From f15cbe6f1a4b4d9df59142fc8e4abb973302cf44 Mon Sep 17 00:00:00 2001 From: Paul Mundt Date: Tue, 29 Jul 2008 08:09:44 +0900 Subject: sh: migrate to arch/sh/include/ This follows the sparc changes a439fe51a1f8eb087c22dd24d69cebae4a3addac. Most of the moving about was done with Sam's directions at: http://marc.info/?l=linux-sh&m=121724823706062&w=2 with subsequent hacking and fixups entirely my fault. Signed-off-by: Sam Ravnborg Signed-off-by: Paul Mundt --- sound/sh/aica.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 9ca113326143..54df8baf916f 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -42,7 +42,7 @@ #include #include #include -#include +#include #include "aica.h" MODULE_AUTHOR("Adrian McMenamin "); -- cgit v1.2.2 From a7b815169aae65072017efb1fba9dcecc82ba7c1 Mon Sep 17 00:00:00 2001 From: Huang Weiyi Date: Sat, 26 Jul 2008 20:43:01 +0800 Subject: ALSA: sound/soc/pxa/tosa.c: removed duplicated include Removed duplicated include in sound/soc/pxa/tosa.c. Signed-off-by: Huang Weiyi Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/tosa.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index fe6cca9c9e76..22971a0f040e 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -33,7 +33,6 @@ #include #include #include -#include #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" -- cgit v1.2.2 From be41e941d5f1a48bde7f44d09d56e8d2605f98e1 Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Mon, 28 Jul 2008 17:04:39 -0500 Subject: ALSA: asoc: restrict sample rate and size in Freescale MPC8610 sound drivers The Freescale MPC8610 SSI device has the option of using one clock for both transmit and receive (synchronous mode), or independent clocks (asynchronous). The SSI driver, however, programs the SSI into synchronous mode and then tries to program the clock registers independently. The result is that the wrong sample size is usually generated during recording. This patch fixes the discrepancy by restricting the sample rate and sample size of the playback and capture streams. The SSI driver remembers which stream is opened first. When a second stream is opened, that stream is constrained to the same sample rate and size as the first stream. A future version of this driver will lift the sample size restriction. Supporting independent sample rates is more difficult, because only certain codecs provide dual independent clocks. Signed-off-by: Timur Tabi Signed-off-by: Takashi Iwai --- sound/soc/fsl/fsl_dma.c | 7 ++++- sound/soc/fsl/fsl_ssi.c | 74 ++++++++++++++++++++++++++++++++++++++++++------- 2 files changed, 70 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index da2bc5902864..7ceea2bba1f5 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -132,12 +132,17 @@ struct fsl_dma_private { * Since each link descriptor has a 32-bit byte count field, we set * period_bytes_max to the largest 32-bit number. We also have no maximum * number of periods. + * + * Note that we specify SNDRV_PCM_INFO_JOINT_DUPLEX here, but only because a + * limitation in the SSI driver requires the sample rates for playback and + * capture to be the same. */ static const struct snd_pcm_hardware fsl_dma_hardware = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID, + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_JOINT_DUPLEX, .formats = FSLDMA_PCM_FORMATS, .rates = FSLDMA_PCM_RATES, .rate_min = 5512, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 71bff33f5528..157a7895ffa1 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -67,6 +67,8 @@ * @ssi: pointer to the SSI's registers * @ssi_phys: physical address of the SSI registers * @irq: IRQ of this SSI + * @first_stream: pointer to the stream that was opened first + * @second_stream: pointer to second stream * @dev: struct device pointer * @playback: the number of playback streams opened * @capture: the number of capture streams opened @@ -79,6 +81,8 @@ struct fsl_ssi_private { struct ccsr_ssi __iomem *ssi; dma_addr_t ssi_phys; unsigned int irq; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; struct device *dev; unsigned int playback; unsigned int capture; @@ -342,6 +346,49 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream) */ } + if (!ssi_private->first_stream) + ssi_private->first_stream = substream; + else { + /* This is the second stream open, so we need to impose sample + * rate and maybe sample size constraints. Note that this can + * cause a race condition if the second stream is opened before + * the first stream is fully initialized. + * + * We provide some protection by checking to make sure the first + * stream is initialized, but it's not perfect. ALSA sometimes + * re-initializes the driver with a different sample rate or + * size. If the second stream is opened before the first stream + * has received its final parameters, then the second stream may + * be constrained to the wrong sample rate or size. + * + * FIXME: This code does not handle opening and closing streams + * repeatedly. If you open two streams and then close the first + * one, you may not be able to open another stream until you + * close the second one as well. + */ + struct snd_pcm_runtime *first_runtime = + ssi_private->first_stream->runtime; + + if (!first_runtime->rate || !first_runtime->sample_bits) { + dev_err(substream->pcm->card->dev, + "set sample rate and size in %s stream first\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "capture" : "playback"); + return -EAGAIN; + } + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, + first_runtime->rate, first_runtime->rate); + + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + first_runtime->sample_bits, + first_runtime->sample_bits); + + ssi_private->second_stream = substream; + } + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ssi_private->playback++; @@ -371,18 +418,16 @@ static int fsl_ssi_prepare(struct snd_pcm_substream *substream) struct fsl_ssi_private *ssi_private = rtd->dai->cpu_dai->private_data; struct ccsr_ssi __iomem *ssi = ssi_private->ssi; - u32 wl; - wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); + if (substream == ssi_private->first_stream) { + u32 wl; - clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + /* The SSI should always be disabled at this points (SSIEN=0) */ + wl = CCSR_SSI_SxCCR_WL(snd_pcm_format_width(runtime->format)); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + /* In synchronous mode, the SSI uses STCCR for capture */ clrsetbits_be32(&ssi->stccr, CCSR_SSI_SxCCR_WL_MASK, wl); - else - clrsetbits_be32(&ssi->srccr, CCSR_SSI_SxCCR_WL_MASK, wl); - - setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + } return 0; } @@ -407,9 +452,13 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - setbits32(&ssi->scr, CCSR_SSI_SCR_TE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); } else { - setbits32(&ssi->scr, CCSR_SSI_SCR_RE); + clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); + setbits32(&ssi->scr, + CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); /* * I think we need this delay to allow time for the SSI @@ -452,6 +501,11 @@ static void fsl_ssi_shutdown(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) ssi_private->capture--; + if (ssi_private->first_stream == substream) + ssi_private->first_stream = ssi_private->second_stream; + + ssi_private->second_stream = NULL; + /* * If this is the last active substream, disable the SSI and release * the IRQ. -- cgit v1.2.2 From 877db3c1af24a65f78ae865b1fb642165e065a8b Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Tue, 29 Jul 2008 11:42:22 +0100 Subject: ALSA: ASoC: Update Poodle to current ASoC API Signed-off-by: Dmitry Baryshkov Cc: Richard Purdie Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/poodle.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 65a4e9a8c39e..d968cf71b569 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -85,17 +85,13 @@ static int poodle_startup(struct snd_pcm_substream *substream) } /* we need to unmute the HP at shutdown as the mute burns power on poodle */ -static int poodle_shutdown(struct snd_pcm_substream *substream) +static void poodle_shutdown(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->socdev->codec; - /* set = unmute headphone */ locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_L, 1); locomo_gpio_write(&poodle_locomo_device.dev, POODLE_LOCOMO_GPIO_MUTE_R, 1); - return 0; } static int poodle_hw_params(struct snd_pcm_substream *substream, @@ -232,7 +228,7 @@ static const struct soc_enum poodle_enum[] = { SOC_ENUM_SINGLE_EXT(2, spk_function), }; -static const snd_kcontrol_new_t wm8731_poodle_controls[] = { +static const struct snd_kcontrol_new wm8731_poodle_controls[] = { SOC_ENUM_EXT("Jack Function", poodle_enum[0], poodle_get_jack, poodle_set_jack), SOC_ENUM_EXT("Speaker Function", poodle_enum[1], poodle_get_spk, -- cgit v1.2.2 From 11589418a1c4cf68be9367f802898d35e07809c4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 29 Jul 2008 11:42:23 +0100 Subject: ALSA: ASoC: Export dapm_reg_event() fully dapm_reg_event() is used by devices using SND_SOC_DAPM_REG() so needs to be exported to support building them as modules and prototyped to avoid sparse warnings and potential build issues. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/soc-dapm.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 820347c9ae4b..f9d100bc8479 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -470,6 +470,7 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, return 0; } +EXPORT_SYMBOL_GPL(dapm_reg_event); /* * Scan each dapm widget for complete audio path. -- cgit v1.2.2 From bf9c8c9ddef7ef761ae9747349175adad0ef16ce Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Fri, 1 Aug 2008 14:58:44 -0500 Subject: ALSA: ASoC: fix SNDCTL_DSP_SYNC support in Freescale 8610 sound drivers If an OSS application calls SNDCTL_DSP_SYNC, then ALSA will call the driver's _hw_params and _prepare functions again. On the Freescale MPC8610 DMA ASoC driver, this caused the DMA controller to be unneccessarily re-programmed, and apparently it doesn't like that. The DMA will then not operate when instructed. This patch relocates much of the DMA programming to fsl_dma_open(), which is called only once. Signed-off-by: Timur Tabi Signed-off-by: Takashi Iwai --- sound/soc/fsl/fsl_dma.c | 235 +++++++++++++++++++++++++----------------------- 1 file changed, 124 insertions(+), 111 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 7ceea2bba1f5..d2d3da9729f2 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -327,14 +327,75 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, * fsl_dma_open: open a new substream. * * Each substream has its own DMA buffer. + * + * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link + * descriptors that ping-pong from one period to the next. For example, if + * there are six periods and two link descriptors, this is how they look + * before playback starts: + * + * The last link descriptor + * ____________ points back to the first + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * | | + * V V + * _________________________________________ + * | | | | | | | The DMA buffer is + * | | | | | | | divided into 6 parts + * |______|______|______|______|______|______| + * + * and here's how they look after the first period is finished playing: + * + * ____________ + * | | + * V | + * ___ ___ | + * | |->| |->| + * |___| |___| + * | | + * |______________ + * | | + * V V + * _________________________________________ + * | | | | | | | + * | | | | | | | + * |______|______|______|______|______|______| + * + * The first link descriptor now points to the third period. The DMA + * controller is currently playing the second period. When it finishes, it + * will jump back to the first descriptor and play the third period. + * + * There are four reasons we do this: + * + * 1. The only way to get the DMA controller to automatically restart the + * transfer when it gets to the end of the buffer is to use chaining + * mode. Basic direct mode doesn't offer that feature. + * 2. We need to receive an interrupt at the end of every period. The DMA + * controller can generate an interrupt at the end of every link transfer + * (aka segment). Making each period into a DMA segment will give us the + * interrupts we need. + * 3. By creating only two link descriptors, regardless of the number of + * periods, we do not need to reallocate the link descriptors if the + * number of periods changes. + * 4. All of the audio data is still stored in a single, contiguous DMA + * buffer, which is what ALSA expects. We're just dividing it into + * contiguous parts, and creating a link descriptor for each one. */ static int fsl_dma_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private; + struct ccsr_dma_channel __iomem *dma_channel; dma_addr_t ld_buf_phys; + u64 temp_link; /* Pointer to next link descriptor */ + u32 mr; unsigned int channel; int ret = 0; + unsigned int i; /* * Reject any DMA buffer whose size is not a multiple of the period @@ -395,68 +456,74 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) snd_soc_set_runtime_hwparams(substream, &fsl_dma_hardware); runtime->private_data = dma_private; + /* Program the fixed DMA controller parameters */ + + dma_channel = dma_private->dma_channel; + + temp_link = dma_private->ld_buf_phys + + sizeof(struct fsl_dma_link_descriptor); + + for (i = 0; i < NUM_DMA_LINKS; i++) { + struct fsl_dma_link_descriptor *link = &dma_private->link[i]; + + link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); + link->next = cpu_to_be64(temp_link); + + temp_link += sizeof(struct fsl_dma_link_descriptor); + } + /* The last link descriptor points to the first */ + dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); + + /* Tell the DMA controller where the first link descriptor is */ + out_be32(&dma_channel->clndar, + CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); + out_be32(&dma_channel->eclndar, + CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); + + /* The manual says the BCR must be clear before enabling EMP */ + out_be32(&dma_channel->bcr, 0); + + /* + * Program the mode register for interrupts, external master control, + * and source/destination hold. Also clear the Channel Abort bit. + */ + mr = in_be32(&dma_channel->mr) & + ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); + + /* + * We want External Master Start and External Master Pause enabled, + * because the SSI is controlling the DMA controller. We want the DMA + * controller to be set up in advance, and then we signal only the SSI + * to start transferring. + * + * We want End-Of-Segment Interrupts enabled, because this will generate + * an interrupt at the end of each segment (each link descriptor + * represents one segment). Each DMA segment is the same thing as an + * ALSA period, so this is how we get an interrupt at the end of every + * period. + * + * We want Error Interrupt enabled, so that we can get an error if + * the DMA controller is mis-programmed somehow. + */ + mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | + CCSR_DMA_MR_EMS_EN; + + /* For playback, we want the destination address to be held. For + capture, set the source address to be held. */ + mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? + CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; + + out_be32(&dma_channel->mr, mr); + return 0; } /** - * fsl_dma_hw_params: allocate the DMA buffer and the DMA link descriptors. - * - * ALSA divides the DMA buffer into N periods. We create NUM_DMA_LINKS link - * descriptors that ping-pong from one period to the next. For example, if - * there are six periods and two link descriptors, this is how they look - * before playback starts: - * - * The last link descriptor - * ____________ points back to the first - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * | | - * V V - * _________________________________________ - * | | | | | | | The DMA buffer is - * | | | | | | | divided into 6 parts - * |______|______|______|______|______|______| - * - * and here's how they look after the first period is finished playing: - * - * ____________ - * | | - * V | - * ___ ___ | - * | |->| |->| - * |___| |___| - * | | - * |______________ - * | | - * V V - * _________________________________________ - * | | | | | | | - * | | | | | | | - * |______|______|______|______|______|______| + * fsl_dma_hw_params: continue initializing the DMA links * - * The first link descriptor now points to the third period. The DMA - * controller is currently playing the second period. When it finishes, it - * will jump back to the first descriptor and play the third period. - * - * There are four reasons we do this: - * - * 1. The only way to get the DMA controller to automatically restart the - * transfer when it gets to the end of the buffer is to use chaining - * mode. Basic direct mode doesn't offer that feature. - * 2. We need to receive an interrupt at the end of every period. The DMA - * controller can generate an interrupt at the end of every link transfer - * (aka segment). Making each period into a DMA segment will give us the - * interrupts we need. - * 3. By creating only two link descriptors, regardless of the number of - * periods, we do not need to reallocate the link descriptors if the - * number of periods changes. - * 4. All of the audio data is still stored in a single, contiguous DMA - * buffer, which is what ALSA expects. We're just dividing it into - * contiguous parts, and creating a link descriptor for each one. + * This function obtains hardware parameters about the opened stream and + * programs the DMA controller accordingly. * * Note that due to a quirk of the SSI's STX register, the target address * for the DMA operations depends on the sample size. So we don't program @@ -468,11 +535,8 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime = substream->runtime; struct fsl_dma_private *dma_private = runtime->private_data; - struct ccsr_dma_channel __iomem *dma_channel = dma_private->dma_channel; dma_addr_t temp_addr; /* Pointer to next period */ - u64 temp_link; /* Pointer to next link descriptor */ - u32 mr; /* Temporary variable for MR register */ unsigned int i; @@ -490,8 +554,6 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, dma_private->dma_buf_next = dma_private->dma_buf_phys; /* - * Initialize each link descriptor. - * * The actual address in STX0 (destination for playback, source for * capture) is based on the sample size, but we don't know the sample * size in this function, so we'll have to adjust that later. See @@ -507,16 +569,11 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, * buffer itself. */ temp_addr = substream->dma_buffer.addr; - temp_link = dma_private->ld_buf_phys + - sizeof(struct fsl_dma_link_descriptor); for (i = 0; i < NUM_DMA_LINKS; i++) { struct fsl_dma_link_descriptor *link = &dma_private->link[i]; link->count = cpu_to_be32(period_size); - link->source_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->dest_attr = cpu_to_be32(CCSR_DMA_ATR_SNOOP); - link->next = cpu_to_be64(temp_link); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) link->source_addr = cpu_to_be32(temp_addr); @@ -524,51 +581,7 @@ static int fsl_dma_hw_params(struct snd_pcm_substream *substream, link->dest_addr = cpu_to_be32(temp_addr); temp_addr += period_size; - temp_link += sizeof(struct fsl_dma_link_descriptor); } - /* The last link descriptor points to the first */ - dma_private->link[i - 1].next = cpu_to_be64(dma_private->ld_buf_phys); - - /* Tell the DMA controller where the first link descriptor is */ - out_be32(&dma_channel->clndar, - CCSR_DMA_CLNDAR_ADDR(dma_private->ld_buf_phys)); - out_be32(&dma_channel->eclndar, - CCSR_DMA_ECLNDAR_ADDR(dma_private->ld_buf_phys)); - - /* The manual says the BCR must be clear before enabling EMP */ - out_be32(&dma_channel->bcr, 0); - - /* - * Program the mode register for interrupts, external master control, - * and source/destination hold. Also clear the Channel Abort bit. - */ - mr = in_be32(&dma_channel->mr) & - ~(CCSR_DMA_MR_CA | CCSR_DMA_MR_DAHE | CCSR_DMA_MR_SAHE); - - /* - * We want External Master Start and External Master Pause enabled, - * because the SSI is controlling the DMA controller. We want the DMA - * controller to be set up in advance, and then we signal only the SSI - * to start transfering. - * - * We want End-Of-Segment Interrupts enabled, because this will generate - * an interrupt at the end of each segment (each link descriptor - * represents one segment). Each DMA segment is the same thing as an - * ALSA period, so this is how we get an interrupt at the end of every - * period. - * - * We want Error Interrupt enabled, so that we can get an error if - * the DMA controller is mis-programmed somehow. - */ - mr |= CCSR_DMA_MR_EOSIE | CCSR_DMA_MR_EIE | CCSR_DMA_MR_EMP_EN | - CCSR_DMA_MR_EMS_EN; - - /* For playback, we want the destination address to be held. For - capture, set the source address to be held. */ - mr |= (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? - CCSR_DMA_MR_DAHE : CCSR_DMA_MR_SAHE; - - out_be32(&dma_channel->mr, mr); return 0; } -- cgit v1.2.2 From 82e68f7ffec3800425f2391c8c86277606860442 Mon Sep 17 00:00:00 2001 From: Willy Tarreau Date: Sat, 2 Aug 2008 18:25:16 +0200 Subject: sound: ensure device number is valid in snd_seq_oss_synth_make_info snd_seq_oss_synth_make_info() incorrectly reports information to userspace without first checking for the validity of the device number, leading to possible information leak (CVE-2008-3272). Reported-By: Tobias Klein Acked-and-tested-by: Takashi Iwai Cc: stable@kernel.org Signed-off-by: Willy Tarreau Signed-off-by: Linus Torvalds --- sound/core/seq/oss/seq_oss_synth.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss_synth.c b/sound/core/seq/oss/seq_oss_synth.c index 558dadbf45f1..e024e4588b82 100644 --- a/sound/core/seq/oss/seq_oss_synth.c +++ b/sound/core/seq/oss/seq_oss_synth.c @@ -604,6 +604,9 @@ snd_seq_oss_synth_make_info(struct seq_oss_devinfo *dp, int dev, struct synth_in { struct seq_oss_synth *rec; + if (dev < 0 || dev >= dp->max_synthdev) + return -ENXIO; + if (dp->synths[dev].is_midi) { struct midi_info minf; snd_seq_oss_midi_make_info(dp, dp->synths[dev].midi_mapped, &minf); -- cgit v1.2.2 From 680db0136e0778a0d7e025af7572c6a8d82279e2 Mon Sep 17 00:00:00 2001 From: Linus Torvalds Date: Wed, 6 Aug 2008 15:14:13 -0700 Subject: pcm_native.c: remove unused label This fixes the warning sound/core/pcm_native.c: In function 'snd_pcm_fasync': sound/core/pcm_native.c:3262: warning: label 'out' defined but not used Signed-off-by: Linus Torvalds --- sound/core/pcm_native.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index c49b9d9e303c..333cff68c150 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3259,7 +3259,6 @@ static int snd_pcm_fasync(int fd, struct file * file, int on) runtime = substream->runtime; err = fasync_helper(fd, file, on, &runtime->fasync); -out: unlock_kernel(); if (err < 0) return err; -- cgit v1.2.2 From 685d87f7ccc649ab92b55e18e507a65d0e694eb9 Mon Sep 17 00:00:00 2001 From: Linus Torvalds Date: Wed, 6 Aug 2008 19:24:47 -0700 Subject: Revert "pcm_native.c: remove unused label" This reverts commit 680db0136e0778a0d7e025af7572c6a8d82279e2. The label is actually used, but hidden behind CONFIG_SND_DEBUG and the horrible snd_assert() macro. That macro could probably be improved to be along the lines of #define snd_assert(expr, args...) do { if ((void)(expr),0) { args; } } while (0) or similar to make sure that we always both evaluate 'expr' and parse 'args', but while gcc should optimize it all away, I'm too lazy to really verify that. So I'll just admit defeat and will continue to live with the annoying warning. Noted-by: Robert P. J. Day Signed-off-by: Linus "Grr.." Torvalds --- sound/core/pcm_native.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 333cff68c150..c49b9d9e303c 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3259,6 +3259,7 @@ static int snd_pcm_fasync(int fd, struct file * file, int on) runtime = substream->runtime; err = fasync_helper(fd, file, on, &runtime->fasync); +out: unlock_kernel(); if (err < 0) return err; -- cgit v1.2.2 From 0f8469a54f7bd65f2c740a5480c56260dc8a7ae0 Mon Sep 17 00:00:00 2001 From: Russell King Date: Sun, 3 Aug 2008 15:06:16 +0100 Subject: [ARM] Eliminate useless includes of asm/mach-types.h There are 43 includes of asm/mach-types.h by files that don't reference anything from that file. Remove these unnecessary includes. Signed-off-by: Russell King --- sound/soc/davinci/davinci-evm.c | 1 - sound/soc/s3c24xx/neo1973_wm8753.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 5e2c306399ed..0722eebe3d6a 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -19,7 +19,6 @@ #include #include -#include #include #include diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 4d7a9aa15f1a..22e281ef639e 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -24,7 +24,6 @@ #include #include -#include #include #include #include -- cgit v1.2.2 From be509729356b7433f73df2b9a966674a437fbbc1 Mon Sep 17 00:00:00 2001 From: Russell King Date: Mon, 4 Aug 2008 10:41:28 +0100 Subject: [ARM] Remove asm/hardware.h, use asm/arch/hardware.h instead Remove includes of asm/hardware.h in addition to asm/arch/hardware.h. Then, since asm/hardware.h only exists to include asm/arch/hardware.h, update everything to directly include asm/arch/hardware.h and remove asm/hardware.h. Signed-off-by: Russell King --- sound/arm/pxa2xx-ac97.c | 2 +- sound/arm/pxa2xx-pcm.c | 2 +- sound/arm/sa11xx-uda1341.c | 2 +- sound/oss/vidc.c | 2 +- sound/oss/vidc_fill.S | 2 +- sound/oss/waveartist.c | 2 +- sound/soc/at91/eti_b1_wm8731.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 2 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c2412-i2s.c | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 15 files changed, 15 insertions(+), 15 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 5b3274b465eb..158f7b50b780 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -26,7 +26,7 @@ #include #include -#include +#include #include #include #include diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index 0ede9e4656a8..9a104e2430f5 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -21,7 +21,7 @@ #include #include -#include +#include #include #include "pxa2xx-pcm.h" diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index faeddf3ecedb..40c213e70593 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -71,7 +71,7 @@ #include #endif -#include +#include #include #include #include diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index bb4a0969f461..41cd4f25de04 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -22,7 +22,7 @@ #include #include -#include +#include #include #include #include diff --git a/sound/oss/vidc_fill.S b/sound/oss/vidc_fill.S index 01ccc074cc11..d2cb210cc402 100644 --- a/sound/oss/vidc_fill.S +++ b/sound/oss/vidc_fill.S @@ -11,7 +11,7 @@ */ #include #include -#include +#include #include .text diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index 88490418f932..d84c49787f6f 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -47,7 +47,7 @@ #include "waveartist.h" #ifdef CONFIG_ARM -#include +#include #include #endif diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index d532de954241..d61a4537e604 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -33,7 +33,7 @@ #include #include -#include +#include #include #include "../codecs/wm8731.h" diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 059af815ea0c..b45a4f199ef7 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -26,7 +26,7 @@ #include #include -#include +#include #include #include #include diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 8f96d87f7b4b..f69870f4f673 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -21,7 +21,7 @@ #include #include -#include +#include #include #include #include diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 2df03ee5819e..77708f879c96 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -22,7 +22,7 @@ #include #include -#include +#include #include #include diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 22e281ef639e..47bf9a0aab79 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -27,7 +27,7 @@ #include #include #include -#include +#include #include #include #include diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index ee4676ed1283..59c3d5355f55 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -28,7 +28,7 @@ #include #include #include -#include +#include #include #include diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 783349b7fede..a37167398ab5 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -27,7 +27,7 @@ #include #include -#include +#include #include #include #include diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 397524282b57..ff614d645e79 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -27,7 +27,7 @@ #include #include -#include +#include #include #include #include diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index cef79b34dc6f..fadd33e2a733 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -27,7 +27,7 @@ #include #include -#include +#include #include #include -- cgit v1.2.2 From a09e64fbc0094e3073dbb09c3b4bfe4ab669244b Mon Sep 17 00:00:00 2001 From: Russell King Date: Tue, 5 Aug 2008 16:14:15 +0100 Subject: [ARM] Move include/asm-arm/arch-* to arch/arm/*/include/mach This just leaves include/asm-arm/plat-* to deal with. Signed-off-by: Russell King --- sound/arm/pxa2xx-ac97.c | 8 ++++---- sound/arm/pxa2xx-pcm.c | 4 ++-- sound/arm/sa11xx-uda1341.c | 4 ++-- sound/oss/vidc.c | 2 +- sound/oss/vidc_fill.S | 2 +- sound/oss/waveartist.c | 2 +- sound/soc/at32/playpaq_wm8510.c | 4 ++-- sound/soc/at91/at91-pcm.c | 4 ++-- sound/soc/at91/at91-pcm.h | 2 +- sound/soc/at91/at91-ssc.c | 6 +++--- sound/soc/at91/eti_b1_wm8731.c | 4 ++-- sound/soc/davinci/davinci-evm.c | 2 +- sound/soc/omap/n810.c | 4 ++-- sound/soc/omap/omap-mcbsp.c | 6 +++--- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/pxa/corgi.c | 8 ++++---- sound/soc/pxa/e800_wm9712.c | 6 +++--- sound/soc/pxa/em-x270.c | 6 +++--- sound/soc/pxa/poodle.c | 8 ++++---- sound/soc/pxa/pxa2xx-ac97.c | 8 ++++---- sound/soc/pxa/pxa2xx-i2s.c | 8 ++++---- sound/soc/pxa/pxa2xx-pcm.c | 6 +++--- sound/soc/pxa/spitz.c | 8 ++++---- sound/soc/pxa/tosa.c | 8 ++++---- sound/soc/s3c24xx/neo1973_wm8753.c | 10 +++++----- sound/soc/s3c24xx/s3c2412-i2s.c | 8 ++++---- sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++++----- sound/soc/s3c24xx/s3c24xx-i2s.c | 10 +++++----- sound/soc/s3c24xx/s3c24xx-pcm.c | 6 +++--- 29 files changed, 83 insertions(+), 83 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index 158f7b50b780..199cca3366df 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -26,10 +26,10 @@ #include #include -#include -#include -#include -#include +#include +#include +#include +#include #include "pxa2xx-pcm.h" diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c index 9a104e2430f5..381094aab235 100644 --- a/sound/arm/pxa2xx-pcm.c +++ b/sound/arm/pxa2xx-pcm.c @@ -21,8 +21,8 @@ #include #include -#include -#include +#include +#include #include "pxa2xx-pcm.h" diff --git a/sound/arm/sa11xx-uda1341.c b/sound/arm/sa11xx-uda1341.c index 40c213e70593..b9c51bf8cd71 100644 --- a/sound/arm/sa11xx-uda1341.c +++ b/sound/arm/sa11xx-uda1341.c @@ -71,8 +71,8 @@ #include #endif -#include -#include +#include +#include #include #include diff --git a/sound/oss/vidc.c b/sound/oss/vidc.c index 41cd4f25de04..725fef0f59a3 100644 --- a/sound/oss/vidc.c +++ b/sound/oss/vidc.c @@ -22,7 +22,7 @@ #include #include -#include +#include #include #include #include diff --git a/sound/oss/vidc_fill.S b/sound/oss/vidc_fill.S index d2cb210cc402..bed34921d176 100644 --- a/sound/oss/vidc_fill.S +++ b/sound/oss/vidc_fill.S @@ -11,7 +11,7 @@ */ #include #include -#include +#include #include .text diff --git a/sound/oss/waveartist.c b/sound/oss/waveartist.c index d84c49787f6f..c47842fad657 100644 --- a/sound/oss/waveartist.c +++ b/sound/oss/waveartist.c @@ -47,7 +47,7 @@ #include "waveartist.h" #ifdef CONFIG_ARM -#include +#include #include #endif diff --git a/sound/soc/at32/playpaq_wm8510.c b/sound/soc/at32/playpaq_wm8510.c index fee5f8e58957..3f326219f1ec 100644 --- a/sound/soc/at32/playpaq_wm8510.c +++ b/sound/soc/at32/playpaq_wm8510.c @@ -36,8 +36,8 @@ #include #include -#include -#include +#include +#include #include "../codecs/wm8510.h" #include "at32-pcm.h" diff --git a/sound/soc/at91/at91-pcm.c b/sound/soc/at91/at91-pcm.c index d47492b2b6e5..7ab48bd25e4c 100644 --- a/sound/soc/at91/at91-pcm.c +++ b/sound/soc/at91/at91-pcm.c @@ -28,8 +28,8 @@ #include #include -#include -#include +#include +#include #include "at91-pcm.h" diff --git a/sound/soc/at91/at91-pcm.h b/sound/soc/at91/at91-pcm.h index 58d0f00a07b2..e5aada2cb102 100644 --- a/sound/soc/at91/at91-pcm.h +++ b/sound/soc/at91/at91-pcm.h @@ -19,7 +19,7 @@ #ifndef _AT91_PCM_H #define _AT91_PCM_H -#include +#include struct at91_ssc_periph { void __iomem *base; diff --git a/sound/soc/at91/at91-ssc.c b/sound/soc/at91/at91-ssc.c index 090e607f8692..5d44515e62e0 100644 --- a/sound/soc/at91/at91-ssc.c +++ b/sound/soc/at91/at91-ssc.c @@ -28,9 +28,9 @@ #include #include -#include -#include -#include +#include +#include +#include #include "at91-pcm.h" #include "at91-ssc.h" diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index d61a4537e604..b081e83766b7 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -33,8 +33,8 @@ #include #include -#include -#include +#include +#include #include "../codecs/wm8731.h" #include "at91-pcm.h" diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 0722eebe3d6a..65fdbd81a379 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -20,7 +20,7 @@ #include #include -#include +#include #include "../codecs/tlv320aic3x.h" #include "davinci-pcm.h" diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 02cec96859b8..7694621ec40b 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -29,9 +29,9 @@ #include #include -#include +#include #include -#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 00b0c9d73cd4..35310e16d7f3 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -30,9 +30,9 @@ #include #include -#include -#include -#include +#include +#include +#include #include "omap-mcbsp.h" #include "omap-pcm.h" diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index e092f3d836d0..690bfeaec4a0 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -27,7 +27,7 @@ #include #include -#include +#include #include "omap-pcm.h" static const struct snd_pcm_hardware omap_pcm_hardware = { diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index c0294464a23a..0a53f72077fd 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -25,10 +25,10 @@ #include #include -#include -#include -#include -#include +#include +#include +#include +#include #include "../codecs/wm8731.h" #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 06e8afb25277..6781c5be242f 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -21,9 +21,9 @@ #include #include -#include -#include -#include +#include +#include +#include #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/em-x270.c b/sound/soc/pxa/em-x270.c index 02dcac39cdf6..d9c3f7b28be2 100644 --- a/sound/soc/pxa/em-x270.c +++ b/sound/soc/pxa/em-x270.c @@ -30,9 +30,9 @@ #include #include -#include -#include -#include +#include +#include +#include #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index d968cf71b569..a4697f7e2921 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -26,10 +26,10 @@ #include #include -#include -#include -#include -#include +#include +#include +#include +#include #include "../codecs/wm8731.h" #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index b45a4f199ef7..d94a495bd6bd 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -26,10 +26,10 @@ #include #include -#include -#include -#include -#include +#include +#include +#include +#include #include "pxa2xx-pcm.h" #include "pxa2xx-ac97.h" diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index f69870f4f673..8548818eea08 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -21,10 +21,10 @@ #include #include -#include -#include -#include -#include +#include +#include +#include +#include #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 77708f879c96..4345f387fe41 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -22,9 +22,9 @@ #include #include -#include -#include -#include +#include +#include +#include #include "pxa2xx-pcm.h" diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 64385797da5d..eefc25b83514 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -26,10 +26,10 @@ #include #include -#include -#include -#include -#include +#include +#include +#include +#include #include "../codecs/wm8750.h" #include "pxa2xx-pcm.h" #include "pxa2xx-i2s.h" diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index 22971a0f040e..2baaa750f123 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -29,10 +29,10 @@ #include #include -#include -#include -#include -#include +#include +#include +#include +#include #include "../codecs/wm9712.h" #include "pxa2xx-pcm.h" diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 47bf9a0aab79..8089f8ee05c0 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -25,12 +25,12 @@ #include #include -#include -#include -#include -#include +#include +#include +#include +#include #include -#include +#include #include diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index 59c3d5355f55..ded7d995a922 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -28,16 +28,16 @@ #include #include #include -#include +#include #include #include #include -#include -#include -#include +#include +#include +#include #include "s3c24xx-pcm.h" #include "s3c2412-i2s.h" diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index a37167398ab5..19c5c3cf5d8c 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -27,13 +27,13 @@ #include #include -#include +#include #include -#include -#include -#include +#include +#include +#include #include -#include +#include #include "s3c24xx-pcm.h" #include "s3c24xx-ac97.h" diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ff614d645e79..ba4476b55fbc 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -27,12 +27,12 @@ #include #include -#include -#include -#include -#include +#include +#include +#include +#include #include -#include +#include #include diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index fadd33e2a733..e13e614bada9 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -27,9 +27,9 @@ #include #include -#include -#include -#include +#include +#include +#include #include "s3c24xx-pcm.h" -- cgit v1.2.2 From 23ba79bd79b94fb0205c15b35bac279237979861 Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Sat, 9 Aug 2008 15:05:28 +0400 Subject: ALSA: wm8750: it's MONO1, not MONO Since first commit wm8750 contained output named MONO, but all routes mentioned MONO1. Correct MONO to be MONO1. Signed-off-by: Dmitry Baryshkov Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8750.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index e23cb09f0d14..2e71394ee970 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -348,7 +348,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("ROUT1"), SND_SOC_DAPM_OUTPUT("LOUT2"), SND_SOC_DAPM_OUTPUT("ROUT2"), - SND_SOC_DAPM_OUTPUT("MONO"), + SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_OUTPUT("OUT3"), SND_SOC_DAPM_INPUT("LINPUT1"), -- cgit v1.2.2 From ea381b7b11f189104af34004c5d832ebe49882cc Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Tue, 12 Aug 2008 02:45:30 +0400 Subject: ALSA: spitz: MONO -> MONO1 Correct route name to be MONO1 instead of MONO to follow recent fix in wm8750. Signed-off-by: Dmitry Baryshkov Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/spitz.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index eefc25b83514..37cb768fc933 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -297,7 +297,7 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec) snd_soc_dapm_disable_pin(codec, "LINPUT3"); snd_soc_dapm_disable_pin(codec, "RINPUT3"); snd_soc_dapm_disable_pin(codec, "OUT3"); - snd_soc_dapm_disable_pin(codec, "MONO"); + snd_soc_dapm_disable_pin(codec, "MONO1"); /* Add spitz specific controls */ for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) { -- cgit v1.2.2 From 04489eeb02a40bc15029886cef7285ada3ab0de6 Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Tue, 12 Aug 2008 02:45:31 +0400 Subject: ALSA: wm8750: add missing VREF output Add missing output VREF. After a65f0568f6cc8433877fb71dd7d36b551854b0bc it's critical, since it makes chip routing initialisation to fail. Signed-off-by: Dmitry Baryshkov Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8750.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 2e71394ee970..c6a8edf302ad 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -350,6 +350,7 @@ static const struct snd_soc_dapm_widget wm8750_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("ROUT2"), SND_SOC_DAPM_OUTPUT("MONO1"), SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("VREF"), SND_SOC_DAPM_INPUT("LINPUT1"), SND_SOC_DAPM_INPUT("LINPUT2"), -- cgit v1.2.2 From b29c2360f11060a8e3fe09b16b550494d979371b Mon Sep 17 00:00:00 2001 From: Seth Heasley Date: Fri, 8 Aug 2008 15:56:39 -0700 Subject: ALSA: hda_intel: ALSA HD Audio patch for Intel Ibex Peak DeviceIDs This patch adds the Intel Ibex Peak (PCH) HD Audio Controller DeviceIDs. Signed-off by: Seth Heasley Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index ef9f072b47fc..986dc8e4f02a 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -101,6 +101,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," "{Intel, ICH8}," "{Intel, ICH9}," "{Intel, ICH10}," + "{Intel, PCH}," "{Intel, SCH}," "{ATI, SB450}," "{ATI, SB600}," @@ -2263,6 +2264,8 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x8086, 0x293f), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x3a3e), .driver_data = AZX_DRIVER_ICH }, { PCI_DEVICE(0x8086, 0x3a6e), .driver_data = AZX_DRIVER_ICH }, + /* PCH */ + { PCI_DEVICE(0x8086, 0x3b56), .driver_data = AZX_DRIVER_ICH }, /* SCH */ { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, /* ATI SB 450/600 */ -- cgit v1.2.2 From 9e6dd47bf365f8f7bccea10f22fbbdbecce429e8 Mon Sep 17 00:00:00 2001 From: Libin Yang Date: Tue, 12 Aug 2008 12:25:46 +0200 Subject: ALSA: hda - support new AMD HDMI Audio (1002:970f) Signed-off-by: Libin Yang 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 986dc8e4f02a..a73d6ca0a906 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2275,6 +2275,7 @@ static struct pci_device_id azx_ids[] = { { PCI_DEVICE(0x1002, 0x793b), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0x7919), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0x960f), .driver_data = AZX_DRIVER_ATIHDMI }, + { PCI_DEVICE(0x1002, 0x970f), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0xaa00), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0xaa08), .driver_data = AZX_DRIVER_ATIHDMI }, { PCI_DEVICE(0x1002, 0xaa10), .driver_data = AZX_DRIVER_ATIHDMI }, -- cgit v1.2.2 From 5430c72b14a06b12e8fe46bca18ca0d7095fb717 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Mon, 28 Jul 2008 10:28:43 +0200 Subject: ALSA: virtuoso: add Xonar D1 support Add support for the Asus Xonar D1. It is the same as the DX, but without the external power detection. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/Kconfig | 2 +- sound/pci/oxygen/virtuoso.c | 73 ++++++++++++++++++++++++++++++++++----------- 2 files changed, 56 insertions(+), 19 deletions(-) (limited to 'sound') diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index f7d95b224a98..31f52d3fc21f 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -845,7 +845,7 @@ config SND_VIRTUOSO select SND_OXYGEN_LIB help Say Y here to include support for sound cards based on the - Asus AV100/AV200 chips, i.e., Xonar D2, DX and D2X. + Asus AV100/AV200 chips, i.e., Xonar D1, DX, D2 and D2X. To compile this driver as a module, choose M here: the module will be called snd-virtuoso. diff --git a/sound/pci/oxygen/virtuoso.c b/sound/pci/oxygen/virtuoso.c index 9a2c16bf94e0..01d7b75f9182 100644 --- a/sound/pci/oxygen/virtuoso.c +++ b/sound/pci/oxygen/virtuoso.c @@ -36,15 +36,15 @@ */ /* - * Xonar DX - * -------- + * Xonar D1/DX + * ----------- * * CMI8788: * * I²C <-> CS4398 (front) * <-> CS4362A (surround, center/LFE, back) * - * GPI 0 <- external power present + * GPI 0 <- external power present (DX only) * * GPIO 0 -> enable output to speakers * GPIO 1 -> enable front panel I/O @@ -96,6 +96,7 @@ MODULE_PARM_DESC(enable, "enable card"); enum { MODEL_D2, MODEL_D2X, + MODEL_D1, MODEL_DX, }; @@ -103,6 +104,7 @@ static struct pci_device_id xonar_ids[] __devinitdata = { { OXYGEN_PCI_SUBID(0x1043, 0x8269), .driver_data = MODEL_D2 }, { OXYGEN_PCI_SUBID(0x1043, 0x8275), .driver_data = MODEL_DX }, { OXYGEN_PCI_SUBID(0x1043, 0x82b7), .driver_data = MODEL_D2X }, + { OXYGEN_PCI_SUBID(0x1043, 0x834f), .driver_data = MODEL_D1 }, { } }; MODULE_DEVICE_TABLE(pci, xonar_ids); @@ -313,15 +315,12 @@ static void cs43xx_init(struct oxygen *chip) cs4362a_write(chip, 0x01, CS4362A_CPEN); } -static void xonar_dx_init(struct oxygen *chip) +static void xonar_d1_init(struct oxygen *chip) { struct xonar_data *data = chip->model_data; data->anti_pop_delay = 800; data->output_enable_bit = GPIO_DX_OUTPUT_ENABLE; - data->ext_power_reg = OXYGEN_GPI_DATA; - data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; - data->ext_power_bit = GPI_DX_EXT_POWER; data->cs4398_fm = CS4398_FM_SINGLE | CS4398_DEM_NONE | CS4398_DIF_LJUST; data->cs4362a_fm = CS4362A_FM_SINGLE | CS4362A_ATAPI_B_R | CS4362A_ATAPI_A_L; @@ -345,6 +344,16 @@ static void xonar_dx_init(struct oxygen *chip) snd_component_add(chip->card, "CS5361"); } +static void xonar_dx_init(struct oxygen *chip) +{ + struct xonar_data *data = chip->model_data; + + data->ext_power_reg = OXYGEN_GPI_DATA; + data->ext_power_int_reg = OXYGEN_GPI_INTERRUPT_MASK; + data->ext_power_bit = GPI_DX_EXT_POWER; + xonar_d1_init(chip); +} + static void xonar_cleanup(struct oxygen *chip) { struct xonar_data *data = chip->model_data; @@ -352,7 +361,7 @@ static void xonar_cleanup(struct oxygen *chip) oxygen_clear_bits16(chip, OXYGEN_GPIO_DATA, data->output_enable_bit); } -static void xonar_dx_cleanup(struct oxygen *chip) +static void xonar_d1_cleanup(struct oxygen *chip) { xonar_cleanup(chip); cs4362a_write(chip, 0x01, CS4362A_PDN | CS4362A_CPEN); @@ -365,7 +374,7 @@ static void xonar_d2_resume(struct oxygen *chip) xonar_enable_output(chip); } -static void xonar_dx_resume(struct oxygen *chip) +static void xonar_d1_resume(struct oxygen *chip) { cs43xx_init(chip); xonar_enable_output(chip); @@ -513,7 +522,7 @@ static const struct snd_kcontrol_new front_panel_switch = { .put = front_panel_put, }; -static void xonar_dx_ac97_switch(struct oxygen *chip, +static void xonar_d1_ac97_switch(struct oxygen *chip, unsigned int reg, unsigned int mute) { if (reg == AC97_LINE) { @@ -536,7 +545,7 @@ static int xonar_d2_control_filter(struct snd_kcontrol_new *template) return 0; } -static int xonar_dx_control_filter(struct snd_kcontrol_new *template) +static int xonar_d1_control_filter(struct snd_kcontrol_new *template) { if (!strncmp(template->name, "CD Capture ", 11)) return 1; /* no CD input */ @@ -548,7 +557,7 @@ static int xonar_mixer_init(struct oxygen *chip) return snd_ctl_add(chip->card, snd_ctl_new1(&alt_switch, chip)); } -static int xonar_dx_mixer_init(struct oxygen *chip) +static int xonar_d1_mixer_init(struct oxygen *chip) { return snd_ctl_add(chip->card, snd_ctl_new1(&front_panel_switch, chip)); } @@ -615,23 +624,51 @@ static const struct oxygen_model xonar_models[] = { .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, }, + [MODEL_D1] = { + .shortname = "Xonar D1", + .longname = "Asus Virtuoso 100", + .chip = "AV200", + .owner = THIS_MODULE, + .init = xonar_d1_init, + .control_filter = xonar_d1_control_filter, + .mixer_init = xonar_d1_mixer_init, + .cleanup = xonar_d1_cleanup, + .suspend = xonar_d1_cleanup, + .resume = xonar_d1_resume, + .set_dac_params = set_cs43xx_params, + .set_adc_params = set_cs53x1_params, + .update_dac_volume = update_cs43xx_volume, + .update_dac_mute = update_cs43xx_mute, + .ac97_switch = xonar_d1_ac97_switch, + .dac_tlv = cs4362a_db_scale, + .model_data_size = sizeof(struct xonar_data), + .pcm_dev_cfg = PLAYBACK_0_TO_I2S | + PLAYBACK_1_TO_SPDIF | + CAPTURE_0_FROM_I2S_2, + .dac_channels = 8, + .dac_volume_min = 0, + .dac_volume_max = 127, + .function_flags = OXYGEN_FUNCTION_2WIRE, + .dac_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + .adc_i2s_format = OXYGEN_I2S_FORMAT_LJUST, + }, [MODEL_DX] = { .shortname = "Xonar DX", .longname = "Asus Virtuoso 100", .chip = "AV200", .owner = THIS_MODULE, .init = xonar_dx_init, - .control_filter = xonar_dx_control_filter, - .mixer_init = xonar_dx_mixer_init, - .cleanup = xonar_dx_cleanup, - .suspend = xonar_dx_cleanup, - .resume = xonar_dx_resume, + .control_filter = xonar_d1_control_filter, + .mixer_init = xonar_d1_mixer_init, + .cleanup = xonar_d1_cleanup, + .suspend = xonar_d1_cleanup, + .resume = xonar_d1_resume, .set_dac_params = set_cs43xx_params, .set_adc_params = set_cs53x1_params, .update_dac_volume = update_cs43xx_volume, .update_dac_mute = update_cs43xx_mute, .gpio_changed = xonar_gpio_changed, - .ac97_switch = xonar_dx_ac97_switch, + .ac97_switch = xonar_d1_ac97_switch, .dac_tlv = cs4362a_db_scale, .model_data_size = sizeof(struct xonar_data), .pcm_dev_cfg = PLAYBACK_0_TO_I2S | -- cgit v1.2.2 From 436a74593c34275807fadef20344bbaca251b8d1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 15 Aug 2008 16:22:32 +0100 Subject: ALSA: wm8990: Fix routing of left DAC to speaker mixer Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8990.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 3ecce5168e94..9505a18fa606 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -920,7 +920,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"SPKMIX", "SPKMIX Left Mixer PGA Switch", "LOPGA"}, {"SPKMIX", "SPKMIX Right Mixer PGA Switch", "ROPGA"}, {"SPKMIX", "SPKMIX Right DAC Switch", "Right DAC"}, - {"SPKMIX", "SPKMIX Left DAC Switch", "Right DAC"}, + {"SPKMIX", "SPKMIX Left DAC Switch", "Left DAC"}, /* LONMIX */ {"LONMIX", "LONMIX Left Mixer PGA Switch", "LOPGA"}, -- cgit v1.2.2 From 97bb8129e5deb3c0584391a5d2348966732e2233 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 15 Aug 2008 16:22:33 +0100 Subject: ALSA: wm8990: Implement speaker volume PGA The latest revisions of the WM8990 provide a programmable gain amplifier for the speaker - configure the register cache and implement controls for this. Older revisions of the device ignore writes to these controls. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/wm8990.c | 8 ++++++-- sound/soc/codecs/wm8990.h | 14 ++++++++++++-- 2 files changed, 18 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 9505a18fa606..e44153fa38de 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -82,7 +82,7 @@ static const u16 wm8990_reg[] = { 0x0003, /* R35 - ClassD1 */ 0x0000, /* R36 */ 0x0100, /* R37 - ClassD3 */ - 0x0000, /* R38 */ + 0x0079, /* R38 - ClassD4 */ 0x0000, /* R39 - Input Mixer1 */ 0x0000, /* R40 - Input Mixer2 */ 0x0000, /* R41 - Input Mixer3 */ @@ -311,11 +311,15 @@ SOC_SINGLE("Speaker Mode Switch", WM8990_CLASSD1, WM8990_CDMODE_BIT, 1, 0), SOC_SINGLE("Speaker Output Attenuation Volume", WM8990_SPEAKER_VOLUME, - WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0), + WM8990_SPKATTN_SHIFT, WM8990_SPKATTN_MASK, 0), SOC_SINGLE("Speaker DC Boost Volume", WM8990_CLASSD3, WM8990_DCGAIN_SHIFT, WM8990_DCGAIN_MASK, 0), SOC_SINGLE("Speaker AC Boost Volume", WM8990_CLASSD3, WM8990_ACGAIN_SHIFT, WM8990_ACGAIN_MASK, 0), +SOC_SINGLE_TLV("Speaker Volume", WM8990_CLASSD4, + WM8990_SPKVOL_SHIFT, WM8990_SPKVOL_MASK, 0, out_pga_tlv), +SOC_SINGLE("Speaker ZC Switch", WM8990_CLASSD4, + WM8990_SPKZC_SHIFT, WM8990_SPKZC_MASK, 0), SOC_WM899X_OUTPGA_SINGLE_R_TLV("Left DAC Digital Volume", WM8990_LEFT_DAC_DIGITAL_VOLUME, diff --git a/sound/soc/codecs/wm8990.h b/sound/soc/codecs/wm8990.h index 6bea57485283..0a08325d5443 100644 --- a/sound/soc/codecs/wm8990.h +++ b/sound/soc/codecs/wm8990.h @@ -54,6 +54,7 @@ #define WM8990_SPEAKER_VOLUME 0x22 #define WM8990_CLASSD1 0x23 #define WM8990_CLASSD3 0x25 +#define WM8990_CLASSD4 0x26 #define WM8990_INPUT_MIXER1 0x27 #define WM8990_INPUT_MIXER2 0x28 #define WM8990_INPUT_MIXER3 0x29 @@ -528,8 +529,8 @@ /* * R34 (0x22) - Speaker Volume */ -#define WM8990_SPKVOL_MASK 0x0003 /* SPKVOL - [1:0] */ -#define WM8990_SPKVOL_SHIFT 0 +#define WM8990_SPKATTN_MASK 0x0003 /* SPKATTN - [1:0] */ +#define WM8990_SPKATTN_SHIFT 0 /* * R35 (0x23) - ClassD1 @@ -544,6 +545,15 @@ #define WM8990_DCGAIN_SHIFT 3 #define WM8990_ACGAIN_MASK 0x0007 /* ACGAIN - [2:0] */ #define WM8990_ACGAIN_SHIFT 0 + +/* + * R38 (0x26) - ClassD4 + */ +#define WM8990_SPKZC_MASK 0x0001 /* SPKZC */ +#define WM8990_SPKZC_SHIFT 7 /* SPKZC */ +#define WM8990_SPKVOL_MASK 0x007F /* SPKVOL - [6:0] */ +#define WM8990_SPKVOL_SHIFT 0 /* SPKVOL - [6:0] */ + /* * R39 (0x27) - Input Mixer1 */ -- cgit v1.2.2 From f511b01c8e747b80635c8b2acd61431abcab4b29 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 15 Aug 2008 16:46:42 +0200 Subject: ALSA: hda - Fix capture source widgets on ALC codecs On some Realtek codecs like ALC882 or ALC883, the capture source is no mux but sum widget. We have to initialize all channels properly for this type, otherwise noises may come in from the unused route. The patch assures to mute unused routes, and unmute the currently selected route. Signed-off-by: Takashi Iwai Tested-by: Daniel J Blueman --- sound/pci/hda/patch_realtek.c | 45 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index add4e87e0b20..b80e725432f0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6437,6 +6437,39 @@ static void alc882_auto_init_analog_input(struct hda_codec *codec) } } +static void alc882_auto_init_input_src(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + const struct hda_input_mux *imux = spec->input_mux; + int c; + + for (c = 0; c < spec->num_adc_nids; c++) { + hda_nid_t conn_list[HDA_MAX_NUM_INPUTS]; + hda_nid_t nid = spec->capsrc_nids[c]; + int conns, mute, idx, item; + + conns = snd_hda_get_connections(codec, nid, conn_list, + ARRAY_SIZE(conn_list)); + if (conns < 0) + continue; + for (idx = 0; idx < conns; idx++) { + /* if the current connection is the selected one, + * unmute it as default - otherwise mute it + */ + mute = AMP_IN_MUTE(idx); + for (item = 0; item < imux->num_items; item++) { + if (imux->items[item].index == idx) { + if (spec->cur_mux[c] == item) + mute = AMP_IN_UNMUTE(idx); + break; + } + } + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, mute); + } + } +} + /* add mic boosts if needed */ static int alc_auto_add_mic_boost(struct hda_codec *codec) { @@ -6491,6 +6524,7 @@ static void alc882_auto_init(struct hda_codec *codec) alc882_auto_init_multi_out(codec); alc882_auto_init_hp_out(codec); alc882_auto_init_analog_input(codec); + alc882_auto_init_input_src(codec); if (spec->unsol_event) alc_sku_automute(codec); } @@ -8285,6 +8319,8 @@ static void alc883_auto_init_analog_input(struct hda_codec *codec) } } +#define alc883_auto_init_input_src alc882_auto_init_input_src + /* almost identical with ALC880 parser... */ static int alc883_parse_auto_config(struct hda_codec *codec) { @@ -8315,6 +8351,7 @@ static void alc883_auto_init(struct hda_codec *codec) alc883_auto_init_multi_out(codec); alc883_auto_init_hp_out(codec); alc883_auto_init_analog_input(codec); + alc883_auto_init_input_src(codec); if (spec->unsol_event) alc_sku_automute(codec); } @@ -9663,6 +9700,7 @@ static int alc262_parse_auto_config(struct hda_codec *codec) #define alc262_auto_init_multi_out alc882_auto_init_multi_out #define alc262_auto_init_hp_out alc882_auto_init_hp_out #define alc262_auto_init_analog_input alc882_auto_init_analog_input +#define alc262_auto_init_input_src alc882_auto_init_input_src /* init callback for auto-configuration model -- overriding the default init */ @@ -9672,6 +9710,7 @@ static void alc262_auto_init(struct hda_codec *codec) alc262_auto_init_multi_out(codec); alc262_auto_init_hp_out(codec); alc262_auto_init_analog_input(codec); + alc262_auto_init_input_src(codec); if (spec->unsol_event) alc_sku_automute(codec); } @@ -13330,6 +13369,8 @@ static void alc861vd_auto_init_analog_input(struct hda_codec *codec) } } +#define alc861vd_auto_init_input_src alc882_auto_init_input_src + #define alc861vd_idx_to_mixer_vol(nid) ((nid) + 0x02) #define alc861vd_idx_to_mixer_switch(nid) ((nid) + 0x0c) @@ -13512,6 +13553,7 @@ static void alc861vd_auto_init(struct hda_codec *codec) alc861vd_auto_init_multi_out(codec); alc861vd_auto_init_hp_out(codec); alc861vd_auto_init_analog_input(codec); + alc861vd_auto_init_input_src(codec); if (spec->unsol_event) alc_sku_automute(codec); } @@ -14677,6 +14719,8 @@ static void alc662_auto_init_analog_input(struct hda_codec *codec) } } +#define alc662_auto_init_input_src alc882_auto_init_input_src + static int alc662_parse_auto_config(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -14733,6 +14777,7 @@ static void alc662_auto_init(struct hda_codec *codec) alc662_auto_init_multi_out(codec); alc662_auto_init_hp_out(codec); alc662_auto_init_analog_input(codec); + alc662_auto_init_input_src(codec); if (spec->unsol_event) alc_sku_automute(codec); } -- cgit v1.2.2 From 320dcc30f498e0a8b282b14cf0feed1897ea3b34 Mon Sep 17 00:00:00 2001 From: Peer Chen Date: Wed, 20 Aug 2008 16:43:24 -0700 Subject: ALSA: hda_intel: enable snoop for nvidia HDA controller Enable the snoop for nvidia hda controller to avoid data coherence issue. Signed-off-by: Peer Chen Signed-off-by: Andrew Morton Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 9 +++++++++ 1 file changed, 9 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index a73d6ca0a906..1c53e337ecb2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -278,6 +278,9 @@ enum { /* Defines for Nvidia HDA support */ #define NVIDIA_HDA_TRANSREG_ADDR 0x4e #define NVIDIA_HDA_ENABLE_COHBITS 0x0f +#define NVIDIA_HDA_ISTRM_COH 0x4d +#define NVIDIA_HDA_OSTRM_COH 0x4c +#define NVIDIA_HDA_ENABLE_COHBIT 0x01 /* Defines for Intel SCH HDA snoop control */ #define INTEL_SCH_HDA_DEVC 0x78 @@ -900,6 +903,12 @@ static void azx_init_pci(struct azx *chip) update_pci_byte(chip->pci, NVIDIA_HDA_TRANSREG_ADDR, 0x0f, NVIDIA_HDA_ENABLE_COHBITS); + update_pci_byte(chip->pci, + NVIDIA_HDA_ISTRM_COH, + 0x01, NVIDIA_HDA_ENABLE_COHBIT); + update_pci_byte(chip->pci, + NVIDIA_HDA_OSTRM_COH, + 0x01, NVIDIA_HDA_ENABLE_COHBIT); break; case AZX_DRIVER_SCH: pci_read_config_word(chip->pci, INTEL_SCH_HDA_DEVC, &snoop); -- cgit v1.2.2 From 1082c7487cbe5a40755ba9e33552b6ecbf419bf2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 22 Aug 2008 15:24:22 +0200 Subject: ALSA: hda - Fix call of alc888_coef_init() Using init_hook to call alc888_coef_init() is problematic for configurations that already set another init_hook. Better to put it in alc_init() as is (although it looks a bit hackish). Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b80e725432f0..909f1c101c95 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -952,7 +952,7 @@ do_sku: tmp | 0x2010); break; case 0x10ec0888: - alc888_coef_init(codec); + /*alc888_coef_init(codec);*/ /* called in alc_init() */ break; case 0x10ec0267: case 0x10ec0268: @@ -2439,6 +2439,8 @@ static int alc_init(struct hda_codec *codec) unsigned int i; alc_fix_pll(codec); + if (codec->vendor_id == 0x10ec0888) + alc888_coef_init(codec); for (i = 0; i < spec->num_init_verbs; i++) snd_hda_sequence_write(codec, spec->init_verbs[i]); @@ -8426,8 +8428,6 @@ static int patch_alc883(struct hda_codec *codec) codec->patch_ops = alc_patch_ops; if (board_config == ALC883_AUTO) spec->init_hook = alc883_auto_init; - else if (codec->vendor_id == 0x10ec0888) - spec->init_hook = alc888_coef_init; #ifdef CONFIG_SND_HDA_POWER_SAVE if (!spec->loopback.amplist) -- cgit v1.2.2 From 7a8fc9b248e77a4eab0613acf30a6811799786b3 Mon Sep 17 00:00:00 2001 From: Adrian Bunk Date: Sun, 17 Aug 2008 17:36:59 +0300 Subject: removed unused #include 's This patch lets the files using linux/version.h match the files that #include it. Signed-off-by: Adrian Bunk Signed-off-by: Linus Torvalds --- sound/mips/au1x00.c | 1 - sound/soc/at91/eti_b1_wm8731.c | 1 - sound/soc/codecs/wm8753.c | 1 - sound/soc/codecs/wm9712.c | 1 - 4 files changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index ee0741f9eb53..fbef38a9604a 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -38,7 +38,6 @@ #include #include #include -#include #include #include #include diff --git a/sound/soc/at91/eti_b1_wm8731.c b/sound/soc/at91/eti_b1_wm8731.c index b081e83766b7..b81d6b2cfa1d 100644 --- a/sound/soc/at91/eti_b1_wm8731.c +++ b/sound/soc/at91/eti_b1_wm8731.c @@ -22,7 +22,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8604809f0c36..dc7b18fd2782 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -34,7 +34,6 @@ #include #include -#include #include #include #include diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 1fb7f9a7aecd..2f1c91b1d556 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -13,7 +13,6 @@ #include #include -#include #include #include #include -- cgit v1.2.2 From c5d44423d55e3abca7b1d544af9e4c97ec203999 Mon Sep 17 00:00:00 2001 From: Travis Place Date: Mon, 25 Aug 2008 08:11:50 +0200 Subject: ALSA: CA0106 on MSI K8N Diamond PLUS Motherboard Correct a previous patch for the ca0106 onboard the MSI K8N Diamond PLUS motherboard. Confirmed to have Line/Mic/Aux working for input, and sound output working as expected. Signed-off-by: Travis Place Signed-off-by: Takashi Iwai --- sound/pci/ca0106/ca0106_main.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 2f8b28add276..03a274becae0 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -249,11 +249,12 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB [SB0438]", .gpio_type = 2, .i2c_adc = 1 } , - /* Another MSI K8N Diamond MB, which has apprently a different SSID */ + /* MSI K8N Diamond PLUS MB */ { .serial = 0x10091102, .name = "MSI K8N Diamond MB", .gpio_type = 2, - .i2c_adc = 1 } , + .i2c_adc = 1, + .spi_dac = 2 } /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". -- cgit v1.2.2 From 3051e41ab7daaa59d4564f20b25dcb8c03f35f2b Mon Sep 17 00:00:00 2001 From: Jean Delvare Date: Mon, 25 Aug 2008 11:49:20 +0100 Subject: ALSA: ASoC: Fix double free and memory leak in many codec drivers Many SoC audio codec drivers have improper freeing of memory in error paths. * codec is allocated in the platform device probe function, but is not freed there in case of error. Instead it is freed in the i2c device probe function's error path. However the success or failure of both functions is not linked, so this could result in a double free (if the platform device is successfully probed, the i2c device probing fails and then the platform driver is unregistered.) * codec->private_data is allocated in many platform device probe functions but not freed in their error paths. This patch hopefully solves all these problems. Signed-off-by: Jean Delvare Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/codecs/ak4535.c | 11 +++++++---- sound/soc/codecs/tlv320aic3x.c | 11 +++++++---- sound/soc/codecs/uda1380.c | 9 +++++---- sound/soc/codecs/wm8510.c | 9 +++++---- sound/soc/codecs/wm8731.c | 11 +++++++---- sound/soc/codecs/wm8750.c | 10 ++++++---- sound/soc/codecs/wm8753.c | 11 +++++++---- sound/soc/codecs/wm8990.c | 11 +++++++---- 8 files changed, 51 insertions(+), 32 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index b26003c4f3e8..7da9f467b7b8 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -562,10 +562,9 @@ static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -583,7 +582,6 @@ static int ak4535_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -660,6 +658,11 @@ static int ak4535_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index b1dce5f459db..5f9abb199435 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1199,10 +1199,9 @@ static int aic3x_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -1221,7 +1220,6 @@ static int aic3x_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -1302,6 +1300,11 @@ static int aic3x_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index a52d6d9e007a..807318fbdc8f 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -729,10 +729,9 @@ static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -750,7 +749,6 @@ static int uda1380_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -817,6 +815,9 @@ static int uda1380_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) + kfree(codec); return ret; } diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 67325fd95447..3d998e6a997e 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -693,10 +693,9 @@ static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -714,7 +713,6 @@ static int wm8510_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -782,6 +780,9 @@ static int wm8510_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) + kfree(codec); return ret; } diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 369d39c3f745..9402fcaf04fa 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -596,10 +596,9 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -617,7 +616,6 @@ static int wm8731_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -693,6 +691,11 @@ static int wm8731_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index c6a8edf302ad..dd1f55404b29 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -869,10 +869,9 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -890,7 +889,6 @@ static int wm8750_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -966,6 +964,10 @@ static int wm8750_probe(struct platform_device *pdev) /* Add other interfaces here */ #endif + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 8604809f0c36..35bf1c369870 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1661,10 +1661,9 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (!i2c) { - kfree(codec); + if (!i2c) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -1683,7 +1682,6 @@ static int wm8753_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -1760,6 +1758,11 @@ static int wm8753_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index e44153fa38de..dd995ef448b4 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1500,10 +1500,9 @@ static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind) client_template.addr = addr; i2c = kmemdup(&client_template, sizeof(client_template), GFP_KERNEL); - if (i2c == NULL) { - kfree(codec); + if (i2c == NULL) return -ENOMEM; - } + i2c_set_clientdata(i2c, codec); codec->control_data = i2c; @@ -1521,7 +1520,6 @@ static int wm8990_codec_probe(struct i2c_adapter *adap, int addr, int kind) return ret; err: - kfree(codec); kfree(i2c); return ret; } @@ -1595,6 +1593,11 @@ static int wm8990_probe(struct platform_device *pdev) #else /* Add other interfaces here */ #endif + + if (ret != 0) { + kfree(codec->private_data); + kfree(codec); + } return ret; } -- cgit v1.2.2 From e5778ec91e823b97262f045814d34d0abde689c0 Mon Sep 17 00:00:00 2001 From: Stephen Rothwell Date: Tue, 26 Aug 2008 10:33:32 +1000 Subject: ALSA: fix for CA0106 on MSI K8N Diamond PLUS Motherboard Signed-off-by: Stephen Rothwell Signed-off-by: Linus Torvalds --- sound/pci/ca0106/ca0106_main.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 03a274becae0..6abe8a3bd365 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -254,7 +254,7 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB", .gpio_type = 2, .i2c_adc = 1, - .spi_dac = 2 } + .spi_dac = 2 }, /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". -- cgit v1.2.2 From 3d839e5b87a70effc629c1cdbf77d837ef141919 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Tue, 26 Aug 2008 11:06:26 +0200 Subject: ALSA: oxygen: prevent muting of nonexistent AC97 controls The Xonar DX does not have CD Capture controls, so we have to check that a control actually exists before muting it. Signed-off-by: Clemens Ladisch Cc: Signed-off-by: Takashi Iwai --- sound/pci/oxygen/oxygen_mixer.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/oxygen/oxygen_mixer.c b/sound/pci/oxygen/oxygen_mixer.c index 6facac5aed90..05eb8994c141 100644 --- a/sound/pci/oxygen/oxygen_mixer.c +++ b/sound/pci/oxygen/oxygen_mixer.c @@ -512,9 +512,12 @@ static int ac97_switch_get(struct snd_kcontrol *ctl, static void mute_ac97_ctl(struct oxygen *chip, unsigned int control) { - unsigned int priv_idx = chip->controls[control]->private_value & 0xff; + unsigned int priv_idx; u16 value; + if (!chip->controls[control]) + return; + priv_idx = chip->controls[control]->private_value & 0xff; value = oxygen_read_ac97(chip, 0, priv_idx); if (!(value & 0x8000)) { oxygen_write_ac97(chip, 0, priv_idx, value | 0x8000); -- cgit v1.2.2 From e784539fe81490a982a013621d39a60c4fce427e Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Tue, 26 Aug 2008 13:32:57 +0300 Subject: ALSA: ASoC: Fix error paths in N810 machine driver init and release clocks at exit Thanks to Felipe Balbi by noticing that if clk_get to sys_clkout2_src fails, then n810_snd_device is never released. Add also sys_clkout2_src release into error path, error code return and release the clocks at exit. Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/omap/n810.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 7694621ec40b..87d0ed01f65a 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -329,12 +329,14 @@ static int __init n810_soc_init(void) sys_clkout2_src = clk_get(dev, "sys_clkout2_src"); if (IS_ERR(sys_clkout2_src)) { dev_err(dev, "Could not get sys_clkout2_src clock\n"); - return -ENODEV; + err = PTR_ERR(sys_clkout2_src); + goto err2; } sys_clkout2 = clk_get(dev, "sys_clkout2"); if (IS_ERR(sys_clkout2)) { dev_err(dev, "Could not get sys_clkout2\n"); - goto err1; + err = PTR_ERR(sys_clkout2); + goto err3; } /* * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use @@ -343,7 +345,8 @@ static int __init n810_soc_init(void) func96m_clk = clk_get(dev, "func_96m_ck"); if (IS_ERR(func96m_clk)) { dev_err(dev, "Could not get func 96M clock\n"); - goto err2; + err = PTR_ERR(func96m_clk); + goto err4; } clk_set_parent(sys_clkout2_src, func96m_clk); clk_set_rate(sys_clkout2, 12000000); @@ -356,20 +359,25 @@ static int __init n810_soc_init(void) gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0); return 0; -err2: +err4: clk_put(sys_clkout2); +err3: + clk_put(sys_clkout2_src); +err2: platform_device_del(n810_snd_device); err1: platform_device_put(n810_snd_device); return err; - } static void __exit n810_soc_exit(void) { gpio_free(N810_SPEAKER_AMP_GPIO); gpio_free(N810_HEADSET_AMP_GPIO); + clk_put(sys_clkout2_src); + clk_put(sys_clkout2); + clk_put(func96m_clk); platform_device_unregister(n810_snd_device); } -- cgit v1.2.2 From 93a1a5eb70be5cc14990b97ef2460212e32658dc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 27 Aug 2008 07:56:43 +0200 Subject: Revert "ALSA: hda - Added model selection for iMac 24"" This reverts commit 3e0e469fa216ec70c93b1593821b759d19ee2e6b. The patch introduced a wrong detection of other intel Macs with ALC88* codec because they share the same PCI SSID (but have different codec subsystem-IDs). See http://lkml.org/lkml/2008/8/24/143 Reported-and-tested-by: Guillaume Chazarain Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 909f1c101c95..d6ec9eef2910 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6197,7 +6197,6 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x817f, "Asus P5LD2", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1043, 0x81d8, "Asus P5WD", ALC882_6ST_DIG), SND_PCI_QUIRK(0x105b, 0x6668, "Foxconn", ALC882_6ST_DIG), - SND_PCI_QUIRK(0x106b, 0x00a0, "Apple iMac 24''", ALC885_IMAC24), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x28fb, "Targa T8", ALC882_TARGA), /* MSI-1049 T8 */ SND_PCI_QUIRK(0x1462, 0x6668, "MSI", ALC882_6ST_DIG), -- cgit v1.2.2 From df91bc23dcb052ff2da71b3482bf3c5fbf4b8a53 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 29 Aug 2008 13:08:34 +0200 Subject: ALSA: oxygen: fix distorted output on AK4396-based cards When changing the sample rate, the CMI8788's master clock output becomes unstable for a short time. The AK4396 needs the master clock to do SPI writes, so writing to an AK4396 control register directly after a sample rate change will garble the value. In our case, this leads to the DACs being misconfigured to I2S sample format, which results in a wrong output level and horrible distortions on samples louder than -6 dB. To fix this, we need to wait until the new master clock signal has become stable before doing SPI writes. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/pci/oxygen/hifier.c | 4 ++++ sound/pci/oxygen/oxygen.c | 4 ++++ 2 files changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/oxygen/hifier.c b/sound/pci/oxygen/hifier.c index 7442460583dd..dad393ae040a 100644 --- a/sound/pci/oxygen/hifier.c +++ b/sound/pci/oxygen/hifier.c @@ -17,6 +17,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ +#include #include #include #include @@ -107,6 +108,9 @@ static void set_ak4396_params(struct oxygen *chip, else value |= AK4396_DFS_QUAD; data->ak4396_ctl2 = value; + + msleep(1); /* wait for the new MCLK to become stable */ + ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB); ak4396_write(chip, AK4396_CONTROL_2, value); ak4396_write(chip, AK4396_CONTROL_1, AK4396_DIF_24_MSB | AK4396_RSTN); diff --git a/sound/pci/oxygen/oxygen.c b/sound/pci/oxygen/oxygen.c index 7c8ae31eb468..c5829d30ef86 100644 --- a/sound/pci/oxygen/oxygen.c +++ b/sound/pci/oxygen/oxygen.c @@ -28,6 +28,7 @@ * GPIO 1 -> DFS1 of AK5385 */ +#include #include #include #include @@ -213,6 +214,9 @@ static void set_ak4396_params(struct oxygen *chip, else value |= AK4396_DFS_QUAD; data->ak4396_ctl2 = value; + + msleep(1); /* wait for the new MCLK to become stable */ + for (i = 0; i < 4; ++i) { ak4396_write(chip, i, AK4396_CONTROL_1, AK4396_DIF_24_MSB); -- cgit v1.2.2 From 20f5f95ded9cdab62c26efb146967a75e12533ec Mon Sep 17 00:00:00 2001 From: Matthew Ranostay Date: Mon, 1 Sep 2008 08:17:56 +0200 Subject: ALSA: hda: Distortion fix for dell_m6_core_init Added the EQ distortion fix to the dell_m6_core_init. Signed-off-by: Matthew Ranostay Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 7fdafcb0015d..ad994fcab725 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -560,8 +560,9 @@ static struct hda_verb dell_eq_core_init[] = { }; static struct hda_verb dell_m6_core_init[] = { - /* set master volume and direct control */ - { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* set master volume to max value without distortion + * and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xec}, /* setup audio connections */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x00}, { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, -- cgit v1.2.2 From 6e5ea7015c62b672020ee0a7c2764942fe63fa25 Mon Sep 17 00:00:00 2001 From: Dmitry Baryshkov Date: Sun, 31 Aug 2008 00:45:02 +0400 Subject: ALSA: ASoC: fix pxa2xx-i2s clk_get call pxa2xx-i2s: probe actual device and use it for clk_get call thus fixing error during startup hook Signed-off-by: Dmitry Baryshkov Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/pxa/pxa2xx-i2s.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 39 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa2xx-i2s.c b/sound/soc/pxa/pxa2xx-i2s.c index 8548818eea08..c796b1882776 100644 --- a/sound/soc/pxa/pxa2xx-i2s.c +++ b/sound/soc/pxa/pxa2xx-i2s.c @@ -16,6 +16,7 @@ #include #include #include +#include #include #include #include @@ -81,7 +82,6 @@ static int pxa2xx_i2s_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; - clk_i2s = clk_get(NULL, "I2SCLK"); if (IS_ERR(clk_i2s)) return PTR_ERR(clk_i2s); @@ -152,6 +152,7 @@ static int pxa2xx_i2s_hw_params(struct snd_pcm_substream *substream, pxa_gpio_mode(gpio_bus[pxa_i2s.master].tx); pxa_gpio_mode(gpio_bus[pxa_i2s.master].frm); pxa_gpio_mode(gpio_bus[pxa_i2s.master].clk); + BUG_ON(IS_ERR(clk_i2s)); clk_enable(clk_i2s); pxa_i2s_wait(); @@ -317,6 +318,43 @@ struct snd_soc_dai pxa_i2s_dai = { EXPORT_SYMBOL_GPL(pxa_i2s_dai); +static int pxa2xx_i2s_probe(struct platform_device *dev) +{ + clk_i2s = clk_get(&dev->dev, "I2SCLK"); + return IS_ERR(clk_i2s) ? PTR_ERR(clk_i2s) : 0; +} + +static int __devexit pxa2xx_i2s_remove(struct platform_device *dev) +{ + clk_put(clk_i2s); + clk_i2s = ERR_PTR(-ENOENT); + return 0; +} + +static struct platform_driver pxa2xx_i2s_driver = { + .probe = pxa2xx_i2s_probe, + .remove = __devexit_p(pxa2xx_i2s_remove), + + .driver = { + .name = "pxa2xx-i2s", + .owner = THIS_MODULE, + }, +}; + +static int __init pxa2xx_i2s_init(void) +{ + clk_i2s = ERR_PTR(-ENOENT); + return platform_driver_register(&pxa2xx_i2s_driver); +} + +static void __exit pxa2xx_i2s_exit(void) +{ + platform_driver_unregister(&pxa2xx_i2s_driver); +} + +module_init(pxa2xx_i2s_init); +module_exit(pxa2xx_i2s_exit); + /* Module information */ MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com"); MODULE_DESCRIPTION("pxa2xx I2S SoC Interface"); -- cgit v1.2.2 From 24fb9173815045ab3f85a670d7df8af5af6ff3be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Sep 2008 14:48:20 +0200 Subject: ALSA: hda - Fix ALC663 auto-probe Fix the wrong DAC assignment for NID 0x17 mono-pin on ALC663. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d6ec9eef2910..635748b122e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14066,6 +14066,13 @@ static struct hda_verb alc662_auto_init_verbs[] = { { } }; +/* additional verbs for ALC663 */ +static struct hda_verb alc663_auto_init_verbs[] = { + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + { } +}; + static struct hda_verb alc663_m51va_init_verbs[] = { {0x21, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, @@ -14594,6 +14601,14 @@ static int alc662_auto_create_extra_out(struct alc_spec *spec, hda_nid_t pin, if (!pin) return 0; + if (pin == 0x17) { + /* ALC663 has a mono output pin on 0x17 */ + sprintf(name, "%s Playback Switch", pfx); + err = add_control(spec, ALC_CTL_WIDGET_MUTE, name, + HDA_COMPOSE_AMP_VAL(pin, 2, 0, HDA_OUTPUT)); + return err; + } + if (alc880_is_fixed_pin(pin)) { nid = alc880_idx_to_dac(alc880_fixed_pin_idx(pin)); /* printk("DAC nid=%x\n",nid); */ @@ -14764,6 +14779,9 @@ static int alc662_parse_auto_config(struct hda_codec *codec) spec->input_mux = &spec->private_imux; spec->init_verbs[spec->num_init_verbs++] = alc662_auto_init_verbs; + if (codec->vendor_id == 0x10ec0663) + spec->init_verbs[spec->num_init_verbs++] = + alc663_auto_init_verbs; spec->mixers[spec->num_mixers] = alc662_capture_mixer; spec->num_mixers++; return 1; -- cgit v1.2.2 From ee979a143cfd999adea8a9e272649a3cd9ec84bc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 2 Sep 2008 15:42:20 +0200 Subject: ALSA: hda - Add mic-boost controls to ALC662/663 auto configuration Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 5 +++++ 1 file changed, 5 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 635748b122e9..66025161bd69 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -14782,6 +14782,11 @@ static int alc662_parse_auto_config(struct hda_codec *codec) if (codec->vendor_id == 0x10ec0663) spec->init_verbs[spec->num_init_verbs++] = alc663_auto_init_verbs; + + err = alc_auto_add_mic_boost(codec); + if (err < 0) + return err; + spec->mixers[spec->num_mixers] = alc662_capture_mixer; spec->num_mixers++; return 1; -- cgit v1.2.2 From 8a656496b21efd95fd55b66e0601c5ad41f9b156 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 6 Sep 2008 11:43:41 +0200 Subject: Fix CONFIG_AC97_BUS dependency CONFIG_AC97_BUS is used from both sound and ucb1400 drivers. The recent change in Kconfig introduced the exclusive dependency on CONFIG_SOUND, and disabled the ucb1400 build without sound. This patch makes CONFIG_AC97_BUS independent. Signed-off-by: Takashi Iwai Tested-by: Randy Dunlap --- sound/Kconfig | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/Kconfig b/sound/Kconfig index a37bee094eba..8ebf512ced6c 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -91,6 +91,9 @@ endif # SOUND_PRIME endif # !M68K +endif # SOUND + +# AC97_BUS is used from both sound and ucb1400 config AC97_BUS tristate help @@ -99,4 +102,3 @@ config AC97_BUS sound although they're sharing the AC97 bus. Concerned drivers should "select" this. -endif # SOUND -- cgit v1.2.2