From 2aba76f014a7b56ab4fe75845c5fd57b5590acc2 Mon Sep 17 00:00:00 2001 From: Michael Williamson Date: Fri, 20 May 2011 10:26:06 -0400 Subject: audio: tlv320aic26: fix PLL register configuration The current PLL configuration code for the tlc320aic26 codec appears to assume a hardcoded system clock of 12 MHz. Use the clock value provided by the DAI_OPS API for the calculation. Tested using a MityDSP-L138 platform providing a 24.576 MHz clock. Signed-off-by: Michael Williamson Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320aic26.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index e2a7608d3944..7859bdcc93db 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -161,10 +161,18 @@ static int aic26_hw_params(struct snd_pcm_substream *substream, dev_dbg(&aic26->spi->dev, "bad format\n"); return -EINVAL; } - /* Configure PLL */ + /** + * Configure PLL + * fsref = (mclk * PLLM) / 2048 + * where PLLM = J.DDDD (DDDD register ranges from 0 to 9999, decimal) + */ pval = 1; - jval = (fsref == 44100) ? 7 : 8; - dval = (fsref == 44100) ? 5264 : 1920; + /* compute J portion of multiplier */ + jval = fsref / (aic26->mclk / 2048); + /* compute fractional DDDD component of multiplier */ + dval = fsref - (jval * (aic26->mclk / 2048)); + dval = (10000 * dval) / (aic26->mclk / 2048); + dev_dbg(&aic26->spi->dev, "Setting PLLM to %d.%04d\n", jval, dval); qval = 0; reg = 0x8000 | qval << 11 | pval << 8 | jval << 2; aic26_reg_write(codec, AIC26_REG_PLL_PROG1, reg); -- cgit v1.2.3-58-ga151 From 508b76864c18f34f8d6ba08d192f5817f8dc8ead Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 20 May 2011 16:52:37 +0300 Subject: ASoC: tlv320aic3x: Don't sync first two registers from register cache There is no need to sync first two registers from cache to hw after a reset. First one is used to select page for register access and this driver is normally accessing page 0 only. Second one does a software reset which is obviously unneeded after hardware or previous software reset command. Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320aic3x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index c3d96fc8c267..9047bb173c6b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1114,7 +1114,7 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) /* Sync reg_cache with the hardware */ codec->cache_only = 0; - for (i = 0; i < ARRAY_SIZE(aic3x_reg); i++) + for (i = AIC3X_SAMPLE_RATE_SEL_REG; i < ARRAY_SIZE(aic3x_reg); i++) snd_soc_write(codec, i, cache[i]); if (aic3x->model == AIC3X_MODEL_3007) aic3x_init_3007(codec); -- cgit v1.2.3-58-ga151 From 9fb352b18b11124ed1ddebc0d74ebbd7ba8defd7 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Fri, 20 May 2011 16:52:38 +0300 Subject: ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state TLV320AIC33, TLV320AIC34 and I believe others too in this family have some hw bugs that cause that analogue and digital VDD supplies remain leaking up to a few mA of current after certain use cases even the hw blocks inside codec are driven to off. Highest leakages occur after using the bypass paths inside codec but it is possible to get smaller leakages just by toggling mute switches in unused audio paths (i.e. no DAPM changes) while codec is on due another active audio path. While some cases are able to workaroud by making sure that e.g. output mixer switches are muted before powering down the output stage this doesn't help all the cases. Therefore use the software reset command to clear possible leakage currents since that works in every cases and affects only this codec instance. Only drawback is that now cache sync is required everytime when codec bias comes out from bias off state, not only when supply regulators were off. Signed-off-by: Jarkko Nikula Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320aic3x.c | 7 +++++++ 1 file changed, 7 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 9047bb173c6b..789453d44ec5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1120,6 +1120,13 @@ static int aic3x_set_power(struct snd_soc_codec *codec, int power) aic3x_init_3007(codec); codec->cache_sync = 0; } else { + /* + * Do soft reset to this codec instance in order to clear + * possible VDD leakage currents in case the supply regulators + * remain on + */ + snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); + codec->cache_sync = 1; aic3x->power = 0; /* HW writes are needless when bias is off */ codec->cache_only = 1; -- cgit v1.2.3-58-ga151 From 840d8e5e964dc51673d0f26e119b27d2898e8417 Mon Sep 17 00:00:00 2001 From: Joachim Eastwood Date: Wed, 1 Jun 2011 23:59:10 +0200 Subject: ASoC: atmel_ssc: Don't try to free ssc if request failed We should only call ssc_free() when ssc_request() succeeds or bad things will happen. Signed-off-by: Joachim Eastwood Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 7fbfa051f6e1..eda955b15834 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -848,9 +848,10 @@ int atmel_ssc_set_audio(int ssc_id) if (IS_ERR(ssc)) pr_warn("Unable to parent ASoC SSC DAI on SSC: %ld\n", PTR_ERR(ssc)); - else + else { ssc_pdev->dev.parent = &(ssc->pdev->dev); - ssc_free(ssc); + ssc_free(ssc); + } ret = platform_device_add(ssc_pdev); if (ret < 0) -- cgit v1.2.3-58-ga151 From 1622ee1822e8adb391b55a09e3cd5144bd9fad47 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2011 17:13:57 +0100 Subject: ASoC: Only update SYSCLK_ENA when pausing WM8915 SYSCLK Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8915.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index a0b1a7278284..28fbf072b9c0 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -1839,7 +1839,7 @@ static int wm8915_set_sysclk(struct snd_soc_dai *dai, int old; /* Disable SYSCLK while we reconfigure */ - old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1); + old = snd_soc_read(codec, WM8915_AIF_CLOCKING_1) & WM8915_SYSCLK_ENA; snd_soc_update_bits(codec, WM8915_AIF_CLOCKING_1, WM8915_SYSCLK_ENA, 0); -- cgit v1.2.3-58-ga151 From 6ac340623c5d2a945030814d900701439772ff57 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 3 Jun 2011 18:20:50 +0100 Subject: ASoC: Add missing break in WM8915 FLL source selection Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/codecs/wm8915.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8915.c b/sound/soc/codecs/wm8915.c index 28fbf072b9c0..e2ab4fac2819 100644 --- a/sound/soc/codecs/wm8915.c +++ b/sound/soc/codecs/wm8915.c @@ -2038,6 +2038,7 @@ static int wm8915_set_fll(struct snd_soc_codec *codec, int fll_id, int source, break; case WM8915_FLL_MCLK2: reg = 1; + break; case WM8915_FLL_DACLRCLK1: reg = 2; break; -- cgit v1.2.3-58-ga151 From fd137e2bba53b7207cbae6a1312e89ef3ae55624 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 6 Jun 2011 11:26:15 +0100 Subject: ASoC: Check for NULL register bank in snd_soc_get_cache_val() Signed-off-by: Mark Brown Acked-by: Liam Girdwood --- sound/soc/soc-cache.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index 06b7b81a1601..c005ceb70c9d 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -466,6 +466,9 @@ static bool snd_soc_set_cache_val(void *base, unsigned int idx, static unsigned int snd_soc_get_cache_val(const void *base, unsigned int idx, unsigned int word_size) { + if (!base) + return -1; + switch (word_size) { case 1: { const u8 *cache = base; -- cgit v1.2.3-58-ga151 From 8ca695f273709a9d147826716a8dee3e0eb2407f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 6 Jun 2011 13:38:35 +0200 Subject: ASoC: AD1836: Fix setting the PCM format Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ad1836.c | 14 +++++++------- sound/soc/codecs/ad1836.h | 6 ++++++ 2 files changed, 13 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index ab63d52e36e1..754c496412bd 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -145,22 +145,22 @@ static int ad1836_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - word_len = 3; + word_len = AD1836_WORD_LEN_16; break; case SNDRV_PCM_FORMAT_S20_3LE: - word_len = 1; + word_len = AD1836_WORD_LEN_20; break; case SNDRV_PCM_FORMAT_S24_LE: case SNDRV_PCM_FORMAT_S32_LE: - word_len = 0; + word_len = AD1836_WORD_LEN_24; break; } - snd_soc_update_bits(codec, AD1836_DAC_CTRL1, - AD1836_DAC_WORD_LEN_MASK, word_len); + snd_soc_update_bits(codec, AD1836_DAC_CTRL1, AD1836_DAC_WORD_LEN_MASK, + word_len << AD1836_DAC_WORD_LEN_OFFSET); - snd_soc_update_bits(codec, AD1836_ADC_CTRL2, - AD1836_ADC_WORD_LEN_MASK, word_len); + snd_soc_update_bits(codec, AD1836_ADC_CTRL2, AD1836_ADC_WORD_LEN_MASK, + word_len << AD1836_ADC_WORD_OFFSET); return 0; } diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h index 845596717fdf..9d6a3f8f8aaf 100644 --- a/sound/soc/codecs/ad1836.h +++ b/sound/soc/codecs/ad1836.h @@ -25,6 +25,7 @@ #define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) #define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) #define AD1836_DAC_WORD_LEN_MASK 0x18 +#define AD1836_DAC_WORD_LEN_OFFSET 3 #define AD1836_DAC_CTRL2 1 #define AD1836_DACL1_MUTE 0 @@ -51,6 +52,7 @@ #define AD1836_ADCL2_MUTE 2 #define AD1836_ADCR2_MUTE 3 #define AD1836_ADC_WORD_LEN_MASK 0x30 +#define AD1836_ADC_WORD_OFFSET 5 #define AD1836_ADC_SERFMT_MASK (7 << 6) #define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) #define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) @@ -60,4 +62,8 @@ #define AD1836_NUM_REGS 16 +#define AD1836_WORD_LEN_24 0x0 +#define AD1836_WORD_LEN_20 0x1 +#define AD1836_WORD_LEN_16 0x2 + #endif -- cgit v1.2.3-58-ga151 From 0a1896b27b030529ec770aefd790544a1bdb7d5a Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Mon, 6 Jun 2011 18:55:34 -0400 Subject: ALSA: hda: Fix quirk for Dell Inspiron 910 BugLink: https://launchpad.net/bugs/792712 The original reporter states that sound from the internal speakers is inaudible until using the model=auto quirk. This symptom is due to an existing quirk mask for 0x102802b* that uses the model=dell quirk. To limit the possible regressions, leave the existing quirk mask but add a higher priority specific mask for the reporter's PCI SSID. Reported-and-tested-by: rodni hipp Cc: [2.6.38+] Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7a4e10002f56..d7007896772b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13860,6 +13860,7 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0x015b, "Acer Aspire One", ALC268_ACER_ASPIRE_ONE), SND_PCI_QUIRK(0x1028, 0x0253, "Dell OEM", ALC268_DELL), + SND_PCI_QUIRK(0x1028, 0x02b0, "Dell Inspiron 910", ALC268_AUTO), SND_PCI_QUIRK_MASK(0x1028, 0xfff0, 0x02b0, "Dell Inspiron Mini9/Vostro A90", ALC268_DELL), /* almost compatible with toshiba but with optional digital outs; -- cgit v1.2.3-58-ga151 From 064d58ee3afb8a865a72d24e069c7258ec38640e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 7 Jun 2011 10:24:46 +0200 Subject: ASoC: Blackfin: bf5xx-ad1836: Fix codec device name Fix the codec_name field of the dai_link to match the actual device name of the codec. Otherwise the card won't be instantiated. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/blackfin/bf5xx-ad1836.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index ea4951cf5526..f79d1655e035 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -75,7 +75,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .cpu_dai_name = "bfin-tdm.0", .codec_dai_name = "ad1836-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", + .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, }, { @@ -84,7 +84,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .cpu_dai_name = "bfin-tdm.1", .codec_dai_name = "ad1836-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", + .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, }, }; -- cgit v1.2.3-58-ga151 From 0f82bdf572fc6e42147151aa4d52542f7fc6d793 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 7 Jun 2011 23:42:04 +0100 Subject: ASoC: Fix WM8962 headphone volume update for use of advanced caches Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8962.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index f90ae427242b..5e05eed96c38 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1999,12 +1999,12 @@ static int wm8962_put_hp_sw(struct snd_kcontrol *kcontrol, return 0; /* If the left PGA is enabled hit that VU bit... */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTL_PGA_ENA) + if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTL_PGA_ENA) return snd_soc_write(codec, WM8962_HPOUTL_VOLUME, reg_cache[WM8962_HPOUTL_VOLUME]); /* ...otherwise the right. The VU is stereo. */ - if (reg_cache[WM8962_PWR_MGMT_2] & WM8962_HPOUTR_PGA_ENA) + if (snd_soc_read(codec, WM8962_PWR_MGMT_2) & WM8962_HPOUTR_PGA_ENA) return snd_soc_write(codec, WM8962_HPOUTR_VOLUME, reg_cache[WM8962_HPOUTR_VOLUME]); -- cgit v1.2.3-58-ga151 From 3115ae174620eeab4b16f52c8d0a9a35d2717e3c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 8 Jun 2011 18:07:49 +0100 Subject: ASoC: WM8804 does not support sample rates below 32kHz Reported-by: Kieran O'Leary Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8804.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 6785688f8806..9a5e67c5a6bd 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -680,20 +680,25 @@ static struct snd_soc_dai_ops wm8804_dai_ops = { #define WM8804_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE) +#define WM8804_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | \ + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000) + static struct snd_soc_dai_driver wm8804_dai = { .name = "wm8804-spdif", .playback = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = WM8804_RATES, .formats = WM8804_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = WM8804_RATES, .formats = WM8804_FORMATS, }, .ops = &wm8804_dai_ops, -- cgit v1.2.3-58-ga151 From 0cd114fff9ace7014c0d3ef8ab385fc5d3cf2d2f Mon Sep 17 00:00:00 2001 From: Timur Tabi Date: Wed, 8 Jun 2011 15:02:56 -0500 Subject: ASoC: fsl: fix initialization of DMA buffers The DMA (PCM) driver used by some Freescale PowerPC supports separate DAIs for playback and capture, so DMA buffers should be allocated only for the initialized streams. Instead of checking for the number of active channels, which apparently is not reliable, check to see if the actual stream object exists. Also provide a better name for the DMA interrupt. Signed-off-by: Timur Tabi Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_dma.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 15dac0f20cd8..6680c0b4d203 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -310,7 +310,7 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, * should allocate a DMA buffer only for the streams that are valid. */ - if (dai->driver->playback.channels_min) { + if (pcm->streams[0].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[0].substream->dma_buffer); @@ -320,13 +320,13 @@ static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, } } - if (dai->driver->capture.channels_min) { + if (pcm->streams[1].substream) { ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, card->dev, fsl_dma_hardware.buffer_bytes_max, &pcm->streams[1].substream->dma_buffer); if (ret) { - snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); dev_err(card->dev, "can't alloc capture dma buffer\n"); + snd_dma_free_pages(&pcm->streams[0].substream->dma_buffer); return ret; } } @@ -449,7 +449,8 @@ static int fsl_dma_open(struct snd_pcm_substream *substream) dma_private->ld_buf_phys = ld_buf_phys; dma_private->dma_buf_phys = substream->dma_buffer.addr; - ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "DMA", dma_private); + ret = request_irq(dma_private->irq, fsl_dma_isr, 0, "fsldma-audio", + dma_private); if (ret) { dev_err(dev, "can't register ISR for IRQ %u (ret=%i)\n", dma_private->irq, ret); -- cgit v1.2.3-58-ga151 From 4b80b8c2eee5282dab57f094fd3893c0c09f750c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 9 Jun 2011 13:22:36 +0200 Subject: ASoC: snd_soc_new_{mixer,mux,pga} make sure to use right DAPM context Currently it is possible that snd_soc_new_{mixer,mux,pga} is called with a DAPM context not matching the widgets context. This can lead to a wrong prefix_len calculation, which will result in undefined behaviour. To avoid this always use the DAPM context from the widget itself. Signed-off-by: Lars-Peter Clausen Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/soc-dapm.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 776e6f418306..32ab7fc4579a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -350,9 +350,9 @@ static int dapm_is_shared_kcontrol(struct snd_soc_dapm_context *dapm, } /* create new dapm mixer control */ -static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_mixer(struct snd_soc_dapm_widget *w) { + struct snd_soc_dapm_context *dapm = w->dapm; int i, ret = 0; size_t name_len, prefix_len; struct snd_soc_dapm_path *path; @@ -450,9 +450,9 @@ static int dapm_new_mixer(struct snd_soc_dapm_context *dapm, } /* create new dapm mux control */ -static int dapm_new_mux(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_mux(struct snd_soc_dapm_widget *w) { + struct snd_soc_dapm_context *dapm = w->dapm; struct snd_soc_dapm_path *path = NULL; struct snd_kcontrol *kcontrol; struct snd_card *card = dapm->card->snd_card; @@ -535,8 +535,7 @@ static int dapm_new_mux(struct snd_soc_dapm_context *dapm, } /* create new dapm volume control */ -static int dapm_new_pga(struct snd_soc_dapm_context *dapm, - struct snd_soc_dapm_widget *w) +static int dapm_new_pga(struct snd_soc_dapm_widget *w) { if (w->num_kcontrols) dev_err(w->dapm->dev, @@ -1826,13 +1825,13 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_mixer: case snd_soc_dapm_mixer_named_ctl: w->power_check = dapm_generic_check_power; - dapm_new_mixer(dapm, w); + dapm_new_mixer(w); break; case snd_soc_dapm_mux: case snd_soc_dapm_virt_mux: case snd_soc_dapm_value_mux: w->power_check = dapm_generic_check_power; - dapm_new_mux(dapm, w); + dapm_new_mux(w); break; case snd_soc_dapm_adc: case snd_soc_dapm_aif_out: @@ -1845,7 +1844,7 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm) case snd_soc_dapm_pga: case snd_soc_dapm_out_drv: w->power_check = dapm_generic_check_power; - dapm_new_pga(dapm, w); + dapm_new_pga(w); break; case snd_soc_dapm_input: case snd_soc_dapm_output: -- cgit v1.2.3-58-ga151 From 33195500edf260e8c8809ab9dfc67f50e0ce031f Mon Sep 17 00:00:00 2001 From: Sangbeom Kim Date: Fri, 10 Jun 2011 10:36:54 +0900 Subject: ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register If DMA active status should be checked, I2SCON register should be referenced. In this patch, Fix the incorrect referencing of I2SCON register. Reported-by : Lakkyung Jung Signed-off-by: Sangbeom Kim Acked-by: Jassi Brar Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/samsung/i2s.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index ffa09b3b2caa..992a732b5211 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -191,7 +191,7 @@ static inline bool tx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SMOD); + active = readl(i2s->addr + I2SCON); if (is_secondary(i2s)) active &= CON_TXSDMA_ACTIVE; @@ -223,7 +223,7 @@ static inline bool rx_active(struct i2s_dai *i2s) if (!i2s) return false; - active = readl(i2s->addr + I2SMOD) & CON_RXDMA_ACTIVE; + active = readl(i2s->addr + I2SCON) & CON_RXDMA_ACTIVE; return active ? true : false; } -- cgit v1.2.3-58-ga151 From 20f5e0b36d968326fab3b720035f226113e34ae9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Jun 2011 09:31:54 +0200 Subject: ALSA: hda - Fix invalid unsol tag for some alc262 model quirks The tag number was forgotten to be fixed after cleaning up the model quirks for ALC262 fujitsu and lenovo-3000 models. Tested-by: Michal Hocko Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d7007896772b..ca211c1cba03 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11924,7 +11924,7 @@ static const struct hda_verb alc262_nec_verbs[] = { * 0x1b = port replicator headphone out */ -#define ALC_HP_EVENT 0x37 +#define ALC_HP_EVENT ALC880_HP_EVENT static const struct hda_verb alc262_fujitsu_unsol_verbs[] = { {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC_HP_EVENT}, -- cgit v1.2.3-58-ga151 From c0a20263dbe1fc5f394913d71063c9cd8282c5db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Jun 2011 15:28:15 +0200 Subject: ALSA: hda - Fix initialization of hp pins with master_mute in Realtek Some Reatlek model quirks use master_mute bool switch for controlling the master-mute of outputs. For these cases, the initialization of HP pins/amps were forgotten during the transition to the common automute helper function in 3.0 development time, and resulted in the muted HP output as default. This patch fixes the issue by adjusting the HP output explicitly with master_mute switch. Tested-by: Michal Hocko Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ca211c1cba03..43fcfbd32847 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1141,6 +1141,13 @@ static void update_speakers(struct hda_codec *codec) struct alc_spec *spec = codec->spec; int on; + /* Control HP pins/amps depending on master_mute state; + * in general, HP pins/amps control should be enabled in all cases, + * but currently set only for master_mute, just to be safe + */ + do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), + spec->autocfg.hp_pins, spec->master_mute, true); + if (!spec->automute) on = 0; else @@ -6201,11 +6208,6 @@ static const struct snd_kcontrol_new alc260_input_mixer[] = { /* update HP, line and mono out pins according to the master switch */ static void alc260_hp_master_update(struct hda_codec *codec) { - struct alc_spec *spec = codec->spec; - - /* change HP pins */ - do_automute(codec, ARRAY_SIZE(spec->autocfg.hp_pins), - spec->autocfg.hp_pins, spec->master_mute, true); update_speakers(codec); } -- cgit v1.2.3-58-ga151 From 890ee02ac12c02c4712b6d7dd062ff4d6d37691c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 10 Jun 2011 15:32:31 +0200 Subject: ALSA: Use %pV for snd_printk() Clean up snd_printk() helper using the %pV prefix for recursive printks. This also automagically fixes an Oops with RO/NX-enabled modules. Tested-by: Maarten Lankhorst Signed-off-by: Takashi Iwai --- sound/core/misc.c | 40 +++++++++++++++++----------------------- 1 file changed, 17 insertions(+), 23 deletions(-) (limited to 'sound') diff --git a/sound/core/misc.c b/sound/core/misc.c index 2c41825c836e..eb9fe2e1d291 100644 --- a/sound/core/misc.c +++ b/sound/core/misc.c @@ -58,26 +58,6 @@ static const char *sanity_file_name(const char *path) else return path; } - -/* print file and line with a certain printk prefix */ -static int print_snd_pfx(unsigned int level, const char *path, int line, - const char *format) -{ - const char *file = sanity_file_name(path); - char tmp[] = "<0>"; - const char *pfx = level ? KERN_DEBUG : KERN_DEFAULT; - int ret = 0; - - if (format[0] == '<' && format[2] == '>') { - tmp[1] = format[1]; - pfx = tmp; - ret = 1; - } - printk("%sALSA %s:%d: ", pfx, file, line); - return ret; -} -#else -#define print_snd_pfx(level, path, line, format) 0 #endif #if defined(CONFIG_SND_DEBUG) || defined(CONFIG_SND_VERBOSE_PRINTK) @@ -85,15 +65,29 @@ void __snd_printk(unsigned int level, const char *path, int line, const char *format, ...) { va_list args; - +#ifdef CONFIG_SND_VERBOSE_PRINTK + struct va_format vaf; + char verbose_fmt[] = KERN_DEFAULT "ALSA %s:%d %pV"; +#endif + #ifdef CONFIG_SND_DEBUG if (debug < level) return; #endif + va_start(args, format); - if (print_snd_pfx(level, path, line, format)) - format += 3; /* skip the printk level-prefix */ +#ifdef CONFIG_SND_VERBOSE_PRINTK + vaf.fmt = format; + vaf.va = &args; + if (format[0] == '<' && format[2] == '>') { + memcpy(verbose_fmt, format, 3); + vaf.fmt = format + 3; + } else if (level) + memcpy(verbose_fmt, KERN_DEBUG, 3); + printk(verbose_fmt, sanity_file_name(path), line, &vaf); +#else vprintk(format, args); +#endif va_end(args); } EXPORT_SYMBOL_GPL(__snd_printk); -- cgit v1.2.3-58-ga151 From 7ab1fc0af3464d231e17eb729a03495d93d0cc5c Mon Sep 17 00:00:00 2001 From: Daniel T Chen Date: Fri, 10 Jun 2011 10:14:01 -0400 Subject: ALSA: hda: Fix inaudible internal speakers on CyberpowerPC Gamer Xplorer N57001 laptop BugLink: https://launchpad.net/bugs/761171 The original reporter needs the model=auto quirk for his internal speakers to be audible in the latest daily snapshot, so add an entry in the quirk table for his PCI SSID. A trivially different version of this patch using the model=asus quirk should be applied to the 2.6.38 and 2.6.39 stable kernels. We don't use the asus quirk in 3.0-rc2, because 3.0-rc2's autoparser is much improved. Reported-and-tested-by: tomdeering7 Signed-off-by: Daniel T Chen Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 3e6b9a8539c2..694b9daf691f 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3102,6 +3102,7 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo", CXT5066_IDEAPAD), /* Fallback for Lenovos without dock mic */ + SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; -- cgit v1.2.3-58-ga151 From c0da00145f9a32ef33b14508e6fd90fc130afbdc Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Sun, 12 Jun 2011 17:26:17 +0200 Subject: ALSA: hdspm - Fix locking in snd_hdspm_midi_input_read For the MIDI part, we need to acquire (and release) the hmidi->lock, access to the global hdspm structure is serialized through hmidi->hdspm->lock instead. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 949691a876d3..32d80af012cc 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1639,12 +1639,14 @@ static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi) } } hmidi->pending = 0; + spin_unlock_irqrestore(&hmidi->lock, flags); + spin_lock_irqsave(&hmidi->hdspm->lock, flags); hmidi->hdspm->control_register |= hmidi->ie; hdspm_write(hmidi->hdspm, HDSPM_controlRegister, hmidi->hdspm->control_register); + spin_unlock_irqrestore(&hmidi->hdspm->lock, flags); - spin_unlock_irqrestore (&hmidi->lock, flags); return snd_hdspm_midi_output_write (hmidi); } -- cgit v1.2.3-58-ga151 From fedf1535ab5ee02acbbc235c2272d84bb9334758 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Sun, 12 Jun 2011 17:26:18 +0200 Subject: ALSA: hdspm - Fix jumping external wordclock frequency in AutoSync mode When using Word Clock on RME MADI cards, AutoSync mode was alternating betweeen MADI and WC due to a typo: AutoSync is indicated in the second status register (status2), not the first one (status). While the proc output was always correct, the reported WC frequency to ALSA was unstable as mentioned in http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006723.html Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 32d80af012cc..d03ef94d570e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -1143,7 +1143,7 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) /* if wordclock has synced freq and wordclock is valid */ if ((status2 & HDSPM_wcLock) != 0 && - (status & HDSPM_SelSyncRef0) == 0) { + (status2 & HDSPM_SelSyncRef0) == 0) { rate_bits = status2 & HDSPM_wcFreqMask; -- cgit v1.2.3-58-ga151 From efef054e8c4bc4fd48a0b4deb5491116d9f557c7 Mon Sep 17 00:00:00 2001 From: Adrian Knoth Date: Sun, 12 Jun 2011 17:26:19 +0200 Subject: ALSA: hdspm - Add firmware revision ID for RME MADI PCI version The PCI version of the RME HDSP MADI card uses 0xcf as revision ID. Just add this to the list of supported cards. Signed-off-by: Adrian Knoth Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index d03ef94d570e..3f08afc0f0d3 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -521,6 +521,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_DMA_AREA_KILOBYTES (HDSPM_DMA_AREA_BYTES/1024) /* revisions >= 230 indicate AES32 card */ +#define HDSPM_MADI_OLD_REV 207 #define HDSPM_MADI_REV 210 #define HDSPM_RAYDAT_REV 211 #define HDSPM_AIO_REV 212 @@ -6379,6 +6380,7 @@ static int __devinit snd_hdspm_create(struct snd_card *card, switch (hdspm->firmware_rev) { case HDSPM_MADI_REV: + case HDSPM_MADI_OLD_REV: hdspm->io_type = MADI; hdspm->card_name = "RME MADI"; hdspm->midiPorts = 3; -- cgit v1.2.3-58-ga151 From ac5d4b404e78bd7eb67fc70c2acb437a25497e98 Mon Sep 17 00:00:00 2001 From: Florian Zeitz Date: Sun, 12 Jun 2011 01:15:42 +0200 Subject: ALSA: emu10k1: Add details for E-mu 0404 PCIe version This patch adds the necessary details to support the PCIe version of E-MU's 0404 card. From comparing the PCBs it seems the PCIe version just added a PCIe chipset and left all other components pretty much in place. For anyone intrigued to take a look at the PCB there are pictures I took at . Signed-off-by: Florian Zeitz Signed-off-by: Takashi Iwai --- sound/pci/emu10k1/emu10k1_main.c | 8 ++++++++ 1 file changed, 8 insertions(+) (limited to 'sound') diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 5e619a84da06..15f0161ce4a2 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1440,6 +1440,14 @@ static struct snd_emu_chip_details emu_chip_details[] = { .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ + /* EMU0404 PCIe */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40051102, + .driver = "Audigy2", .name = "E-mu 0404 PCIe [MAEM8984]", + .id = "EMU0404", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 PCIe ver_03 */ /* Note that all E-mu cards require kernel 2.6 or newer. */ {.vendor = 0x1102, .device = 0x0008, .driver = "Audigy2", .name = "SB Audigy 2 Value [Unknown]", -- cgit v1.2.3-58-ga151 From 54463a66b91cf491a7c9af612b0e310babc5fa24 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 13 Jun 2011 08:32:06 +0200 Subject: ALSA: hda - Fix wrong auto-mute type for Acer Aspire-one The auto-mute setup for Acer Aspire-one with ALC268 was set wrongly during the clean-up of auto-mute function. Fixed now. Tested-by: Borislav Petkov Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 43fcfbd32847..61a774b3d3cb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13316,9 +13316,8 @@ static void alc268_acer_lc_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute_mixer_nid[0] = 0x0f; spec->automute = 1; - spec->automute_mode = ALC_AUTOMUTE_MIXER; + spec->automute_mode = ALC_AUTOMUTE_AMP; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; spec->int_mic.pin = 0x12; -- cgit v1.2.3-58-ga151 From 2308f4add3de9f6c9c9f02e49461e94d84bb200a Mon Sep 17 00:00:00 2001 From: Joe Perches Date: Sun, 12 Jun 2011 13:02:43 -0700 Subject: ALSA: hda - Fix beep_device compilation warnings MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Using static inline functions can reduce compilation messages and macro misuse. sound/pci/hda/patch_conexant.c: In function ‘patch_cxt5045’: sound/pci/hda/patch_conexant.c:1232:3: warning: statement with no effect Signed-off-by: Joe Perches Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.h | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index f1de1bac042c..4967eabe774e 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -50,7 +50,12 @@ int snd_hda_enable_beep_device(struct hda_codec *codec, int enable); int snd_hda_attach_beep_device(struct hda_codec *codec, int nid); void snd_hda_detach_beep_device(struct hda_codec *codec); #else -#define snd_hda_attach_beep_device(...) 0 -#define snd_hda_detach_beep_device(...) +static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) +{ + return 0; +} +void snd_hda_detach_beep_device(struct hda_codec *codec) +{ +} #endif #endif -- cgit v1.2.3-58-ga151 From e9c039052be59753e6bcc7c8b59763899dc1161c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Jun 2011 19:05:58 +0100 Subject: ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O bus This will be removed in -next so let's drop it from mainline as soon as we can in order to minimise surprises. Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +-- sound/soc/soc-cache.c | 3 --- 2 files changed, 1 insertion(+), 5 deletions(-) (limited to 'sound') diff --git a/include/sound/soc.h b/include/sound/soc.h index f1de3e0c75bc..3a4bd3a3c68d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -248,8 +248,7 @@ typedef int (*hw_write_t)(void *,const char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; enum snd_soc_control_type { - SND_SOC_CUSTOM = 1, - SND_SOC_I2C, + SND_SOC_I2C = 1, SND_SOC_SPI, }; diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index c005ceb70c9d..039b9532b270 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -409,9 +409,6 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, codec->bulk_write_raw = snd_soc_hw_bulk_write_raw; switch (control) { - case SND_SOC_CUSTOM: - break; - case SND_SOC_I2C: #if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) codec->hw_write = (hw_write_t)i2c_master_send; -- cgit v1.2.3-58-ga151 From 37f7ec38ea5c31180461f82e895e13fdd549b595 Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Mon, 13 Jun 2011 23:52:02 +0200 Subject: ALSA: 6fire: Fix double-free bug in usb6fire_fw_ezusb_upload() We have a double-free bug in sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload(). We already call release_firmware(fw) on line 258, so when we then do it again after usb6fire_fw_ezusb_write() returns <0, we have a double-free. Easily fixed by just removing the last call to release_firmware(). Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/usb/6fire/firmware.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index a91719d5918b..1e3ae3327dd3 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -270,7 +270,6 @@ static int usb6fire_fw_ezusb_upload( data = 0x00; /* resume ezusb cpu */ ret = usb6fire_fw_ezusb_write(device, 0xa0, 0xe600, &data, 1); if (ret < 0) { - release_firmware(fw); snd_printk(KERN_ERR PREFIX "unable to upload ezusb " "firmware %s: end message.\n", fwname); return ret; -- cgit v1.2.3-58-ga151 From ca2585afa013ec2cf99a48e46d6b82df2e240493 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 14 Jun 2011 08:14:32 +0200 Subject: ALSA: hda - Fix missing static inline to beep dummy function The commit 2308f4add3de9f6c9c9f02e49461e94d84bb200a missed static inline thus it resulted in multiple-definitions error at linking. Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_beep.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_beep.h b/sound/pci/hda/hda_beep.h index 4967eabe774e..55f0647458c7 100644 --- a/sound/pci/hda/hda_beep.h +++ b/sound/pci/hda/hda_beep.h @@ -54,7 +54,7 @@ static inline int snd_hda_attach_beep_device(struct hda_codec *codec, int nid) { return 0; } -void snd_hda_detach_beep_device(struct hda_codec *codec) +static inline void snd_hda_detach_beep_device(struct hda_codec *codec) { } #endif -- cgit v1.2.3-58-ga151 From e72888e91cc902ccdc089f237b6eed7587e2b4df Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 15 Jun 2011 15:14:49 +0200 Subject: ALSA: lola - Fix section mismatch Add missing __devinit. Signed-off-by: Takashi Iwai --- sound/pci/lola/lola.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/lola/lola.c b/sound/pci/lola/lola.c index 34b24286d279..2692e5ae5f2d 100644 --- a/sound/pci/lola/lola.c +++ b/sound/pci/lola/lola.c @@ -445,7 +445,7 @@ static void lola_reset_setups(struct lola *chip) lola_setup_all_analog_gains(chip, PLAY, false); /* output, update */ } -static int lola_parse_tree(struct lola *chip) +static int __devinit lola_parse_tree(struct lola *chip) { unsigned int val; int nid, err; -- cgit v1.2.3-58-ga151 From 0ec5258d68c626922d92e2f0e4e5c689e5360a5d Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Thu, 16 Jun 2011 21:06:27 +0200 Subject: ALSA: 6fire - Fix signedness bug Fixed remaining issues of the signedness bug discovered by Dan Carpenter. A check was remaining that tests if unsigned rt->rate is >= 0. Changed that so that rt->rate now consistently uses ARRAY_SIZE(rates) as invalid rate value and not -1. Signed-off-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index b137b25865cc..d144cdb2f159 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -395,12 +395,12 @@ static int usb6fire_pcm_open(struct snd_pcm_substream *alsa_sub) alsa_rt->hw = pcm_hw; if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (rt->rate >= 0) + if (rt->rate < ARRAY_SIZE(rates)) alsa_rt->hw.rates = rates_alsaid[rt->rate]; alsa_rt->hw.channels_max = OUT_N_CHANNELS; sub = &rt->playback; } else if (alsa_sub->stream == SNDRV_PCM_STREAM_CAPTURE) { - if (rt->rate >= 0) + if (rt->rate < ARRAY_SIZE(rates)) alsa_rt->hw.rates = rates_alsaid[rt->rate]; alsa_rt->hw.channels_max = IN_N_CHANNELS; sub = &rt->capture; -- cgit v1.2.3-58-ga151 From cf6f1ff17f56c275424c5a341fc4d27ccbbfa71c Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Fri, 17 Jun 2011 08:18:35 +0200 Subject: ALSA: isight: adjust for new queueing API Since commit 13882a82ee16 (optimize iso queueing by setting wake only after the last packet), drivers are required to call fw_iso_context_queue_flush() after queueing a batch of packets. The missing call would have an effect only if the controller queue underruns, but then the DMA would stop completely. Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/firewire/isight.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/firewire/isight.c b/sound/firewire/isight.c index 86ee16ca365e..440030818db7 100644 --- a/sound/firewire/isight.c +++ b/sound/firewire/isight.c @@ -209,6 +209,7 @@ static void isight_packet(struct fw_iso_context *context, u32 cycle, isight->packet_index = -1; return; } + fw_iso_context_queue_flush(isight->context); if (++index >= QUEUE_LENGTH) index = 0; -- cgit v1.2.3-58-ga151 From ad2409413d09fca763be1ac5161f2a9d82367903 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 17 Jun 2011 14:23:46 +0200 Subject: ALSA: hda - Fix no NID error with VIA codecs The via driver spews warnigs like hda-codec: no NID for mapping control Independent HP:0:0 with some codecs because snd_hda_add_nid() is called with nid=0. This patch fixes it by skipping the call when no corresponding widget is found. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 605c99e1e520..c952582fb218 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -832,10 +832,13 @@ static int via_hp_build(struct hda_codec *codec) knew->subdevice = HDA_SUBDEV_NID_FLAG | nid; knew->private_value = nid; - knew = via_clone_control(spec, &via_hp_mixer[1]); - if (knew == NULL) - return -ENOMEM; - knew->subdevice = side_mute_channel(spec); + nid = side_mute_channel(spec); + if (nid) { + knew = via_clone_control(spec, &via_hp_mixer[1]); + if (knew == NULL) + return -ENOMEM; + knew->subdevice = nid; + } return 0; } -- cgit v1.2.3-58-ga151 From 6f2e810ad5d162c2bfa063c1811087277b299e4e Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Mon, 20 Jun 2011 10:27:07 +0200 Subject: ALSA: HDA: Remove quirk for an HP device The reporter, who is running kernel 2.6.38, reports that he needs to set model=auto for the headphone output to work correctly. BugLink: http://bugs.launchpad.net/bugs/761022 Cc: stable@kernel.org (v2.6.38+) Reported-by: Jo Signed-off-by: David Henningsson 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 61a774b3d3cb..c923b2cc9e53 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4883,7 +4883,6 @@ static const struct snd_pci_quirk alc880_cfg_tbl[] = { SND_PCI_QUIRK(0x1025, 0xe309, "ULI", ALC880_3ST_DIG), SND_PCI_QUIRK(0x1025, 0xe310, "ULI", ALC880_3ST), SND_PCI_QUIRK(0x1039, 0x1234, NULL, ALC880_6ST_DIG), - SND_PCI_QUIRK(0x103c, 0x2a09, "HP", ALC880_5ST), SND_PCI_QUIRK(0x1043, 0x10b3, "ASUS W1V", ALC880_ASUS_W1V), SND_PCI_QUIRK(0x1043, 0x10c2, "ASUS W6A", ALC880_ASUS_DIG), SND_PCI_QUIRK(0x1043, 0x10c3, "ASUS Wxx", ALC880_ASUS_DIG), -- cgit v1.2.3-58-ga151 From c933790614529c06b221f73ff36e2456aecee30d Mon Sep 17 00:00:00 2001 From: Tony Vroon Date: Mon, 20 Jun 2011 22:11:11 +0100 Subject: ALSA: hda - Remove ALC268 model override for CPR2000 The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER. This keeps headphone automute and microphone input from operating on at least one laptop from Opti Systems. Without the override, the BIOS parser does a fine job setting the card up and everything works. Tested-By: Peter Schneider Signed-off-by: Tony Vroon 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 c923b2cc9e53..475ed1e8ffc6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -13871,7 +13871,6 @@ static const struct snd_pci_quirk alc268_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC268_3ST), SND_PCI_QUIRK(0x1170, 0x0040, "ZEPTO", ALC268_ZEPTO), SND_PCI_QUIRK(0x14c0, 0x0025, "COMPAL IFL90/JFL-92", ALC268_TOSHIBA), - SND_PCI_QUIRK(0x152d, 0x0763, "Diverse (CPR2000)", ALC268_ACER), SND_PCI_QUIRK(0x152d, 0x0771, "Quanta IL1", ALC267_QUANTA_IL1), {} }; -- cgit v1.2.3-58-ga151 From 42467b32ce4f1ba933673b396f807110e3618ff5 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Mon, 20 Jun 2011 14:14:37 +0800 Subject: ALSA: VIA HDA: Modify initial verbs list for VT1718S. Remove some invalid initial verbs and correct some wrong initial verbs for VT1718S codec. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index c952582fb218..abee9ac15902 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4283,9 +4283,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = { {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, {0x21, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(5)}, - - /* Setup default input of Front HP to MW9 */ - {0x28, AC_VERB_SET_CONNECT_SEL, 0x1}, /* PW9 PW10 Output enable */ {0x2d, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, {0x2e, AC_VERB_SET_PIN_WIDGET_CONTROL, AC_PINCTL_OUT_EN}, @@ -4294,10 +4291,10 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = { /* Enable Boost Volume backdoor */ {0x1, 0xf88, 0x8}, /* MW0/1/2/3/4: un-mute index 0 (AOWx), mute index 1 (MW9) */ - {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, @@ -4307,8 +4304,6 @@ static const struct hda_verb vt1718S_volume_init_verbs[] = { /* set MUX1 = 2 (AOW4), MUX2 = 1 (AOW3) */ {0x34, AC_VERB_SET_CONNECT_SEL, 0x2}, {0x35, AC_VERB_SET_CONNECT_SEL, 0x1}, - /* Unmute MW4's index 0 */ - {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { } }; -- cgit v1.2.3-58-ga151 From ba31a60d0fd8a3976d44d32f2b82491c62646b2a Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Mon, 20 Jun 2011 14:16:33 +0800 Subject: ALSA: VIA HDA: Mute/unmute mixer conncted to Headphone for VT1718S. When switch HP independent mode, mute/unmute connctions of mixer which is connected to headphone for VT1718S. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 13 ++++++++++++- 1 file changed, 12 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index abee9ac15902..f1a80cd6afe4 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -745,12 +745,23 @@ static int via_independent_hp_put(struct snd_kcontrol *kcontrol, struct via_spec *spec = codec->spec; hda_nid_t nid = kcontrol->private_value; unsigned int pinsel = ucontrol->value.enumerated.item[0]; + unsigned int parm0, parm1; /* Get Independent Mode index of headphone pin widget */ spec->hp_independent_mode = spec->hp_independent_mode_index == pinsel ? 1 : 0; - if (spec->codec_type == VT1718S) + if (spec->codec_type == VT1718S) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel ? 2 : 0); + /* Set correct mute switch for MW3 */ + parm0 = spec->hp_independent_mode ? + AMP_IN_UNMUTE(0) : AMP_IN_MUTE(0); + parm1 = spec->hp_independent_mode ? + AMP_IN_MUTE(1) : AMP_IN_UNMUTE(1); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm0); + snd_hda_codec_write(codec, 0x1b, 0, + AC_VERB_SET_AMP_GAIN_MUTE, parm1); + } else snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, pinsel); -- cgit v1.2.3-58-ga151 From e905a83acd7bf8989c3d5ba3099b72675f5d7d29 Mon Sep 17 00:00:00 2001 From: Lydia Wang Date: Mon, 20 Jun 2011 14:17:56 +0800 Subject: ALSA: VIA HDA: Create a master amplifier control for VT1718S. Create a master volume and mute control of playback for VT1718S. Signed-off-by: Lydia Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_via.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index f1a80cd6afe4..f43bb0eaed8b 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -4462,6 +4462,19 @@ static int vt1718S_auto_create_multi_out_ctls(struct via_spec *spec, if (err < 0) return err; } else if (i == AUTO_SEQ_FRONT) { + /* add control to mixer index 0 */ + err = via_add_control(spec, VIA_CTL_WIDGET_VOL, + "Master Front Playback Volume", + HDA_COMPOSE_AMP_VAL(0x21, 3, 5, + HDA_INPUT)); + if (err < 0) + return err; + err = via_add_control(spec, VIA_CTL_WIDGET_MUTE, + "Master Front Playback Switch", + HDA_COMPOSE_AMP_VAL(0x21, 3, 5, + HDA_INPUT)); + if (err < 0) + return err; /* Front */ sprintf(name, "%s Playback Volume", chname[i]); err = via_add_control( -- cgit v1.2.3-58-ga151 From d2a19da79d3ea5b7859248b0f132c479ed4505e2 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Wed, 22 Jun 2011 09:58:37 +0200 Subject: ALSA: HDA: Pinfix quirk for HP Z200 Workstation BIOS lists the internal speaker as an internal line-out. Change to internal speaker + model=auto for better auto-mute capabilities. BugLink: http://bugs.launchpad.net/bugs/754964 Reported-by: Marc Legris Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 475ed1e8ffc6..d21191dcfe88 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12599,6 +12599,7 @@ static const struct hda_verb alc262_toshiba_rx1_unsol_verbs[] = { */ enum { PINFIX_FSC_H270, + PINFIX_HP_Z200, }; static const struct alc_fixup alc262_fixups[] = { @@ -12611,9 +12612,17 @@ static const struct alc_fixup alc262_fixups[] = { { } } }, + [PINFIX_HP_Z200] = { + .type = ALC_FIXUP_PINS, + .v.pins = (const struct alc_pincfg[]) { + { 0x16, 0x99130120 }, /* internal speaker */ + { } + } + }, }; static const struct snd_pci_quirk alc262_fixup_tbl[] = { + SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", PINFIX_HP_Z200), SND_PCI_QUIRK(0x1734, 0x1147, "FSC Celsius H270", PINFIX_FSC_H270), {} }; @@ -12730,6 +12739,8 @@ static const struct snd_pci_quirk alc262_cfg_tbl[] = { ALC262_HP_BPC), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1500, "HP z series", ALC262_HP_BPC), + SND_PCI_QUIRK(0x103c, 0x170b, "HP Z200", + ALC262_AUTO), SND_PCI_QUIRK_MASK(0x103c, 0xff00, 0x1700, "HP xw series", ALC262_HP_BPC), SND_PCI_QUIRK(0x103c, 0x2800, "HP D7000", ALC262_HP_BPC_D7000_WL), -- cgit v1.2.3-58-ga151 From f6d96e0da1ee3cfe67b719570fba3bb2ea057131 Mon Sep 17 00:00:00 2001 From: "Arnaud Patard (Rtp)" Date: Wed, 22 Jun 2011 22:21:48 +0200 Subject: ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entry SND_MXC_SOC_SSI looks to be unused, so kill it. Signed-off-by: Arnaud Patard Acked-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/Kconfig | 7 ------- 1 file changed, 7 deletions(-) (limited to 'sound') diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index d8f130d39dd9..bb699bb55a50 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -11,9 +11,6 @@ menuconfig SND_IMX_SOC if SND_IMX_SOC -config SND_MXC_SOC_SSI - tristate - config SND_MXC_SOC_FIQ tristate @@ -24,7 +21,6 @@ config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Enable support for audio on the i.MX31ADS with the WM1133-EV1 @@ -34,7 +30,6 @@ config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 select SND_SOC_TVL320AIC32X4 - select SND_MXC_SOC_SSI select SND_MXC_SOC_MX2 help Say Y if you want to add support for SoC audio on Visstrim SM10 @@ -44,7 +39,6 @@ config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 select SND_SOC_WM9712 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -57,7 +51,6 @@ config SND_SOC_EUKREA_TLV320 || MACH_EUKREA_MBIMXSD35_BASEBOARD \ || MACH_EUKREA_MBIMXSD51_BASEBOARD select SND_SOC_TLV320AIC23 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Enable I2S based access to the TLV320AIC23B codec attached -- cgit v1.2.3-58-ga151 From 96dcabb99b9f63f2b65f2b0bfe5d4eb48f11b177 Mon Sep 17 00:00:00 2001 From: "Arnaud Patard (Rtp)" Date: Wed, 22 Jun 2011 22:21:49 +0200 Subject: ASoC: imx: add missing module informations - add some modules aliases - add module license to avoid tainted kernel when loading the imx-pcm-audio driver Signed-off-by: Arnaud Patard Acked-by: Sascha Hauer Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/imx/imx-pcm-dma-mx2.c | 2 ++ sound/soc/imx/imx-ssi.c | 2 +- 2 files changed, 3 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index aab7765f401a..4173b3d87f97 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -337,3 +337,5 @@ static void __exit snd_imx_pcm_exit(void) platform_driver_unregister(&imx_pcm_driver); } module_exit(snd_imx_pcm_exit); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:imx-pcm-audio"); diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 5b13feca7537..61fceb09cdb5 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -774,4 +774,4 @@ module_exit(imx_ssi_exit); MODULE_AUTHOR("Sascha Hauer, "); MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface"); MODULE_LICENSE("GPL"); - +MODULE_ALIAS("platform:imx-ssi"); -- cgit v1.2.3-58-ga151 From 53dea36c70c1857149a8c447224e3936eb8b5339 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 22 Jun 2011 20:48:25 +0200 Subject: ASoC: pxa-ssp: Correct check for stream presence Don't rely on the codec's channels_min information to decide wheter or not allocate a substream's DMA buffer. Rather check if the substream itself was allocated previously. Signed-off-by: Daniel Mack Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/pxa/pxa2xx-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 2ce0b2d891d5..fab20a54e863 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -95,14 +95,14 @@ static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, if (!card->dev->coherent_dma_mask) card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - if (dai->driver->playback.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_PLAYBACK); if (ret) goto out; } - if (dai->driver->capture.channels_min) { + if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { ret = pxa2xx_pcm_preallocate_dma_buffer(pcm, SNDRV_PCM_STREAM_CAPTURE); if (ret) -- cgit v1.2.3-58-ga151 From 16866741bda5d16f3d30d1656ce941faf5dad34c Mon Sep 17 00:00:00 2001 From: Jesper Juhl Date: Thu, 23 Jun 2011 23:54:40 +0200 Subject: ALSA: Remove unneeded version.h includes from sound/ In the sound/ directory there are two files (flagged by 'make versioncheck'); sound/pci/asihpi/asihpi.c and sound/soc/codecs/wm8991.c that include linux/version.h although they don't need it. This patch removes the unneeded includes. Signed-off-by: Jesper Juhl Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 1 - sound/soc/codecs/wm8991.c | 1 - 2 files changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 2ca6f4f85b41..e3569bdd3b64 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -27,7 +27,6 @@ #include "hpioctl.h" #include -#include #include #include #include diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 3c2ee1bb73cd..6af23d06870f 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -13,7 +13,6 @@ #include #include -#include #include #include #include -- cgit v1.2.3-58-ga151 From f0ca89b031d327b80b612a0608d31b8e13e6dc33 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 21 Jun 2011 20:51:34 +0200 Subject: ALSA: HDA: Add a new Conexant codec ID (506c) Conexant ID 506c was found on Acer Aspire 3830TG. As users report no playback, sending to stable should be safe. Cc: stable@kernel.org BugLink: https://bugs.launchpad.net/bugs/783582 Reported-by: andROOM Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 694b9daf691f..4158949ea078 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4389,6 +4389,8 @@ static const struct hda_codec_preset snd_hda_preset_conexant[] = { .patch = patch_cxt5066 }, { .id = 0x14f15069, .name = "CX20585", .patch = patch_cxt5066 }, + { .id = 0x14f1506c, .name = "CX20588", + .patch = patch_cxt5066 }, { .id = 0x14f1506e, .name = "CX20590", .patch = patch_cxt5066 }, { .id = 0x14f15097, .name = "CX20631", @@ -4417,6 +4419,7 @@ MODULE_ALIAS("snd-hda-codec-id:14f15066"); MODULE_ALIAS("snd-hda-codec-id:14f15067"); MODULE_ALIAS("snd-hda-codec-id:14f15068"); MODULE_ALIAS("snd-hda-codec-id:14f15069"); +MODULE_ALIAS("snd-hda-codec-id:14f1506c"); MODULE_ALIAS("snd-hda-codec-id:14f1506e"); MODULE_ALIAS("snd-hda-codec-id:14f15097"); MODULE_ALIAS("snd-hda-codec-id:14f15098"); -- cgit v1.2.3-58-ga151 From 9966db22caf8f74c0e6d84a569e6d7d56332e127 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Tue, 21 Jun 2011 21:01:52 +0200 Subject: ALSA: HDA: Add model=auto quirk for Acer Aspire 3830TG Since we're not using the new auto parser as a fallback yet, add it manually as a quirk. Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4158949ea078..7bbc5f237a5e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3074,6 +3074,7 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { }; static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT5066_AUTO), SND_PCI_QUIRK_MASK(0x1025, 0xff00, 0x0400, "Acer", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), -- cgit v1.2.3-58-ga151 From 0cfae7c9378cf77434f6be89b5fb65d8f9a5031f Mon Sep 17 00:00:00 2001 From: Hans-Christian Egtvedt Date: Tue, 28 Jun 2011 16:59:14 +0200 Subject: ALSA: atmel - update author email for ABDAC, AC97C and AT73C213 This patch updates the email address of the sound drivers supported by me to an email account I will use on a more regular basis in the future. Signed-off-by: Hans-Christian Egtvedt Signed-off-by: Takashi Iwai --- sound/atmel/abdac.c | 2 +- sound/atmel/ac97c.c | 2 +- sound/spi/at73c213.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 6e2409181895..bfee60c4d4c0 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -599,4 +599,4 @@ module_exit(atmel_abdac_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Driver for Atmel Audio Bitstream DAC (ABDAC)"); -MODULE_AUTHOR("Hans-Christian Egtvedt "); +MODULE_AUTHOR("Hans-Christian Egtvedt "); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b310702c646e..ac35222ad0dd 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1199,4 +1199,4 @@ module_exit(atmel_ac97c_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Driver for Atmel AC97 controller"); -MODULE_AUTHOR("Hans-Christian Egtvedt "); +MODULE_AUTHOR("Hans-Christian Egtvedt "); diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 337a00241a1f..4dd051bdf4fd 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1124,6 +1124,6 @@ static void __exit at73c213_exit(void) } module_exit(at73c213_exit); -MODULE_AUTHOR("Hans-Christian Egtvedt "); +MODULE_AUTHOR("Hans-Christian Egtvedt "); MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC"); MODULE_LICENSE("GPL"); -- cgit v1.2.3-58-ga151 From f5b2d0ef631bb0647ae8ed1752d2127b8fb6da70 Mon Sep 17 00:00:00 2001 From: Wu Fengguang Date: Wed, 29 Jun 2011 14:26:07 +0800 Subject: ALSA: HDMI - fix ELD monitor name length I noticed that the last character of the ELD monitor name is lost, this fixes the issue. This fix should be confirming to the HDA spec, and works together with the DRM part of the ELD patch. The HDA spec does not mention that Monitor_Name_String is an '\0' ending string, and it allows NML to be 1, which is only valid when MNL does not count the possible ending '\0'. Signed-off-by: Wu Fengguang Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_eld.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_eld.c b/sound/pci/hda/hda_eld.c index b05f7be9dc1b..e3e853153d14 100644 --- a/sound/pci/hda/hda_eld.c +++ b/sound/pci/hda/hda_eld.c @@ -294,7 +294,7 @@ static int hdmi_update_eld(struct hdmi_eld *e, snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl); goto out_fail; } else - strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl); + strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1); for (i = 0; i < e->sad_count; i++) { if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) { -- cgit v1.2.3-58-ga151 From e999dc50404d401150a5429b6459473a691fd1a0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 13 Jun 2011 12:14:07 +0100 Subject: ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values The Blackfin DMA controller can report one frame beyond the end of the buffer in the wraparound case but ALSA requires that the pointer always be in the buffer. Do the wraparound to handle this. A similar bug is likely to apply to the other Blackfin PCM drivers but the code is less obvious to inspection and I don't have a user to test. Reported-by: Kieran O'Leary Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index b5101efd1c87..f1fd95bb6416 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -138,11 +138,20 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) pr_debug("%s enter\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff = sport_curr_offset_tx(sport); - frames = bytes_to_frames(substream->runtime, diff); } else { diff = sport_curr_offset_rx(sport); - frames = bytes_to_frames(substream->runtime, diff); } + + /* + * TX at least can report one frame beyond the end of the + * buffer if we hit the wraparound case - clamp to within the + * buffer as the ALSA APIs require. + */ + if (diff == snd_pcm_lib_buffer_bytes(substream)) + diff = 0; + + frames = bytes_to_frames(substream->runtime, diff); + return frames; } -- cgit v1.2.3-58-ga151 From 71276410e17653cfacfa238a363475cde9e18fb3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2011 12:31:23 +0200 Subject: ALSA: cs5535 - Fix invalid big-endian conversions Fix the wrongly converted short values: sound/pci/cs5535audio/cs5535audio_pcm.c:152: warning: large integer implicitly truncated to unsigned type sound/pci/cs5535audio/cs5535audio_pcm.c:160: warning: large integer implicitly truncated to unsigned type Signed-off-by: Takashi Iwai --- sound/pci/cs5535audio/cs5535audio_pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/pci/cs5535audio/cs5535audio_pcm.c b/sound/pci/cs5535audio/cs5535audio_pcm.c index f16bc8aad6ed..e083122ca55a 100644 --- a/sound/pci/cs5535audio/cs5535audio_pcm.c +++ b/sound/pci/cs5535audio/cs5535audio_pcm.c @@ -149,7 +149,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[i]; desc->addr = cpu_to_le32(addr); desc->size = cpu_to_le32(period_bytes); - desc->ctlreserved = cpu_to_le32(PRD_EOP); + desc->ctlreserved = cpu_to_le16(PRD_EOP); desc_addr += sizeof(struct cs5535audio_dma_desc); addr += period_bytes; } @@ -157,7 +157,7 @@ static int cs5535audio_build_dma_packets(struct cs5535audio *cs5535au, lastdesc = &((struct cs5535audio_dma_desc *) dma->desc_buf.area)[periods]; lastdesc->addr = cpu_to_le32((u32) dma->desc_buf.addr); lastdesc->size = 0; - lastdesc->ctlreserved = cpu_to_le32(PRD_JMP); + lastdesc->ctlreserved = cpu_to_le16(PRD_JMP); jmpprd_addr = cpu_to_le32(lastdesc->addr + (sizeof(struct cs5535audio_dma_desc)*periods)); -- cgit v1.2.3-58-ga151 From 286bed0f0c447b6660e72093d7e778784fdd9ee6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 30 Jun 2011 12:45:36 +0200 Subject: ALSA: hdspm - Fix compile warnings with PPC The char can be unsigned on some architectures. Since the code checks the negative values, they should be declared as signed char explicitly. sound/pci/rme9652/hdspm.c:5449: warning: comparison is always false due to limited range of data type sound/pci/rme9652/hdspm.c:5462: warning: comparison is always false due to limited range of data type Signed-off-by: Takashi Iwai --- sound/pci/rme9652/hdspm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 3f08afc0f0d3..c8e402fc3782 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -896,11 +896,11 @@ struct hdspm { unsigned char max_channels_in; unsigned char max_channels_out; - char *channel_map_in; - char *channel_map_out; + signed char *channel_map_in; + signed char *channel_map_out; - char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs; - char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs; + signed char *channel_map_in_ss, *channel_map_in_ds, *channel_map_in_qs; + signed char *channel_map_out_ss, *channel_map_out_ds, *channel_map_out_qs; char **port_names_in; char **port_names_out; -- cgit v1.2.3-58-ga151 From 713d1369789f2a2336c3431b15276c968862bdb7 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 1 Jul 2011 13:56:13 -0600 Subject: ASoC: Tegra: I2S: Ensure clock is enabled when writing regs The I2S controller needs a clock to respond to register writes. Without this, register writes will at worst hang the CPU. In practice, I've only observed writes being dropped. Luckily, the dropped register writes historically had no effect: TEGRA_I2S_TIMING: The value we wrote was the reset default. TEGRA_I2S_FIFO_SCR: The default was for the FIFOs to request more data when one slot was empty. The requested value was for the FIFOs to request when four slots were empty. The DMA controller in the mainline kernel is configured to burst a single entry at a time into the FIFO, hence there was no issue. The only negative effect was on bus efficiency losses due to an increased number of arbitration attempts. However, in various non-upstream changes, the DMA controller now bursts four entries at a time into the FIFO. If there is only space for one entry, the data is simply dropped. In practice, this resulted in 3/4 of samples being dropped, and playback at 4x the expected rate and pitch. By fixing the clocking issue, this is solved. Signed-off-by: Stephen Warren Acked-by: Liam Girdwood Signed-off-by: Mark Brown --- sound/soc/tegra/tegra_i2s.c | 6 ++++++ 1 file changed, 6 insertions(+) (limited to 'sound') diff --git a/sound/soc/tegra/tegra_i2s.c b/sound/soc/tegra/tegra_i2s.c index 6b817e20548c..95f03c10b4f7 100644 --- a/sound/soc/tegra/tegra_i2s.c +++ b/sound/soc/tegra/tegra_i2s.c @@ -222,12 +222,18 @@ static int tegra_i2s_hw_params(struct snd_pcm_substream *substream, if (i2sclock % (2 * srate)) reg |= TEGRA_I2S_TIMING_NON_SYM_ENABLE; + if (!i2s->clk_refs) + clk_enable(i2s->clk_i2s); + tegra_i2s_write(i2s, TEGRA_I2S_TIMING, reg); tegra_i2s_write(i2s, TEGRA_I2S_FIFO_SCR, TEGRA_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | TEGRA_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); + if (!i2s->clk_refs) + clk_disable(i2s->clk_i2s); + return 0; } -- cgit v1.2.3-58-ga151 From 8e9ddf811ba021506d2316fcfe619faa0ab3f567 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 1 Jul 2011 17:24:46 -0700 Subject: ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting This delay is very conservative. Signed-off-by: Mark Brown Acked-by: Liam Girdwood Cc: stable@kernel.org --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 970a95c5360b..c2fc0356c2a4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1713,6 +1713,8 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, snd_soc_update_bits(codec, WM8994_FLL1_CONTROL_1 + reg_offset, WM8994_FLL1_ENA | WM8994_FLL1_FRAC, reg); + + msleep(5); } wm8994->fll[id].in = freq_in; -- cgit v1.2.3-58-ga151 From 873bd4cb4fbba6a3e99f750e17ef2ba6ef96e9d3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 5 Jul 2011 09:25:59 +0200 Subject: ASoC: Don't set invalid name string to snd_card->driver field The snd_card->driver field contains a driver name string, and in general it shouldn't contain space or special letters. The commit 2b39535b9e54888649923beaab443af212b6c0fd changed the string copy from card->name, but the long name string may contain such letters, thus it may still lead to a segfault. A temporary fix is not to copy the long name string but just keep it empty as the earlier version did. Reported-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Takashi Iwai --- sound/soc/soc-core.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d75043ed7fc0..b194be09e74d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1929,8 +1929,9 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) "%s", card->name); snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->long_name ? card->long_name : card->name); - snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), - "%s", card->driver_name ? card->driver_name : card->name); + if (card->driver_name) + strlcpy(card->snd_card->driver, card->driver_name, + sizeof(card->snd_card->driver)); if (card->late_probe) { ret = card->late_probe(card); -- cgit v1.2.3-58-ga151 From 4c7c5374ce6876d3d2c7013ef815051df4258952 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 4 Jul 2011 10:27:51 -0700 Subject: ASoC: Manage WM8731 ACTIVE bit as a supply widget Now we have supply widgets there's no need to open code the handling of the ACTIVE bit. Signed-off-by: Mark Brown Tested-by: Nicolas Ferre Acked-by: Liam Girdwood --- sound/soc/codecs/wm8731.c | 29 +++-------------------------- 1 file changed, 3 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2dc964b55e4f..76b4361e9b80 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls = SOC_DAPM_ENUM("Input Select", wm8731_insel_enum); static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("ACTIVE",WM8731_ACTIVE, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, &wm8731_output_mixer_controls[0], @@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source, static const struct snd_soc_dapm_route wm8731_intercon[] = { {"DAC", NULL, "OSC", wm8731_check_osc}, {"ADC", NULL, "OSC", wm8731_check_osc}, + {"DAC", NULL, "ACTIVE"}, + {"ADC", NULL, "ACTIVE"}, /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, @@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - /* set active */ - snd_soc_write(codec, WM8731_ACTIVE, 0x0001); - - return 0; -} - -static void wm8731_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - /* deactivate */ - if (!codec->active) { - udelay(50); - snd_soc_write(codec, WM8731_ACTIVE, 0x0); - } -} - static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - snd_soc_write(codec, WM8731_ACTIVE, 0x0); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_ops wm8731_dai_ops = { - .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, -- cgit v1.2.3-58-ga151 From 9c7a083d94656ad6d6f2e03ba90194f2cc5bced5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 7 Jul 2011 09:25:54 +0200 Subject: ALSA: hda - Change all ADCs for dual-adc switching mode for Realtek When the dual-adc switching mode is active in Realtek auto-parser, we need to couple all ADCs as a single capture-volume. Currently, the volume control changes only the first ADC, thus others may remain silent. This patch fixes the problem. Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 33 +++++++++++++++++++++++---------- 1 file changed, 23 insertions(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d21191dcfe88..7d492713c1c1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2715,17 +2715,30 @@ typedef int (*getput_call_t)(struct snd_kcontrol *kcontrol, static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol, - getput_call_t func) + getput_call_t func, bool check_adc_switch) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); - int err; + int i, err; mutex_lock(&codec->control_mutex); - kcontrol->private_value = HDA_COMPOSE_AMP_VAL(spec->adc_nids[adc_idx], - 3, 0, HDA_INPUT); - err = func(kcontrol, ucontrol); + if (check_adc_switch && spec->dual_adc_switch) { + for (i = 0; i < spec->num_adc_nids; i++) { + kcontrol->private_value = + HDA_COMPOSE_AMP_VAL(spec->adc_nids[i], + 3, 0, HDA_INPUT); + err = func(kcontrol, ucontrol); + if (err < 0) + goto error; + } + } else { + i = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + kcontrol->private_value = + HDA_COMPOSE_AMP_VAL(spec->adc_nids[i], + 3, 0, HDA_INPUT); + err = func(kcontrol, ucontrol); + } + error: mutex_unlock(&codec->control_mutex); return err; } @@ -2734,14 +2747,14 @@ static int alc_cap_vol_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_volume_get); + snd_hda_mixer_amp_volume_get, false); } static int alc_cap_vol_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_volume_put); + snd_hda_mixer_amp_volume_put, true); } /* capture mixer elements */ @@ -2751,14 +2764,14 @@ static int alc_cap_sw_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_switch_get); + snd_hda_mixer_amp_switch_get, false); } static int alc_cap_sw_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { return alc_cap_getput_caller(kcontrol, ucontrol, - snd_hda_mixer_amp_switch_put); + snd_hda_mixer_amp_switch_put, true); } #define _DEFINE_CAPMIX(num) \ -- cgit v1.2.3-58-ga151 From bd7fdbcaa2d06d446577fd3c9b81847b04469e01 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 6 Jul 2011 17:58:56 -0700 Subject: ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2 mask didn't cover update-data Signed-off-by: Kuninori Morimoto Acked-by: Liam Girdwood Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/ak4642.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 4be0570e3f1f..65f46047b1cb 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) default: return -EINVAL; } - snd_soc_update_bits(codec, PW_MGMT2, MS, data); + snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); /* format type */ -- cgit v1.2.3-58-ga151 From abaead6ac55dbda8b4bae40251d69dc3f0c49b1c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 9 Jul 2011 11:55:28 +0200 Subject: ALSA: hda - Fix a copmile warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit It's harmless but annyoing. sound/pci/hda/patch_realtek.c: In function ‘alc_cap_getput_caller’: sound/pci/hda/patch_realtek.c:2722:9: warning: ‘err’ may be used uninitialized in this function Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7d492713c1c1..b48fb43b5448 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2719,7 +2719,7 @@ static int alc_cap_getput_caller(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - int i, err; + int i, err = 0; mutex_lock(&codec->control_mutex); if (check_adc_switch && spec->dual_adc_switch) { -- cgit v1.2.3-58-ga151