From 7f62b6ee767586ee7e5d12787dbaaaf47a91979a Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 4 Dec 2013 11:18:36 +0800 Subject: ASoC: soc-pcm: Use valid condition for snd_soc_dai_digital_mute() in hw_free() The snd_soc_dai_digital_mute() here will be never executed because we only decrease codec->active in snd_soc_close(). Thus correct it. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 11a90cd027fa..891b9a9bcbf8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -600,12 +600,13 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* apply codec digital mute */ - if (!codec->active) + if ((playback && codec_dai->playback_active == 1) || + (!playback && codec_dai->capture_active == 1)) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ -- cgit v1.2.3 From a8f1f100ad994725a8295f6997524c57c72e06f5 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 4 Dec 2013 10:37:03 +0800 Subject: ASoC: atmel_ssc_dai: add dai trigger ops According to the SSC specifiation, it should be enabled after DMA is enabled. So, add trigger operation to make sure the right sequence. Signed-off-by: Bo Shen Tested-by: Richard Genoud Signed-off-by: Mark Brown --- sound/soc/atmel/atmel_ssc_dai.c | 30 +++++++++++++++++++++++++++++- 1 file changed, 29 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 8697cedccd21..1ead3c977a51 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -648,7 +648,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; - ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error); pr_debug("%s enabled SSC_SR=0x%08x\n", @@ -657,6 +657,33 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, return 0; } +static int atmel_ssc_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + break; + default: + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + break; + } + + return 0; +} #ifdef CONFIG_PM static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai) @@ -731,6 +758,7 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = { .startup = atmel_ssc_startup, .shutdown = atmel_ssc_shutdown, .prepare = atmel_ssc_prepare, + .trigger = atmel_ssc_trigger, .hw_params = atmel_ssc_hw_params, .set_fmt = atmel_ssc_set_dai_fmt, .set_clkdiv = atmel_ssc_set_dai_clkdiv, -- cgit v1.2.3 From bc567a93502275755492141524935269dcf0ea1b Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 4 Dec 2013 10:37:04 +0800 Subject: ASoC: sam9x5_wm8731: change to work in DSP A mode Change sam9x5 with wm8731 work in DSP A mode, this will fix the left/right channel swap issue. Signed-off-by: Bo Shen Tested-by: Richard Genoud Signed-off-by: Mark Brown --- sound/soc/atmel/sam9x5_wm8731.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 1b372283bd01..7d6a9055874b 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -109,7 +109,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) dai->stream_name = "WM8731 PCM"; dai->codec_dai_name = "wm8731-hifi"; dai->init = sam9x5_wm8731_init; - dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; ret = snd_soc_of_parse_card_name(card, "atmel,model"); -- cgit v1.2.3 From 75704ecfbb4124139b78b71dd603f05d61abe689 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Wed, 4 Dec 2013 17:22:16 +0800 Subject: ASoC: wm8962: Enable SYSCLK provisonally before fetching generated DSPCLK_DIV DSPCLK_DIV can be only generated correctly after enabling SYSCLK. But if the current bias_level hasn't reached SND_SOC_BIAS_ON, DAPM won't enable SYSCLK, which would cause the calculation result from DSPCLK_DIV invalid since bit DSPCLK_DIV will be finally turned to its true value after DAPM enables SYSCLK while the driver won't calculate it again for the current instance. In this circumstance, a playback which needs non-zero DSPCLK_DIV would be distorted due to unexpected clock frequency resulted from an invalid DSPCLK_DIV value. So this patch provisionally enables the SYSCLK to get a valid DSPCLK_DIV for calculation and then disables it afterward. Signed-off-by: Nicolin Chen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 543c5c2631b6..0f17ed3e29f4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8962_CLOCKING_4, WM8962_SYSCLK_RATE_MASK, clocking4); + /* DSPCLK_DIV can be only generated correctly after enabling SYSCLK. + * So we here provisionally enable it and then disable it afterward + * if current bias_level hasn't reached SND_SOC_BIAS_ON. + */ + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA); + dspclk = snd_soc_read(codec, WM8962_CLOCKING1); + + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, 0); + if (dspclk < 0) { dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk); return; -- cgit v1.2.3 From 241bf43321a10815225f477bba96a42285a2da73 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 6 Dec 2013 13:34:50 -0700 Subject: ASoC: tegra: fix uninitialized variables in set_fmt In tegra*_i2s_set_fmt(), in the (fmt == SND_SOC_DAIFMT_CBM_CFM) case, "val" is never assigned to, but left uninitialized. The other case does initialized it. Fix this by initializing val at the start of the function, and only ever ORing into it. Update the handling of "mask" so it works the same way for consistency. Update tegra20_spdif.c to use the same code-style for consistency, even though it doesn't happen to suffer from the same problem at present. Signed-off-by: Stephen Warren Reviewed-by: Thierry Reding Signed-off-by: Mark Brown Fixes: 0f163546a772 ("ASoC: tegra: use regmap more directly") Cc: --- sound/soc/tegra/tegra20_i2s.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 10 +++++----- sound/soc/tegra/tegra30_i2s.c | 6 +++--- 3 files changed, 11 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 364bf6a907e1..8c819f811470 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA20_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA20_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA20_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 08bc6931c7c7..8c7c1028e579 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, { struct device *dev = dai->dev; struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; int ret, spdifclock; - mask = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + mask |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - val = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + val |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; break; default: return -EINVAL; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 231a785b3921..02247fee1cf7 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -118,7 +118,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -127,10 +127,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA30_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA30_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA30_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; -- cgit v1.2.3 From 5dfc03f141993c101c626c84d639019e98a4f39c Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Fri, 6 Dec 2013 23:38:28 +0800 Subject: ASoC: fsl: imx-wm8962: Don't update bias_level in machine driver If we update it here, the set_bias_level() of Codec driver won't be normally called and we will then miss some essential procedures in set_bias_level() of the Codec driver. Thus drop it. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/fsl/imx-wm8962.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 72064e995687..72718e19a3c7 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -130,8 +130,6 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, break; } - dapm->bias_level = level; - return 0; } -- cgit v1.2.3 From 6b9f3e65282b3bd7ed77e7b2b1edfe7cfed48115 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 3 Dec 2013 14:26:33 -0700 Subject: ASoC: don't leak on error in snd_dmaengine_pcm_register If snd_dmaengine_pcm_register()'s call to snd_soc_add_platform() fails, all objects allocated during registration are leaked. Fix this by adding error-handling code. Signed-off-by: Stephen Warren Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 38 +++++++++++++++++++++++++---------- 1 file changed, 27 insertions(+), 11 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index cbc9c96ce1f4..41949af3baae 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -305,6 +305,20 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, } } +static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) +{ + unsigned int i; + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; + i++) { + if (!pcm->chan[i]) + continue; + dma_release_channel(pcm->chan[i]); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; + } +} + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -315,6 +329,7 @@ int snd_dmaengine_pcm_register(struct device *dev, const struct snd_dmaengine_pcm_config *config, unsigned int flags) { struct dmaengine_pcm *pcm; + int ret; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); if (!pcm) @@ -326,11 +341,20 @@ int snd_dmaengine_pcm_register(struct device *dev, dmaengine_pcm_request_chan_of(pcm, dev); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) - return snd_soc_add_platform(dev, &pcm->platform, + ret = snd_soc_add_platform(dev, &pcm->platform, &dmaengine_no_residue_pcm_platform); else - return snd_soc_add_platform(dev, &pcm->platform, + ret = snd_soc_add_platform(dev, &pcm->platform, &dmaengine_pcm_platform); + if (ret) + goto err_free_dma; + + return 0; + +err_free_dma: + dmaengine_pcm_release_chan(pcm); + kfree(pcm); + return ret; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register); @@ -345,7 +369,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev) { struct snd_soc_platform *platform; struct dmaengine_pcm *pcm; - unsigned int i; platform = snd_soc_lookup_platform(dev); if (!platform) @@ -353,15 +376,8 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_platform_to_pcm(platform); - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - if (pcm->chan[i]) { - dma_release_channel(pcm->chan[i]); - if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) - break; - } - } - snd_soc_remove_platform(platform); + dmaengine_pcm_release_chan(pcm); kfree(pcm); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister); -- cgit v1.2.3 From 693e0cb052c607e2d41edf9e9f1fa99ff8c266c1 Mon Sep 17 00:00:00 2001 From: David Henningsson Date: Thu, 12 Dec 2013 09:52:03 +0100 Subject: ALSA: hda - Add enable_msi=0 workaround for four HP machines While enabling these machines, we found we would sometimes lose an interrupt if we change hardware volume during playback, and that disabling msi fixed this issue. (Losing the interrupt caused underruns and crackling audio, as the one second timeout is usually bigger than the period size.) The machines were all machines from HP, running AMD Hudson controller, and Realtek ALC282 codec. Cc: stable@vger.kernel.org BugLink: https://bugs.launchpad.net/bugs/1260225 Signed-off-by: David Henningsson Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_intel.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 27aa14007cbd..956871d8b3d2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3433,6 +3433,10 @@ static void check_probe_mask(struct azx *chip, int dev) * white/black-list for enable_msi */ static struct snd_pci_quirk msi_black_list[] = { + SND_PCI_QUIRK(0x103c, 0x2191, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x2192, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x21f7, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x21fa, "HP", 0), /* AMD Hudson */ SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ -- cgit v1.2.3 From c29cb5eb8157a0049c882672a7f941261f23ea34 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Fri, 13 Dec 2013 11:57:05 +0800 Subject: ALSA: hda - Add Dell headset detection quirk for three laptop models On the Dell machines with codec whose Subsystem Id is 0x10280610, 0x10280629 or 0x1028063e, no external microphone can be detected when plugging a 3-ring headset. If we add "model=dell-headset-multi" for the snd-hda-intel.ko, the problem will disappear. The codecs on these machines belong to alc_269 family. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson Cc: stable@vger.kernel.org Signed-off-by: Hui Wang Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 34de5dc2fe9b..5ab8e1631190 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4247,11 +4247,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- cgit v1.2.3 From 280484e708a3cc38fe6807718caa460e744c0b20 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 17 Dec 2013 13:16:16 +0000 Subject: ASoC: wm5110: Correct HPOUT3 DAPM route typo Reported-by: Kyung-Kwee Ryu Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm5110.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c3c7396a6181..b3d0b284cca1 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1037,7 +1037,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AEC Loopback", "HPOUT3L", "OUT3L" }, { "AEC Loopback", "HPOUT3R", "OUT3R" }, { "HPOUT3L", NULL, "OUT3L" }, - { "HPOUT3R", NULL, "OUT3L" }, + { "HPOUT3R", NULL, "OUT3R" }, { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, -- cgit v1.2.3 From 02fc17c10258ad70c1b9a93f8884bdaf0ac3f766 Mon Sep 17 00:00:00 2001 From: Jean-Francois Moine Date: Wed, 27 Nov 2013 21:10:24 +0100 Subject: ASoC: kirkwood: Fix the CPU DAI rates This patch fixes the rates declared in the CPU DAI parameters: - SNDRV_PCM_RATE_KNOT and the discrete rates SNDRV_PCM_RATE_xxx should not be used with SNDRV_PCM_RATE_CONTINUOUS, - SNDRV_PCM_RATE_CONTINUOUS asks for rate_min and rate_max, - the device may do streaming down to 5512Hz. Signed-off-by: Jean-Francois Moine Signed-off-by: Mark Brown --- sound/soc/kirkwood/kirkwood-i2s.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index 0b18f654b413..3920a5e8125f 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -473,17 +473,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_I2S_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, @@ -494,17 +494,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, -- cgit v1.2.3 From ed697e1aaf7237b1a62af39f64463b05c262808d Mon Sep 17 00:00:00 2001 From: JongHo Kim Date: Tue, 17 Dec 2013 23:02:24 +0900 Subject: ALSA: Add SNDRV_PCM_STATE_PAUSED case in wait_for_avail function When the process is sleeping at the SNDRV_PCM_STATE_PAUSED state from the wait_for_avail function, the sleep process will be woken by timeout(10 seconds). Even if the sleep process wake up by timeout, by this patch, the process will continue with sleep and wait for the other state. Signed-off-by: JongHo Kim Cc: Signed-off-by: Takashi Iwai --- sound/core/pcm_lib.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6e03b465e44e..a2104671f51d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1937,6 +1937,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, case SNDRV_PCM_STATE_DISCONNECTED: err = -EBADFD; goto _endloop; + case SNDRV_PCM_STATE_PAUSED: + continue; } if (!tout) { snd_printd("%s write error (DMA or IRQ trouble?)\n", -- cgit v1.2.3 From 939fd1e8d9deff206f12bd9d4e54aa7a4bd0ffd6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 18 Dec 2013 09:25:49 +0000 Subject: ASoC: wm_adsp: Add small delay while polling DSP RAM start Some devices are getting very close to the limit whilst polling the RAM start, this patch adds a small delay to this loop to give a longer startup timeout. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm_adsp.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 46ec0e9744d4..4fbcab63e61f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1474,13 +1474,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return ret; /* Wait for the RAM to start, should be near instantaneous */ - count = 0; - do { + for (count = 0; count < 10; ++count) { ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1, &val); if (ret != 0) return ret; - } while (!(val & ADSP2_RAM_RDY) && ++count < 10); + + if (val & ADSP2_RAM_RDY) + break; + + msleep(1); + } if (!(val & ADSP2_RAM_RDY)) { adsp_err(dsp, "Failed to start DSP RAM\n"); -- cgit v1.2.3 From f0199bc5e3a3ec13f9bc938556517ec430b36437 Mon Sep 17 00:00:00 2001 From: Bo Shen Date: Wed, 18 Dec 2013 11:26:23 +0800 Subject: ASoC: wm8904: fix DSP mode B configuration When wm8904 work in DSP mode B, we still need to configure it to work in DSP mode. Or else, it will work in Right Justified mode. Signed-off-by: Bo Shen Acked-by: Charles Keepax Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8904.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3938fb1c203e..53bbfac6a83a 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1444,7 +1444,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: - aif1 |= WM8904_AIF_LRCLK_INV; + aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; -- cgit v1.2.3 From 3a6c5d8ad0a9253aafb76df3577edcb68c09b939 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 18 Dec 2013 18:09:56 +0800 Subject: ALSA: hda - Add Dell headset detection quirk for one more laptop model On the Dell machines with codec whose Subsystem Id is 0x10280640, no external microphone can be detected when plugging a 3-ring headset. Using ALC255_FIXUP_DELL1_MIC_NO_PRESENCE can fix this problem. The codec (Vendor ID: 0x10ec0255) on the machine belongs to alc_269 family. BugLink: https://bugs.launchpad.net/bugs/1260303 Cc: David Henningsson Cc: stable@vger.kernel.org Signed-off-by: Hui Wang 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 5ab8e1631190..c5646941539a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4256,6 +4256,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- cgit v1.2.3