.c index a98f40c..326f0a9 100644 --- a/sound/soc/s3c24xx/s3c-pcm.c +++ b/sound/soc/s3c24xx/s3c-pcm.c @@ -178,6 +178,7 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai_link *dai = rtd->dai; struct s3c_pcm_info *pcm = to_info(dai->cpu_dai); + struct s3c_dma_params *dma_data; void __iomem *regs = pcm->regs; struct clk *clk; int sclk_div, sync_div; @@ -187,9 +188,11 @@ static int s3c_pcm_hw_params(struct snd_pcm_substream *substream, dev_dbg(pcm->dev, "Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dai->cpu_dai->dma_data = pcm->dma_playback; + dma_data = pcm->dma_playback; else - dai->cpu_dai->dma_data = pcm->dma_capture; + dma_data = pcm->dma_capture; + + snd_soc_dai_set_dma_data(dai->cpu_dai, substream, dma_data); /* Strictly check for sample size */ switch (params_format(params)) { diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 0bc5950..c3ac890 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -242,14 +242,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_out; + dma_data = &s3c24xx_i2s_pcm_stereo_out; else - rtd->dai->cpu_dai->dma_data = &s3c24xx_i2s_pcm_stereo_in; + dma_data = &s3c24xx_i2s_pcm_stereo_in; + + snd_soc_dai_set_dma_data(rtd->dai->cpu_dai, substream, dma_data); /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -258,13 +261,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->dma_size = 1; + dma_data->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->dma_size = 2; + dma_data->dma_size = 2; break; default: return -EINVAL; @@ -280,8 +281,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; - int channel = ((struct s3c_dma_params *) - rtd->dai->cpu_dai->dma_data)->channel; + struct s3c_dma_params *dma_data = + snd_soc_dai_get_dma_data(rtd->dai->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -300,7 +301,7 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED); + s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: diff --git a/sound/soc/s6000/s6000-i2s.c b/sound/soc/s6000/s6000-i2s.c index c5cda18..fa23854 100644 --- a/sound/soc/s6000/s6000-i2s.c +++ b/sound/soc/s6000/s6000-i2s.c @@ -518,7 +518,8 @@ static int __devinit s6000_i2s_probe(struct platform_device *pdev) s6000_i2s_dai.dev = &pdev->dev; s6000_i2s_dai.private_data = dev; - s6000_i2s_dai.dma_data = &dev->dma_params; + s6000_i2s_dai.capture.dma_data = &dev->dma_params; + s6000_i2s_dai.playback.dma_data = &dev->dma_params; dev->sifbase = sifmem->start; dev->scbbase = mmio; diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 1d61109..9c7f7f0 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -58,13 +58,15 @@ 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; + struct s6000_pcm_dma_params *par; int channel; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; dma_addr_t src, dst; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + period_size = snd_pcm_lib_period_bytes(substream); dma_offset = prtd->period * period_size; dma_pos = runtime->dma_addr + dma_offset; @@ -101,7 +103,8 @@ 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_pcm_dma_params *params = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); struct s6000_runtime_data *prtd; unsigned int has_xrun; int i, ret = IRQ_NONE; @@ -172,11 +175,13 @@ 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; + struct s6000_pcm_dma_params *par; unsigned long flags; int srcinc; u32 dma; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + spin_lock_irqsave(&prtd->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { @@ -212,10 +217,12 @@ 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; + struct s6000_pcm_dma_params *par; unsigned long flags; u32 channel; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) channel = par->dma_out; else @@ -236,9 +243,11 @@ static int s6000_pcm_stop(struct snd_pcm_substream *substream) 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; + struct s6000_pcm_dma_params *par; int ret; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + ret = par->trigger(substream, cmd, 0); if (ret < 0) return ret; @@ -275,13 +284,15 @@ static int s6000_pcm_prepare(struct snd_pcm_substream *substream) 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 s6000_pcm_dma_params *par; 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; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + spin_lock_irqsave(&prtd->lock, flags); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) @@ -305,11 +316,12 @@ static snd_pcm_uframes_t s6000_pcm_pointer(struct snd_pcm_substream *substream) 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 s6000_pcm_dma_params *par; struct snd_pcm_runtime *runtime = substream->runtime; struct s6000_runtime_data *prtd; int ret; + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); snd_soc_set_runtime_hwparams(substream, &s6000_pcm_hardware); ret = snd_pcm_hw_constraint_step(runtime, 0, @@ -364,7 +376,7 @@ 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; + struct s6000_pcm_dma_params *par; int ret; ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); @@ -373,6 +385,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, return ret; } + par = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (par->same_rate) { spin_lock(&par->lock); if (par->rate == -1 || @@ -392,7 +406,8 @@ static int s6000_pcm_hw_params(struct snd_pcm_substream *substream, 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; + struct s6000_pcm_dma_params *par = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); spin_lock(&par->lock); par->in_use &= ~(1 << substream->stream); @@ -417,7 +432,8 @@ static struct snd_pcm_ops s6000_pcm_ops = { 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; + struct s6000_pcm_dma_params *params = + snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); free_irq(params->irq, pcm); snd_pcm_lib_preallocate_free_for_all(pcm); @@ -429,9 +445,11 @@ 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; + struct s6000_pcm_dma_params *params; int res; + params = snd_soc_dai_get_dma_data(soc_runtime->dai->cpu_dai, substream); + if (!card->dev->dma_mask) card->dev->dma_mask = &s6000_pcm_dmamask; if (!card->dev->coherent_dma_mask) -- cgit v1.1 From f9700d5a4575e7fb343df10a1d29d425e4b81082 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 5 Apr 2010 23:25:13 +0200 Subject: ALSA: hda - Fix a wrong array range check in patch_realtek.c The commit 6a4f2ccb467e00281470cde2dee08fe5ecde62d1 introduced a wrong comparision for the array range check, which effectively skips the whole initialization of DAC connections. Fixed now. Reference: bko#15689 https://bugzilla.kernel.org/show_bug.cgi?id=15689 Reported-by: Adrian Ulrich Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5472062..c7730db 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10110,13 +10110,12 @@ static void alc882_auto_set_output_and_unmute(struct hda_codec *codec, int idx; alc_set_pin_output(codec, nid, pin_type); + if (dac_idx >= spec->multiout.num_dacs) + return; if (spec->multiout.dac_nids[dac_idx] == 0x25) idx = 4; - else { - if (spec->multiout.num_dacs >= dac_idx) - return; + else idx = spec->multiout.dac_nids[dac_idx] - 2; - } snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, idx); } -- cgit v1.1 From b0cc58a25d04160d39a80e436847eaa2fbc5aa09 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Tue, 6 Apr 2010 19:31:26 +0300 Subject: ALSA: mixart: range checking proc file The original code doesn't take into consideration that the value of MIXART_BA0_SIZE - pos can be less than zero which would lead to a large unsigned value for "count". Also I moved the check that read size is a multiple of 4 bytes below the code that adjusts "count". Signed-off-by: Dan Carpenter Cc: Acked-by: Linus Torvalds Signed-off-by: Takashi Iwai --- sound/pci/mixart/mixart.c | 24 ++++++++++++++---------- 1 file changed, 14 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/mixart/mixart.c b/sound/pci/mixart/mixart.c index 7e8e7da..ea4256b 100644 --- a/sound/pci/mixart/mixart.c +++ b/sound/pci/mixart/mixart.c @@ -1161,13 +1161,15 @@ static long snd_mixart_BA0_read(struct snd_info_entry *entry, void *file_private unsigned long count, unsigned long pos) { struct mixart_mgr *mgr = entry->private_data; + unsigned long maxsize; - count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ - if(count <= 0) + if (pos >= MIXART_BA0_SIZE) return 0; - if(pos + count > MIXART_BA0_SIZE) - count = (long)(MIXART_BA0_SIZE - pos); - if(copy_to_user_fromio(buf, MIXART_MEM( mgr, pos ), count)) + maxsize = MIXART_BA0_SIZE - pos; + if (count > maxsize) + count = maxsize; + count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ + if (copy_to_user_fromio(buf, MIXART_MEM(mgr, pos), count)) return -EFAULT; return count; } @@ -1180,13 +1182,15 @@ static long snd_mixart_BA1_read(struct snd_info_entry *entry, void *file_private unsigned long count, unsigned long pos) { struct mixart_mgr *mgr = entry->private_data; + unsigned long maxsize; - count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ - if(count <= 0) + if (pos > MIXART_BA1_SIZE) return 0; - if(pos + count > MIXART_BA1_SIZE) - count = (long)(MIXART_BA1_SIZE - pos); - if(copy_to_user_fromio(buf, MIXART_REG( mgr, pos ), count)) + maxsize = MIXART_BA1_SIZE - pos; + if (count > maxsize) + count = maxsize; + count = count & ~3; /* make sure the read size is a multiple of 4 bytes */ + if (copy_to_user_fromio(buf, MIXART_REG(mgr, pos), count)) return -EFAULT; return count; } -- cgit v1.1