From 016fcab8ff46fca29375d484226ec91932aa4a07 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 4 Jul 2013 20:01:02 -0300 Subject: ASoC: sglt5000: Fix the default value of CHIP_SSS_CTRL According to the sgtl5000 reference manual, the default value of CHIP_SSS_CTRL is 0x10. Reported-by: Oskar Schirmer Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d441559dc92..d659d3adcfb 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -38,7 +38,7 @@ static const struct reg_default sgtl5000_reg_defaults[] = { { SGTL5000_CHIP_CLK_CTRL, 0x0008 }, { SGTL5000_CHIP_I2S_CTRL, 0x0010 }, - { SGTL5000_CHIP_SSS_CTRL, 0x0008 }, + { SGTL5000_CHIP_SSS_CTRL, 0x0010 }, { SGTL5000_CHIP_DAC_VOL, 0x3c3c }, { SGTL5000_CHIP_PAD_STRENGTH, 0x015f }, { SGTL5000_CHIP_ANA_HP_CTRL, 0x1818 }, -- cgit v1.2.3-70-g09d2 From 5c78dfe87ea04b501ee000a7f03b9432ac9d008c Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 4 Jul 2013 20:01:03 -0300 Subject: ASoC: sglt5000: Fix SGTL5000_PLL_FRAC_DIV_MASK SGTL5000_PLL_FRAC_DIV_MASK is used to mask bits 0-10 (11 bits in total) of register CHIP_PLL_CTRL, so fix the mask to accomodate all this bit range. Reported-by: Oskar Schirmer Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/sgtl5000.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.h b/sound/soc/codecs/sgtl5000.h index 4b69229a981..2f8c88931f6 100644 --- a/sound/soc/codecs/sgtl5000.h +++ b/sound/soc/codecs/sgtl5000.h @@ -347,7 +347,7 @@ #define SGTL5000_PLL_INT_DIV_MASK 0xf800 #define SGTL5000_PLL_INT_DIV_SHIFT 11 #define SGTL5000_PLL_INT_DIV_WIDTH 5 -#define SGTL5000_PLL_FRAC_DIV_MASK 0x0700 +#define SGTL5000_PLL_FRAC_DIV_MASK 0x07ff #define SGTL5000_PLL_FRAC_DIV_SHIFT 0 #define SGTL5000_PLL_FRAC_DIV_WIDTH 11 -- cgit v1.2.3-70-g09d2 From 82e414fa1dbbc07e7b6d582e4fbcc9b0a5299f7f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 5 Jul 2013 12:36:24 +0100 Subject: ASoC: wm8994: Remove overly noisy debug logging This was committed in error. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 4 ---- 1 file changed, 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 25580b5a853..1b89aa9029e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3856,8 +3856,6 @@ static void wm8958_mic_work(struct work_struct *work) mic_complete_work.work); struct snd_soc_codec *codec = wm8994->hubs.codec; - dev_crit(codec->dev, "MIC WORK %x\n", wm8994->mic_status); - pm_runtime_get_sync(codec->dev); mutex_lock(&wm8994->accdet_lock); @@ -3867,8 +3865,6 @@ static void wm8958_mic_work(struct work_struct *work) mutex_unlock(&wm8994->accdet_lock); pm_runtime_put(codec->dev); - - dev_crit(codec->dev, "MIC WORK %x DONE\n", wm8994->mic_status); } static irqreturn_t wm8958_mic_irq(int irq, void *data) -- cgit v1.2.3-70-g09d2 From 770100108be7dbe614361dbcc450096b4cdfc98b Mon Sep 17 00:00:00 2001 From: Padmavathi Venna Date: Thu, 11 Jul 2013 12:38:25 +0530 Subject: ASoC: Samsung: Set RFS and BFS in slave mode As per the User Manual, the RFS and BFS should be set in slave mode for correct operation. Signed-off-by: Padmavathi Venna Signed-off-by: Andrew Bresticker Reviewed-by: Simon Glass Signed-off-by: Mark Brown --- sound/soc/samsung/i2s.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 7a173469743..959c702235c 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -742,13 +742,13 @@ static int config_setup(struct i2s_dai *i2s) return -EAGAIN; } - /* Don't bother RFS, BFS & PSR in Slave mode */ - if (is_slave(i2s)) - return 0; - set_bfs(i2s, bfs); set_rfs(i2s, rfs); + /* Don't bother with PSR in Slave mode */ + if (is_slave(i2s)) + return 0; + if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { psr = i2s->rclk_srcrate / i2s->frmclk / rfs; writel(((psr - 1) << 8) | PSR_PSREN, i2s->addr + I2SPSR); -- cgit v1.2.3-70-g09d2 From f6becf0b2ffef0bed813d7f910b5d276c5dc45e1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 11 Jul 2013 14:35:43 +0200 Subject: ASoC: omap-pcm: Request the DMA channel differently when DT is involved When booting with DT the platform_get_resource_byname() is not available to get the DMA resource. In this case the DAI drivers will set the filter_data to the name of the DMA and omap-pcm can use this to request the DMA channel. Signed-off-by: Peter Ujfalusi Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/omap-pcm.c | 17 ++++++++++++++--- 1 file changed, 14 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index c28e042f220..a11405de86e 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -113,14 +113,25 @@ static int omap_pcm_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_dmaengine_dai_dma_data *dma_data; + int ret; snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware); dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - return snd_dmaengine_pcm_open_request_chan(substream, - omap_dma_filter_fn, - dma_data->filter_data); + /* DT boot: filter_data is the DMA name */ + if (rtd->cpu_dai->dev->of_node) { + struct dma_chan *chan; + + chan = dma_request_slave_channel(rtd->cpu_dai->dev, + dma_data->filter_data); + ret = snd_dmaengine_pcm_open(substream, chan); + } else { + ret = snd_dmaengine_pcm_open_request_chan(substream, + omap_dma_filter_fn, + dma_data->filter_data); + } + return ret; } static int omap_pcm_mmap(struct snd_pcm_substream *substream, -- cgit v1.2.3-70-g09d2 From a8035f073cb508a0c1223db2662510575627b41d Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 11 Jul 2013 14:35:44 +0200 Subject: ASoC: omap-mcpdm: Do not use platform_get_resource_byname() for DMA The DMA resource no longer available via this API when booting with DT. McPDM is only available on OMAP4/5 and both can boot with DT only. Set the dma_data.filter_data to the DMA name which will be used by omap-pcm to request the DMA channel. Signed-off-by: Peter Ujfalusi Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 16 ++-------------- 1 file changed, 2 insertions(+), 14 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index eb05c7ed6d0..a49dc52f8ab 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -66,7 +66,6 @@ struct omap_mcpdm { bool restart; struct snd_dmaengine_dai_dma_data dma_data[2]; - unsigned int dma_req[2]; }; /* @@ -477,19 +476,8 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dma_data[0].addr = res->start + MCPDM_REG_DN_DATA; mcpdm->dma_data[1].addr = res->start + MCPDM_REG_UP_DATA; - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "dn_link"); - if (!res) - return -ENODEV; - - mcpdm->dma_req[0] = res->start; - mcpdm->dma_data[0].filter_data = &mcpdm->dma_req[0]; - - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "up_link"); - if (!res) - return -ENODEV; - - mcpdm->dma_req[1] = res->start; - mcpdm->dma_data[1].filter_data = &mcpdm->dma_req[1]; + mcpdm->dma_data[0].filter_data = "dn_link"; + mcpdm->dma_data[1].filter_data = "up_link"; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (res == NULL) -- cgit v1.2.3-70-g09d2 From 2ebef44789223389708505e33c67d44e9f999d4a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 11 Jul 2013 14:35:45 +0200 Subject: ASoC: omap-dmic: Do not use platform_get_resource_byname() for DMA The DMA resource no longer available via this API when booting with DT. DMIC is only available on OMAP4/5 and both can boot with DT only. Set the dma_data.filter_data to the DMA name which will be used by omap-pcm to request the DMA channel. Signed-off-by: Peter Ujfalusi Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/omap-dmic.c | 11 +---------- 1 file changed, 1 insertion(+), 10 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/omap-dmic.c b/sound/soc/omap/omap-dmic.c index 2ad0370146f..4db1f8e6e17 100644 --- a/sound/soc/omap/omap-dmic.c +++ b/sound/soc/omap/omap-dmic.c @@ -57,7 +57,6 @@ struct omap_dmic { struct mutex mutex; struct snd_dmaengine_dai_dma_data dma_data; - unsigned int dma_req; }; static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val) @@ -478,15 +477,7 @@ static int asoc_dmic_probe(struct platform_device *pdev) } dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG; - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(dmic->dev, "invalid dma resource\n"); - ret = -ENODEV; - goto err_put_clk; - } - - dmic->dma_req = res->start; - dmic->dma_data.filter_data = &dmic->dma_req; + dmic->dma_data.filter_data = "up_link"; res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu"); if (!res) { -- cgit v1.2.3-70-g09d2 From 9ab1fac4829b3da0ba4d3f44d95d3e8ad13e6629 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 11 Jul 2013 14:35:46 +0200 Subject: ASoC: omap-mcbsp: Use different method for DMA request when booted with DT The DMA resource no longer available via this API when booting with DT. When the board is booted with DT do not use platform_get_resource_byname(), instead set the dma_data.filter_data to the name of the DMA channel and omap-pcm can use this name to request the DMA channel. Signed-off-by: Peter Ujfalusi Reviewed-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/omap/mcbsp.c | 39 ++++++++++++++++++++++----------------- 1 file changed, 22 insertions(+), 17 deletions(-) (limited to 'sound') diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index eb68c7db1cf..361e4c03646 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -1012,28 +1012,33 @@ int omap_mcbsp_init(struct platform_device *pdev) } } - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); - if (!res) { - dev_err(&pdev->dev, "invalid rx DMA channel\n"); - return -ENODEV; - } - /* RX DMA request number, and port address configuration */ - mcbsp->dma_req[1] = res->start; - mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; - mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, 1); - mcbsp->dma_data[1].maxburst = 4; + if (!pdev->dev.of_node) { + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); + if (!res) { + dev_err(&pdev->dev, "invalid tx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[0] = res->start; + mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; - res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx"); - if (!res) { - dev_err(&pdev->dev, "invalid tx DMA channel\n"); - return -ENODEV; + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx"); + if (!res) { + dev_err(&pdev->dev, "invalid rx DMA channel\n"); + return -ENODEV; + } + mcbsp->dma_req[1] = res->start; + mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1]; + } else { + mcbsp->dma_data[0].filter_data = "tx"; + mcbsp->dma_data[1].filter_data = "rx"; } - /* TX DMA request number, and port address configuration */ - mcbsp->dma_req[0] = res->start; - mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0]; + mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp, 0); mcbsp->dma_data[0].maxburst = 4; + mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp, 1); + mcbsp->dma_data[1].maxburst = 4; + mcbsp->fclk = clk_get(&pdev->dev, "fck"); if (IS_ERR(mcbsp->fclk)) { ret = PTR_ERR(mcbsp->fclk); -- cgit v1.2.3-70-g09d2 From 5f17482a3244c07646279d16c0e5b8c0b2b76d0e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 11 Jul 2013 20:09:43 -0700 Subject: ASoC: wm8978: enable symmetric rates wm8978 needs .symmetric_rates = 1. The playback/capture will be strange without this patch when it used asymmetric rate in same time Tested-by: Yusuke Goda Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index 029f31c8e70..d8fc531c0e5 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -921,6 +921,7 @@ static struct snd_soc_dai_driver wm8978_dai = { .formats = WM8978_FORMATS, }, .ops = &wm8978_dai_ops, + .symmetric_rates = 1, }; static int wm8978_suspend(struct snd_soc_codec *codec) -- cgit v1.2.3-70-g09d2 From 8331b9e332a6e72d5285b05f56a7b66b692cb67a Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Jul 2013 11:58:55 +0200 Subject: sound: oss/vwsnd: Add missing inclusion of linux/delay.h Reported-by: Fengguang Wu Signed-off-by: Takashi Iwai --- sound/oss/vwsnd.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index 7e814a5c367..d8db9023bc5 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -149,6 +149,7 @@ #include #include #include +#include #include -- cgit v1.2.3-70-g09d2 From 4b8846062faac4e5c3f08e2e06bbb33c949aa51f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 15 Jul 2013 12:00:24 +0200 Subject: sound: oss/vwsnd: Always define vwsnd_mutex While the conversion of BKL to mutex in commit 645ef9ef, the mutex definition was put in a wrong place inside #ifdef WSND_DEBUG, which leads to the build error. Just move it outside the ifdef. Reported-by: Fengguang Wu Signed-off-by: Takashi Iwai --- sound/oss/vwsnd.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/oss/vwsnd.c b/sound/oss/vwsnd.c index d8db9023bc5..4bbcc0fcd4e 100644 --- a/sound/oss/vwsnd.c +++ b/sound/oss/vwsnd.c @@ -155,12 +155,13 @@ #include "sound_config.h" +static DEFINE_MUTEX(vwsnd_mutex); + /*****************************************************************************/ /* debug stuff */ #ifdef VWSND_DEBUG -static DEFINE_MUTEX(vwsnd_mutex); static int shut_up = 1; /* -- cgit v1.2.3-70-g09d2 From 60478295d6876619f8f47f6d1a5c25eaade69ee3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:55:57 +0200 Subject: ALSA: asihpi: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Signed-off-by: Takashi Iwai --- sound/pci/asihpi/asihpi.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index 185d54a5cb1..dc632cdc387 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -769,7 +769,10 @@ static void snd_card_asihpi_timer_function(unsigned long data) s->number); ds->drained_count++; if (ds->drained_count > 20) { + unsigned long flags; + snd_pcm_stream_lock_irqsave(s, flags); snd_pcm_stop(s, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(s, flags); continue; } } else { -- cgit v1.2.3-70-g09d2 From cc7282b8d5abbd48c81d1465925d464d9e3eaa8f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:56:56 +0200 Subject: ALSA: atiixp: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Signed-off-by: Takashi Iwai --- sound/pci/atiixp.c | 2 ++ sound/pci/atiixp_modem.c | 2 ++ 2 files changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/pci/atiixp.c b/sound/pci/atiixp.c index fe4c61bdb8b..f6dec3ea371 100644 --- a/sound/pci/atiixp.c +++ b/sound/pci/atiixp.c @@ -689,7 +689,9 @@ static void snd_atiixp_xrun_dma(struct atiixp *chip, struct atiixp_dma *dma) if (! dma->substream || ! dma->running) return; snd_printdd("atiixp: XRUN detected (DMA %d)\n", dma->ops->type); + snd_pcm_stream_lock(dma->substream); snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(dma->substream); } /* diff --git a/sound/pci/atiixp_modem.c b/sound/pci/atiixp_modem.c index cf29b9a1d65..289563ecb6d 100644 --- a/sound/pci/atiixp_modem.c +++ b/sound/pci/atiixp_modem.c @@ -638,7 +638,9 @@ static void snd_atiixp_xrun_dma(struct atiixp_modem *chip, if (! dma->substream || ! dma->running) return; snd_printdd("atiixp-modem: XRUN detected (DMA %d)\n", dma->ops->type); + snd_pcm_stream_lock(dma->substream); snd_pcm_stop(dma->substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(dma->substream); } /* -- cgit v1.2.3-70-g09d2 From 5b9ab3f7324a1b94a5a5a76d44cf92dfeb3b5e80 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:57:55 +0200 Subject: ALSA: 6fire: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index c5b9cac37dc..2aa4e13063a 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -639,17 +639,25 @@ int usb6fire_pcm_init(struct sfire_chip *chip) void usb6fire_pcm_abort(struct sfire_chip *chip) { struct pcm_runtime *rt = chip->pcm; + unsigned long flags; int i; if (rt) { rt->panic = true; - if (rt->playback.instance) + if (rt->playback.instance) { + snd_pcm_stream_lock_irqsave(rt->playback.instance, flags); snd_pcm_stop(rt->playback.instance, SNDRV_PCM_STATE_XRUN); - if (rt->capture.instance) + snd_pcm_stream_unlock_irqrestore(rt->playback.instance, flags); + } + + if (rt->capture.instance) { + snd_pcm_stream_lock_irqsave(rt->capture.instance, flags); snd_pcm_stop(rt->capture.instance, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(rt->capture.instance, flags); + } for (i = 0; i < PCM_N_URBS; i++) { usb_poison_urb(&rt->in_urbs[i].instance); -- cgit v1.2.3-70-g09d2 From cb6f66a2d278e57a6c9d8fb59bd9ebd8ab3965c2 Mon Sep 17 00:00:00 2001 From: Chih-Chung Chang Date: Mon, 15 Jul 2013 09:38:46 -0700 Subject: ASoC: max98088 - fix element type of the register cache. The registers of max98088 are 8 bits, not 16 bits. This bug causes the contents of registers to be overwritten with bad values when the codec is suspended and then resumed. Signed-off-by: Chih-Chung Chang Signed-off-by: Dylan Reid Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/max98088.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 3eeada57e87..566a367c94f 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1612,7 +1612,7 @@ static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int mute) static void max98088_sync_cache(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; + u8 *reg_cache = codec->reg_cache; int i; if (!codec->cache_sync) -- cgit v1.2.3-70-g09d2 From 9538aa46c2427d6782aa10036c4da4c541605e0e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:58:25 +0200 Subject: ALSA: ua101: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Acked-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/misc/ua101.c | 14 ++++++++++++-- 1 file changed, 12 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/misc/ua101.c b/sound/usb/misc/ua101.c index 8b5d2c564e0..509315937f2 100644 --- a/sound/usb/misc/ua101.c +++ b/sound/usb/misc/ua101.c @@ -613,14 +613,24 @@ static int start_usb_playback(struct ua101 *ua) static void abort_alsa_capture(struct ua101 *ua) { - if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) + unsigned long flags; + + if (test_bit(ALSA_CAPTURE_RUNNING, &ua->states)) { + snd_pcm_stream_lock_irqsave(ua->capture.substream, flags); snd_pcm_stop(ua->capture.substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(ua->capture.substream, flags); + } } static void abort_alsa_playback(struct ua101 *ua) { - if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) + unsigned long flags; + + if (test_bit(ALSA_PLAYBACK_RUNNING, &ua->states)) { + snd_pcm_stream_lock_irqsave(ua->playback.substream, flags); snd_pcm_stop(ua->playback.substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(ua->playback.substream, flags); + } } static int set_stream_hw(struct ua101 *ua, struct snd_pcm_substream *substream, -- cgit v1.2.3-70-g09d2 From 5be1efb4c2ed79c3d7c0cbcbecae768377666e84 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:58:47 +0200 Subject: ALSA: usx2y: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Signed-off-by: Takashi Iwai --- sound/usb/usx2y/usbusx2yaudio.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound') diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 4967fe9c938..63fb5219f0f 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -273,7 +273,11 @@ static void usX2Y_clients_stop(struct usX2Ydev *usX2Y) struct snd_usX2Y_substream *subs = usX2Y->subs[s]; if (subs) { if (atomic_read(&subs->state) >= state_PRERUNNING) { + unsigned long flags; + + snd_pcm_stream_lock_irqsave(subs->pcm_substream, flags); snd_pcm_stop(subs->pcm_substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock_irqrestore(subs->pcm_substream, flags); } for (u = 0; u < NRURBS; u++) { struct urb *urb = subs->urb[u]; -- cgit v1.2.3-70-g09d2 From 46f6c1aaf790be9ea3c8ddfc8f235a5f677d08e2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 17:59:33 +0200 Subject: ALSA: pxa2xx: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/arm/pxa2xx-pcm-lib.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 76e0d569507..823359ed95e 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -166,7 +166,9 @@ void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id) } else { printk(KERN_ERR "%s: DMA error on channel %d (DCSR=%#x)\n", rtd->params->name, dma_ch, dcsr); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); } } EXPORT_SYMBOL(pxa2xx_pcm_dma_irq); -- cgit v1.2.3-70-g09d2 From 571185717f8d7f2a088a7ac38d94a9ad5fd9da5c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 18:00:01 +0200 Subject: ASoC: atmel: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/atmel/atmel-pcm-dma.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/atmel/atmel-pcm-dma.c b/sound/soc/atmel/atmel-pcm-dma.c index 1d38fd0bc4e..d1282652679 100644 --- a/sound/soc/atmel/atmel-pcm-dma.c +++ b/sound/soc/atmel/atmel-pcm-dma.c @@ -81,7 +81,9 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, /* stop RX and capture: will be enabled again at restart */ ssc_writex(prtd->ssc->regs, SSC_CR, prtd->mask->ssc_disable); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); /* now drain RHR and read status to remove xrun condition */ ssc_readx(prtd->ssc->regs, SSC_RHR); -- cgit v1.2.3-70-g09d2 From 61be2b9a18ec70f3cbe3deef7a5f77869c71b5ae Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 11 Jul 2013 18:00:25 +0200 Subject: ASoC: s6000: Fix unlocked snd_pcm_stop() call snd_pcm_stop() must be called in the PCM substream lock context. Cc: Acked-by: Mark Brown Signed-off-by: Takashi Iwai --- sound/soc/s6000/s6000-pcm.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index 1358c7de252..d0740a76296 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -128,7 +128,9 @@ static irqreturn_t s6000_pcm_irq(int irq, void *data) substream->runtime && snd_pcm_running(substream)) { dev_dbg(pcm->dev, "xrun\n"); + snd_pcm_stream_lock(substream); snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + snd_pcm_stream_unlock(substream); ret = IRQ_HANDLED; } -- cgit v1.2.3-70-g09d2 From d52392b1a80458c0510810789c7db4a39b88022a Mon Sep 17 00:00:00 2001 From: Aaron Plattner Date: Fri, 12 Jul 2013 11:01:37 -0700 Subject: ALSA: hda - Add new GPU codec ID to snd-hda Vendor ID 0x10de0060 is used by a yet-to-be-named GPU chip. Reviewed-by: Andy Ritger Signed-off-by: Aaron Plattner Cc: Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 540bdef2f90..030ca8652a1 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -2622,6 +2622,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_generic_hdmi }, +{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_generic_hdmi }, { .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch }, { .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi }, @@ -2674,6 +2675,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0042"); MODULE_ALIAS("snd-hda-codec-id:10de0043"); MODULE_ALIAS("snd-hda-codec-id:10de0044"); MODULE_ALIAS("snd-hda-codec-id:10de0051"); +MODULE_ALIAS("snd-hda-codec-id:10de0060"); MODULE_ALIAS("snd-hda-codec-id:10de0067"); MODULE_ALIAS("snd-hda-codec-id:10de8001"); MODULE_ALIAS("snd-hda-codec-id:11069f80"); -- cgit v1.2.3-70-g09d2 From 46a5905e1cd4a9d9d238ec7beece49ce49e2ad85 Mon Sep 17 00:00:00 2001 From: Shawn Guo Date: Tue, 16 Jul 2013 09:17:27 +0800 Subject: ASoC: sgtl5000: defer the probe if clock is not found It's not always the case that clock is already available when sgtl5000 get probed at the first time, e.g. the clock is provided by CPU DAI which may be probed after sgtl5000. So let's defer the probe when devm_clk_get() call fails and give it chance to try later. It fixes the regression on imx28 since commit 9e13f34 (ASoC: sgtl5000: Let the codec acquire its clock). [ 1.927637] sgtl5000 0-000a: Failed to get mclock: -2 [ 1.934280] sgtl5000: probe of 0-000a failed with error -2 [ 1.945906] mxs-sgtl5000 sound.13: ASoC: CODEC (null) not registered [ 1.953787] mxs-sgtl5000 sound.13: snd_soc_register_card failed (-517) [ 1.960865] platform sound.13: Driver mxs-sgtl5000 requests probe deferral Signed-off-by: Shawn Guo Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index d659d3adcfb..6c8a9e7bee2 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1527,6 +1527,9 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, if (IS_ERR(sgtl5000->mclk)) { ret = PTR_ERR(sgtl5000->mclk); dev_err(&client->dev, "Failed to get mclock: %d\n", ret); + /* Defer the probe to see if the clk will be provided later */ + if (ret == -ENOENT) + return -EPROBE_DEFER; return ret; } -- cgit v1.2.3-70-g09d2 From 256ca9c3ad5013ff8a8f165e5a82fab437628c8e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 16 Jul 2013 12:17:49 +0200 Subject: ALSA: seq-oss: Initialize MIDI clients asynchronously We've got bug reports that the module loading stuck on Debian system with 3.10 kernel. The debugging session revealed that the initial registration of OSS sequencer clients stuck at module loading time, which involves again with request_module() at the init phase. This is triggered only by special --install stuff Debian is using, but it's still not good to have such loops. As a workaround, call the registration part asynchronously. This is a better approach irrespective of the hang fix, in anyway. Reported-and-tested-by: Philipp Matthias Hahn Cc: Signed-off-by: Takashi Iwai --- sound/core/seq/oss/seq_oss_init.c | 16 +++++++++++++--- sound/core/seq/oss/seq_oss_midi.c | 2 +- 2 files changed, 14 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/core/seq/oss/seq_oss_init.c b/sound/core/seq/oss/seq_oss_init.c index e3cb46fef2c..b3f39b5ed74 100644 --- a/sound/core/seq/oss/seq_oss_init.c +++ b/sound/core/seq/oss/seq_oss_init.c @@ -31,6 +31,7 @@ #include #include #include +#include /* * common variables @@ -60,6 +61,14 @@ static void free_devinfo(void *private); #define call_ctl(type,rec) snd_seq_kernel_client_ctl(system_client, type, rec) +/* call snd_seq_oss_midi_lookup_ports() asynchronously */ +static void async_call_lookup_ports(struct work_struct *work) +{ + snd_seq_oss_midi_lookup_ports(system_client); +} + +static DECLARE_WORK(async_lookup_work, async_call_lookup_ports); + /* * create sequencer client for OSS sequencer */ @@ -85,9 +94,6 @@ snd_seq_oss_create_client(void) system_client = rc; debug_printk(("new client = %d\n", rc)); - /* look up midi devices */ - snd_seq_oss_midi_lookup_ports(system_client); - /* create annoucement receiver port */ memset(port, 0, sizeof(*port)); strcpy(port->name, "Receiver"); @@ -115,6 +121,9 @@ snd_seq_oss_create_client(void) } rc = 0; + /* look up midi devices */ + schedule_work(&async_lookup_work); + __error: kfree(port); return rc; @@ -160,6 +169,7 @@ receive_announce(struct snd_seq_event *ev, int direct, void *private, int atomic int snd_seq_oss_delete_client(void) { + cancel_work_sync(&async_lookup_work); if (system_client >= 0) snd_seq_delete_kernel_client(system_client); diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 677dc84590c..862d84893ee 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -72,7 +72,7 @@ static int send_midi_event(struct seq_oss_devinfo *dp, struct snd_seq_event *ev, * look up the existing ports * this looks a very exhausting job. */ -int __init +int snd_seq_oss_midi_lookup_ports(int client) { struct snd_seq_client_info *clinfo; -- cgit v1.2.3-70-g09d2 From 1ea9a69d1a36a5b62bf281ba8bb304fcac656dad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Jul 2013 07:58:02 +0200 Subject: ALSA: hda - Fix EAPD GPIO control for Sigmatel codecs The EAPD GPIO is dynamically turned on/off for some machines with Sigmatel codecs, but this didn't work as expected, and it resulted in spontaneous lost of speaker outputs per HP plugging or power-saving. This patch fixes the bug by simply including spec->eapd_mask into spec->gpio_mask and spec->gpio_data bits. Reported-and-tested-by: Eric Shattow Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index e2f83591161..766e56754c6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -417,9 +417,11 @@ static void stac_update_outputs(struct hda_codec *codec) val &= ~spec->eapd_mask; else val |= spec->eapd_mask; - if (spec->gpio_data != val) + if (spec->gpio_data != val) { + spec->gpio_data = val; stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, val); + } } } @@ -3612,20 +3614,18 @@ static int stac_parse_auto_config(struct hda_codec *codec) static int stac_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; - unsigned int gpio; int i; /* override some hints */ stac_store_hints(codec); /* set up GPIO */ - gpio = spec->gpio_data; /* turn on EAPD statically when spec->eapd_switch isn't set. * otherwise, unsol event will turn it on/off dynamically */ if (!spec->eapd_switch) - gpio |= spec->eapd_mask; - stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, gpio); + spec->gpio_data |= spec->eapd_mask; + stac_gpio_set(codec, spec->gpio_mask, spec->gpio_dir, spec->gpio_data); snd_hda_gen_init(codec); @@ -3915,6 +3915,7 @@ static void stac_setup_gpio(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; + spec->gpio_mask |= spec->eapd_mask; if (spec->gpio_led) { if (!spec->vref_mute_led_nid) { spec->gpio_mask |= spec->gpio_led; -- cgit v1.2.3-70-g09d2 From f3e351eef3a7fd1e36a3e18d4f2f069b00deb23c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 19 Jul 2013 08:02:25 +0200 Subject: ALSA: hda - Remove NO_PRESENCE bit override for Dell 1420n Laptop The quirk for Dell laptops with STAC9228 overrides the pin default config of NID 0x0f to the value with AC_DEFCFG_MISC_NO_PRESENCE bit on. I'm not quite sure why this was done so, but can guess that this was introduced for avoiding this to be muted by another headphone plug. Now, after transition to the generic parser, this workaround rather causes a problem (notably as unexpected speaker mutes) because the pin is seen as if it's always plugged in. Since the generic parser can handle multiple headphone plugging gracefully, we can get rid of this override now. Reported-and-tested-by: Eric Shattow Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_sigmatel.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 766e56754c6..92b9b432437 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -3233,7 +3233,7 @@ static const struct hda_fixup stac927x_fixups[] = { /* configure the analog microphone on some laptops */ { 0x0c, 0x90a79130 }, /* correct the front output jack as a hp out */ - { 0x0f, 0x0227011f }, + { 0x0f, 0x0221101f }, /* correct the front input jack as a mic */ { 0x0e, 0x02a79130 }, {} -- cgit v1.2.3-70-g09d2 From 83e2e4eeb85fd45ff592b79ea11a19df49df872e Mon Sep 17 00:00:00 2001 From: H Hartley Sweeten Date: Fri, 19 Jul 2013 09:53:25 -0700 Subject: ASoC: ep93xx: fix build of ep93xx-ac97.c Fix the build of this driver. It was broken by: Commit 453807f3006757a5661c4000262d7d9284b5214c ASoC: ep93xx: Use ep93xx_dma_params instead of ep93xx_pcm_dma_params The removed struct ep93xx_pcm_dma_params use the member 'dma_port' to select the dma channel. The struct ep93xx_dma_data uses the member 'port'. Signed-off-by: H Hartley Sweeten Cc: Ryan Mallon Cc: Lars-Peter Clausen Cc: Mark Brown Cc: Liam Girdwood Cc: Jaroslav Kysela Cc: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/cirrus/ep93xx-ac97.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/cirrus/ep93xx-ac97.c b/sound/soc/cirrus/ep93xx-ac97.c index ac73c607410..04491f0e8d1 100644 --- a/sound/soc/cirrus/ep93xx-ac97.c +++ b/sound/soc/cirrus/ep93xx-ac97.c @@ -102,13 +102,13 @@ static struct ep93xx_ac97_info *ep93xx_ac97_info; static struct ep93xx_dma_data ep93xx_ac97_pcm_out = { .name = "ac97-pcm-out", - .dma_port = EP93XX_DMA_AAC1, + .port = EP93XX_DMA_AAC1, .direction = DMA_MEM_TO_DEV, }; static struct ep93xx_dma_data ep93xx_ac97_pcm_in = { .name = "ac97-pcm-in", - .dma_port = EP93XX_DMA_AAC1, + .port = EP93XX_DMA_AAC1, .direction = DMA_DEV_TO_MEM, }; -- cgit v1.2.3-70-g09d2 From be2f93a4c4981b3646b6f98f477154411b8516cb Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Fri, 19 Jul 2013 18:26:53 +0200 Subject: ALSA: usb-audio: 6fire: return correct XRUN indication Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of 6fire, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Signed-off-by: Eldad Zack Cc: Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 2aa4e13063a..3d2551cc10f 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -543,7 +543,7 @@ static snd_pcm_uframes_t usb6fire_pcm_pointer( snd_pcm_uframes_t ret; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); ret = sub->dma_off; -- cgit v1.2.3-70-g09d2 From b5c745fb75b7e5ab06e9c99d63427595a234cc89 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Mon, 22 Jul 2013 09:56:54 +0300 Subject: ASoC: core: double free in snd_soc_add_platform() There are three callers for this function, and none of them want it to free platform for them. It leads to a double free. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0ec070cf723..d82ee386eab 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3908,10 +3908,8 @@ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, { /* create platform component name */ platform->name = fmt_single_name(dev, &platform->id); - if (platform->name == NULL) { - kfree(platform); + if (platform->name == NULL) return -ENOMEM; - } platform->dev = dev; platform->driver = platform_drv; -- cgit v1.2.3-70-g09d2 From 647ab784c507763bfda79155f125b6edd1244806 Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Sun, 21 Jul 2013 10:34:09 +0800 Subject: ASoC: tegra: correct playback_dma_data setup The errors were caused by copy/paste mistake in below commit since v3.10: 3489d50 ASoC: tegra: Use common DAI DMA data struct It also corrects slave_id initialization in tegra20_ac97 driver. Signed-off-by: Richard Zhao Acked-by: Stephen Warren Acked-by: Lucas Stach Signed-off-by: Mark Brown Cc: # 3.10 --- sound/soc/tegra/tegra20_ac97.c | 6 +++--- sound/soc/tegra/tegra20_spdif.c | 4 ++-- 2 files changed, 5 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index e58233f7df6..6c486625321 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -389,9 +389,9 @@ static int tegra20_ac97_platform_probe(struct platform_device *pdev) ac97->capture_dma_data.slave_id = of_dma[1]; ac97->playback_dma_data.addr = mem->start + TEGRA20_AC97_FIFO_TX1; - ac97->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - ac97->capture_dma_data.maxburst = 4; - ac97->capture_dma_data.slave_id = of_dma[0]; + ac97->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + ac97->playback_dma_data.maxburst = 4; + ac97->playback_dma_data.slave_id = of_dma[1]; ret = tegra_asoc_utils_init(&ac97->util_data, &pdev->dev); if (ret) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 5eaa12cdc6e..551b3c93ce9 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -323,8 +323,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) } spdif->playback_dma_data.addr = mem->start + TEGRA20_SPDIF_DATA_OUT; - spdif->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; - spdif->capture_dma_data.maxburst = 4; + spdif->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + spdif->playback_dma_data.maxburst = 4; spdif->playback_dma_data.slave_id = dmareq->start; pm_runtime_enable(&pdev->dev); -- cgit v1.2.3-70-g09d2 From fee4b700a4e9e446151eb5a03874ca8666323113 Mon Sep 17 00:00:00 2001 From: Eldad Zack Date: Tue, 23 Jul 2013 11:15:06 +0200 Subject: ALSA: hiface: return correct XRUN indication Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer function of hiface, as expected by snd_pcm_update_hw_ptr0(). Caught by sparse. Cc: Antonio Ospite Signed-off-by: Eldad Zack Cc: Signed-off-by: Takashi Iwai --- sound/usb/hiface/pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c index 6430ed2a9f6..c21a3df9a0d 100644 --- a/sound/usb/hiface/pcm.c +++ b/sound/usb/hiface/pcm.c @@ -503,7 +503,7 @@ static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) snd_pcm_uframes_t dma_offset; if (rt->panic || !sub) - return SNDRV_PCM_STATE_XRUN; + return SNDRV_PCM_POS_XRUN; spin_lock_irqsave(&sub->lock, flags); dma_offset = sub->dma_off; -- cgit v1.2.3-70-g09d2 From 56a678344273fd63f8ade26876283a2586a9bf3a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 24 Jul 2013 15:27:35 +0200 Subject: ASoC: dapm: Fix return value of snd_soc_dapm_put_{volsw,enum_virt}() The ALSA core expect the put callback of a control to return 1 if the value of the control changed and 0 if it did not. Both snd_soc_dapm_put_volsw() and snd_soc_dapm_put_enum_virt() currently always returns 0. For both functions we already have a 'change' variable which either contains 1 or 0 depending on whether the value has changed or not, so just return that. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b94190820e8..bd16010441c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2733,7 +2733,7 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, } mutex_unlock(&card->dapm_mutex); - return 0; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_volsw); @@ -2861,7 +2861,6 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int change; - int ret = 0; int wi; if (ucontrol->value.enumerated.item[0] >= e->max) @@ -2881,7 +2880,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, } mutex_unlock(&card->dapm_mutex); - return ret; + return change; } EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_virt); -- cgit v1.2.3-70-g09d2 From a8d30608eaed6cc759b8e2e8a8bbbb42591f797f Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Mon, 29 Jul 2013 15:10:22 +0530 Subject: ALSA: compress: fix the return value for SNDRV_COMPRESS_VERSION the return value of SNDRV_COMPRESS_VERSION always return default -ENOTTY as the return value was never updated for this call assign return value from put_user() Reported-by: Haynes CC: stable@vger.kernel.org Signed-off-by: Vinod Koul Signed-off-by: Takashi Iwai --- sound/core/compress_offload.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 99db892d729..98969541cbc 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -743,7 +743,7 @@ static long snd_compr_ioctl(struct file *f, unsigned int cmd, unsigned long arg) mutex_lock(&stream->device->lock); switch (_IOC_NR(cmd)) { case _IOC_NR(SNDRV_COMPRESS_IOCTL_VERSION): - put_user(SNDRV_COMPRESS_VERSION, + retval = put_user(SNDRV_COMPRESS_VERSION, (int __user *)arg) ? -EFAULT : 0; break; case _IOC_NR(SNDRV_COMPRESS_GET_CAPS): -- cgit v1.2.3-70-g09d2 From 1deb57042fe2bd14cd7d4687f3c9418d26862053 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 29 Jul 2013 08:50:28 +0100 Subject: ASoC: bfin-ac97: Fix prototype error following AC'97 refactoring As part of the multiplatform refactoring for AC'97 the AC'97 bus ops were staticised meaning that the prototype (which was never needed) conflicts with the declaration causing build failures. Signed-off-by: Mark Brown Acked-by: Lars-Peter Clausen --- sound/soc/blackfin/bf5xx-ac97.h | 1 - 1 file changed, 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h index 15c635e33f4..0c3e22d90a8 100644 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ b/sound/soc/blackfin/bf5xx-ac97.h @@ -9,7 +9,6 @@ #ifndef _BF5XX_AC97_H #define _BF5XX_AC97_H -extern struct snd_ac97_bus_ops bf5xx_ac97_ops; extern struct snd_ac97 *ac97; /* Frame format in memory, only support stereo currently */ struct ac97_frame { -- cgit v1.2.3-70-g09d2 From 610d80eaa987e7b1a2d07ee800c9722e227a3b47 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 30 Jul 2013 13:34:09 +0200 Subject: ASoC: bf5xx-ac97: Fix compile error with SND_BF5XX_HAVE_COLD_RESET MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit If CONFIG_SND_BF5XX_HAVE_COLD_RESET is enabled building the blackfin ac97 driver fails with the following compile error: sound/soc/blackfin/bf5xx-ac97.c: In function ‘asoc_bfin_ac97_probe’: sound/soc/blackfin/bf5xx-ac97.c:297: error: expected ‘;’ before ‘{’ token sound/soc/blackfin/bf5xx-ac97.c:302: error: label ‘gpio_err’ used but not defined The issue was introduced in commit 6dab2fd7 ("ASoC: bf5xx-ac97: Convert to devm_gpio_request_one()"). Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/bf5xx-ac97.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c index efb1daecd0d..e82eb373a73 100644 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ b/sound/soc/blackfin/bf5xx-ac97.c @@ -294,11 +294,12 @@ static int asoc_bfin_ac97_probe(struct platform_device *pdev) /* Request PB3 as reset pin */ ret = devm_gpio_request_one(&pdev->dev, CONFIG_SND_BF5XX_RESET_GPIO_NUM, - GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET") { + GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET"); + if (ret) { dev_err(&pdev->dev, "Failed to request GPIO_%d for reset: %d\n", CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); - goto gpio_err; + return ret; } #endif -- cgit v1.2.3-70-g09d2 From d2ee88d0aaacac664aff6ca5fc0bd7705d8f2414 Mon Sep 17 00:00:00 2001 From: Ralf Baechle Date: Wed, 31 Jul 2013 10:15:19 +0200 Subject: ASoC: au1x: Fix build MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit d8b51c11ff5a70244753ba60abfd47088cf4dcd4 [ASoC: ac97c: Use module_platform_driver()] broke the build: CC sound/soc/au1x/ac97c.o /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected identifier or ‘(’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: pasting "__initcall_" and "&" does not give a valid preprocessing token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected identifier or ‘(’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: pasting "__exitcall_" and "&" does not give a valid preprocessing token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:344:1: error: expected ‘=’, ‘,’, ‘;’, ‘asm’ or ‘__attribute__’ before ‘&’ token /home/ralf/src/linux/upstream-sfr/sound/soc/au1x/ac97c.c:334:31: warning: ‘au1xac97c_driver’ defined but not used [-Wunused-variable] make[5]: *** [sound/soc/au1x/ac97c.o] Error 1 make[4]: *** [sound/soc/au1x] Error 2 make[3]: *** [sound/soc] Error 2 Signed-off-by: Ralf Baechle Signed-off-by: Mark Brown --- sound/soc/au1x/ac97c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d6f7694fcad..c8a2de103c5 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -341,7 +341,7 @@ static struct platform_driver au1xac97c_driver = { .remove = au1xac97c_drvremove, }; -module_platform_driver(&au1xac97c_driver); +module_platform_driver(au1xac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au1000/1500/1100 AC97C ASoC driver"); -- cgit v1.2.3-70-g09d2 From 4f8b19143d74e1c3360b21640065765a12bafb1b Mon Sep 17 00:00:00 2001 From: Dimitris Papastamos Date: Wed, 31 Jul 2013 13:28:52 +0100 Subject: ASoC: wm0010: Fix resource leak If kzalloc() fails for `img' then we are going to leak the memory for `out'. We are freeing the memory of all the tx/rx transfers but the tx/rx buf pointers will be NULL if we drop out earlier. Signed-off-by: Dimitris Papastamos Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f5e835662cd..10adc4145d4 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -410,6 +410,16 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) rec->command, rec->length); len = rec->length + 8; + xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); + if (!xfer) { + dev_err(codec->dev, "Failed to allocate xfer\n"); + ret = -ENOMEM; + goto abort; + } + + xfer->codec = codec; + list_add_tail(&xfer->list, &xfer_list); + out = kzalloc(len, GFP_KERNEL); if (!out) { dev_err(codec->dev, @@ -417,6 +427,7 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.rx_buf = out; img = kzalloc(len, GFP_KERNEL); if (!img) { @@ -425,24 +436,13 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = -ENOMEM; goto abort1; } + xfer->t.tx_buf = img; byte_swap_64((u64 *)&rec->command, img, len); - xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); - if (!xfer) { - dev_err(codec->dev, "Failed to allocate xfer\n"); - ret = -ENOMEM; - goto abort1; - } - - xfer->codec = codec; - list_add_tail(&xfer->list, &xfer_list); - spi_message_init(&xfer->m); xfer->m.complete = wm0010_boot_xfer_complete; xfer->m.context = xfer; - xfer->t.tx_buf = img; - xfer->t.rx_buf = out; xfer->t.len = len; xfer->t.bits_per_word = 8; -- cgit v1.2.3-70-g09d2 From f091f3f07328f75d20a2a5970d1f8b58d95fc990 Mon Sep 17 00:00:00 2001 From: Lothar Waßmann Date: Wed, 31 Jul 2013 16:44:29 +0200 Subject: ASoC: sgtl5000: prevent playback to be muted when terminating concurrent capture MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit When a sound capture/playback is terminated while a playback/capture is running, power_vag_event() will clear SGTL5000_CHIP_ANA_POWER in the SND_SOC_DAPM_PRE_PMD event, thus muting the respective other channel. Don't clear SGTL5000_CHIP_ANA_POWER when both DAC and ADC are active to prevent this. Signed-off-by: Lothar Waßmann Reviewed-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 16 +++++++++++++--- 1 file changed, 13 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 6c8a9e7bee2..9303c7d011b 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -153,6 +153,8 @@ static int mic_bias_event(struct snd_soc_dapm_widget *w, static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + const u32 mask = SGTL5000_DAC_POWERUP | SGTL5000_ADC_POWERUP; + switch (event) { case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, @@ -160,9 +162,17 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_VAG_POWERUP, 0); - msleep(400); + /* + * Don't clear VAG_POWERUP, when both DAC and ADC are + * operational to prevent inadvertently starving the + * other one of them. + */ + if ((snd_soc_read(w->codec, SGTL5000_CHIP_ANA_POWER) & + mask) != mask) { + snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, + SGTL5000_VAG_POWERUP, 0); + msleep(400); + } break; default: break; -- cgit v1.2.3-70-g09d2 From 65f2b226763bc348a9b9145aa5e17e7e3f6d8c35 Mon Sep 17 00:00:00 2001 From: Lothar Waßmann Date: Wed, 31 Jul 2013 16:44:30 +0200 Subject: ASoC: sgtl5000: fix buggy 'Capture Attenuate Switch' control MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The SGTL5000 Capture Attenuate Switch (or "ADC Volume Range Reduction" as it is called in the manual) is single bit only. Signed-off-by: Lothar Waßmann Reviewed-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 9303c7d011b..760e8bfeaca 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -398,7 +398,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { SOC_DOUBLE("Capture Volume", SGTL5000_CHIP_ANA_ADC_CTRL, 0, 4, 0xf, 0), SOC_SINGLE_TLV("Capture Attenuate Switch (-6dB)", SGTL5000_CHIP_ANA_ADC_CTRL, - 8, 2, 0, capture_6db_attenuate), + 8, 1, 0, capture_6db_attenuate), SOC_SINGLE("Capture ZC Switch", SGTL5000_CHIP_ANA_CTRL, 1, 1, 0), SOC_DOUBLE_TLV("Headphone Playback Volume", -- cgit v1.2.3-70-g09d2 From fe581391147cb3d738d961d0f1233d91a9e1113c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 1 Aug 2013 18:30:38 +0200 Subject: ASoC: dapm: Fix empty list check in dapm_new_mux() list_first_entry() will always return a valid pointer, even if the list is empty. So the check whether path is NULL will always be false. So we end up calling dapm_create_or_share_mixmux_kcontrol() with a path struct that points right in the middle of the widget struct and by trying to modify the path the widgets memory will become corrupted. Fix this by using list_emtpy() to check if the widget doesn't have any paths. Signed-off-by: Lars-Peter Clausen Tested-by: Stephen Warren Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/soc-dapm.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index bd16010441c..4375c9f2b79 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -679,13 +679,14 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return -EINVAL; } - path = list_first_entry(&w->sources, struct snd_soc_dapm_path, - list_sink); - if (!path) { + if (list_empty(&w->sources)) { dev_err(dapm->dev, "ASoC: mux %s has no paths\n", w->name); return -EINVAL; } + path = list_first_entry(&w->sources, struct snd_soc_dapm_path, + list_sink); + ret = dapm_create_or_share_mixmux_kcontrol(w, 0, path); if (ret < 0) return ret; -- cgit v1.2.3-70-g09d2 From 697aebab78a88c6b164cfb74d19b86817d2ccd82 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 1 Aug 2013 08:38:27 +0200 Subject: ALSA: hda - Fix missing fixup for Mac Mini with STAC9221 A fixup for Apple Mac Mini was lost during the adaption to the generic parser because the fallback for the generic ID 8384:7680 was dropped, and it resulted in the silence output (and maybe other problems). Unfortunately, just adding the missing subsystem ID wasn't enough, in this case. The subsystem ID of this machine is 0000:0100 (what Apple thought...?), and since snd_hda_pick_fixup() doesn't take the vendor id zero into account, the driver ignored this entry. Now it's fixed to regard the vendor id zero as a valid value. Reported-and-tested-by: Linus Torvalds Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_auto_parser.c | 2 +- sound/pci/hda/patch_sigmatel.c | 1 + 2 files changed, 2 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_auto_parser.c b/sound/pci/hda/hda_auto_parser.c index 7c11d46b84d..48a9d004d6d 100644 --- a/sound/pci/hda/hda_auto_parser.c +++ b/sound/pci/hda/hda_auto_parser.c @@ -860,7 +860,7 @@ void snd_hda_pick_fixup(struct hda_codec *codec, } } if (id < 0 && quirk) { - for (q = quirk; q->subvendor; q++) { + for (q = quirk; q->subvendor || q->subdevice; q++) { unsigned int vendorid = q->subdevice | (q->subvendor << 16); unsigned int mask = 0xffff0000 | q->subdevice_mask; diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 92b9b432437..6d1924c19ab 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2819,6 +2819,7 @@ static const struct hda_pintbl ecs202_pin_configs[] = { /* codec SSIDs for Intel Mac sharing the same PCI SSID 8384:7680 */ static const struct snd_pci_quirk stac922x_intel_mac_fixup_tbl[] = { + SND_PCI_QUIRK(0x0000, 0x0100, "Mac Mini", STAC_INTEL_MAC_V3), SND_PCI_QUIRK(0x106b, 0x0800, "Mac", STAC_INTEL_MAC_V1), SND_PCI_QUIRK(0x106b, 0x0600, "Mac", STAC_INTEL_MAC_V2), SND_PCI_QUIRK(0x106b, 0x0700, "Mac", STAC_INTEL_MAC_V2), -- cgit v1.2.3-70-g09d2 From e2c98a8bba958045bde861fe1d66be54315c7790 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 6 Aug 2013 12:57:21 -0500 Subject: ASoC: cs42l52: Reorder Min/Max and update to SX_TLV for Beep Volume Beep Volume Min/Max was backwards. Change to SOC_SONGLE_SX_TLV for correct volume representation Signed-off-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@kernel.org --- sound/soc/codecs/cs42l52.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 987f728718c..ee25f325d65 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -451,7 +451,7 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x1f, 0x07, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- cgit v1.2.3-70-g09d2 From 8806d96db7b04fffba4cfc9ceac31d24c8517fe9 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 6 Aug 2013 12:57:22 -0500 Subject: ASoC: cs42l52: Add new TLV for Beep Volume CS42L52 Beep control uses 2dB scale from -56dB Signed-off-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index ee25f325d65..be2ba1b6fe4 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -195,6 +195,8 @@ static DECLARE_TLV_DB_SCALE(pga_tlv, -600, 50, 0); static DECLARE_TLV_DB_SCALE(mix_tlv, -50, 50, 0); +static DECLARE_TLV_DB_SCALE(beep_tlv, -56, 200, 0); + static const unsigned int limiter_tlv[] = { TLV_DB_RANGE_HEAD(2), 0, 2, TLV_DB_SCALE_ITEM(-3000, 600, 0), @@ -451,7 +453,8 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("Beep Pitch", beep_pitch_enum), SOC_ENUM("Beep on Time", beep_ontime_enum), SOC_ENUM("Beep off Time", beep_offtime_enum), - SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, 0, 0x07, 0x1f, hl_tlv), + SOC_SINGLE_SX_TLV("Beep Volume", CS42L52_BEEP_VOL, + 0, 0x07, 0x1f, beep_tlv), SOC_SINGLE("Beep Mixer Switch", CS42L52_BEEP_TONE_CTL, 5, 1, 1), SOC_ENUM("Beep Treble Corner Freq", beep_treble_enum), SOC_ENUM("Beep Bass Corner Freq", beep_bass_enum), -- cgit v1.2.3-70-g09d2 From ddb6b5a964371e8e52e696b2b258bda144c8bd3f Mon Sep 17 00:00:00 2001 From: Jussi Kivilinna Date: Tue, 6 Aug 2013 14:53:24 +0300 Subject: ALSA: 6fire: fix DMA issues with URB transfer_buffer usage Patch fixes 6fire not to use stack as URB transfer_buffer. URB buffers need to be DMA-able, which stack is not. Furthermore, transfer_buffer should not be allocated as part of larger device structure because DMA coherency issues and patch fixes this issue too. Cc: stable@vger.kernel.org Signed-off-by: Jussi Kivilinna Tested-by: Torsten Schenk Signed-off-by: Takashi Iwai --- sound/usb/6fire/comm.c | 38 +++++++++++++++++++++++++++++++++----- sound/usb/6fire/comm.h | 2 +- 2 files changed, 34 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/comm.c b/sound/usb/6fire/comm.c index 9e6e3ffd86b..23452ee617e 100644 --- a/sound/usb/6fire/comm.c +++ b/sound/usb/6fire/comm.c @@ -110,19 +110,37 @@ static int usb6fire_comm_send_buffer(u8 *buffer, struct usb_device *dev) static int usb6fire_comm_write8(struct comm_runtime *rt, u8 request, u8 reg, u8 value) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, value, 0x00); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } static int usb6fire_comm_write16(struct comm_runtime *rt, u8 request, u8 reg, u8 vl, u8 vh) { - u8 buffer[13]; /* 13: maximum length of message */ + u8 *buffer; + int ret; + + /* 13: maximum length of message */ + buffer = kmalloc(13, GFP_KERNEL); + if (!buffer) + return -ENOMEM; usb6fire_comm_init_buffer(buffer, 0x00, request, reg, vl, vh); - return usb6fire_comm_send_buffer(buffer, rt->chip->dev); + ret = usb6fire_comm_send_buffer(buffer, rt->chip->dev); + + kfree(buffer); + return ret; } int usb6fire_comm_init(struct sfire_chip *chip) @@ -135,6 +153,12 @@ int usb6fire_comm_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + rt->receiver_buffer = kzalloc(COMM_RECEIVER_BUFSIZE, GFP_KERNEL); + if (!rt->receiver_buffer) { + kfree(rt); + return -ENOMEM; + } + urb = &rt->receiver; rt->serial = 1; rt->chip = chip; @@ -153,6 +177,7 @@ int usb6fire_comm_init(struct sfire_chip *chip) urb->interval = 1; ret = usb_submit_urb(urb, GFP_KERNEL); if (ret < 0) { + kfree(rt->receiver_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "cannot create comm data receiver."); return ret; @@ -171,6 +196,9 @@ void usb6fire_comm_abort(struct sfire_chip *chip) void usb6fire_comm_destroy(struct sfire_chip *chip) { - kfree(chip->comm); + struct comm_runtime *rt = chip->comm; + + kfree(rt->receiver_buffer); + kfree(rt); chip->comm = NULL; } diff --git a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h index 6a0840b0dcf..780d5ed8e5d 100644 --- a/sound/usb/6fire/comm.h +++ b/sound/usb/6fire/comm.h @@ -24,7 +24,7 @@ struct comm_runtime { struct sfire_chip *chip; struct urb receiver; - u8 receiver_buffer[COMM_RECEIVER_BUFSIZE]; + u8 *receiver_buffer; u8 serial; /* urb serial */ -- cgit v1.2.3-70-g09d2 From 57e6dae1087bbaa6b33d3dd8a8e90b63888939a3 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Thu, 8 Aug 2013 11:24:55 +0200 Subject: ALSA: usb-audio: do not trust too-big wMaxPacketSize values The driver used to assume that the streaming endpoint's wMaxPacketSize value would be an indication of how much data the endpoint expects or sends, and compute the number of packets per URB using this value. However, the Focusrite Scarlett 2i4 declares a value of 1024 bytes, while only about 88 or 44 bytes are be actually used. This discrepancy would result in URBs with far too few packets, which would not work correctly on the EHCI driver. To get correct URBs, use wMaxPacketSize only as an upper limit on the packet size. Reported-by: James Stone Tested-by: James Stone Cc: # 2.6.35+ Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/endpoint.c | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) (limited to 'sound') diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 7a444b5501d..659950e5b94 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -591,17 +591,16 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, ep->stride = frame_bits >> 3; ep->silence_value = pcm_format == SNDRV_PCM_FORMAT_U8 ? 0x80 : 0; - /* calculate max. frequency */ - if (ep->maxpacksize) { + /* assume max. frequency is 25% higher than nominal */ + ep->freqmax = ep->freqn + (ep->freqn >> 2); + maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) + >> (16 - ep->datainterval); + /* but wMaxPacketSize might reduce this */ + if (ep->maxpacksize && ep->maxpacksize < maxsize) { /* whatever fits into a max. size packet */ maxsize = ep->maxpacksize; ep->freqmax = (maxsize / (frame_bits >> 3)) << (16 - ep->datainterval); - } else { - /* no max. packet size: just take 25% higher than nominal */ - ep->freqmax = ep->freqn + (ep->freqn >> 2); - maxsize = ((ep->freqmax + 0xffff) * (frame_bits >> 3)) - >> (16 - ep->datainterval); } if (ep->fill_max) -- cgit v1.2.3-70-g09d2 From db8a38e5063a4daf61252e65d47ab3495c705f4c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 9 Aug 2013 12:34:42 +0200 Subject: ALSA: hda - Add pinfix for LG LW25 laptop Correct the pins for a line-in and a headphone on LG LW25 laptop with ALC880 codec. Other pins seem fine. Reported-and-tested-by: Joonas Saarinen Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound') diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 8bd22614986..5b22bf95876 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1031,6 +1031,7 @@ enum { ALC880_FIXUP_GPIO2, ALC880_FIXUP_MEDION_RIM, ALC880_FIXUP_LG, + ALC880_FIXUP_LG_LW25, ALC880_FIXUP_W810, ALC880_FIXUP_EAPD_COEF, ALC880_FIXUP_TCL_S700, @@ -1089,6 +1090,14 @@ static const struct hda_fixup alc880_fixups[] = { { } } }, + [ALC880_FIXUP_LG_LW25] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x0181344f }, /* line-in */ + { 0x1b, 0x0321403f }, /* headphone */ + { } + } + }, [ALC880_FIXUP_W810] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -1341,6 +1350,7 @@ static const struct snd_pci_quirk alc880_fixup_tbl[] = { SND_PCI_QUIRK(0x1854, 0x003b, "LG", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x005f, "LG P1 Express", ALC880_FIXUP_LG), SND_PCI_QUIRK(0x1854, 0x0068, "LG w1", ALC880_FIXUP_LG), + SND_PCI_QUIRK(0x1854, 0x0077, "LG LW25", ALC880_FIXUP_LG_LW25), SND_PCI_QUIRK(0x19db, 0x4188, "TCL S700", ALC880_FIXUP_TCL_S700), /* Below is the copied entries from alc880_quirks.c. -- cgit v1.2.3-70-g09d2 From 5ece263f1d93fba8d992e67e3ab8a71acf674db9 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Sun, 11 Aug 2013 11:11:19 +0200 Subject: ALSA: 6fire: make buffers DMA-able (pcm) Patch makes pcm buffers DMA-able by allocating each one separately. Signed-off-by: Torsten Schenk Cc: Signed-off-by: Takashi Iwai --- sound/usb/6fire/pcm.c | 41 ++++++++++++++++++++++++++++++++++++++++- sound/usb/6fire/pcm.h | 2 +- 2 files changed, 41 insertions(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/pcm.c b/sound/usb/6fire/pcm.c index 3d2551cc10f..b5eb97fdc84 100644 --- a/sound/usb/6fire/pcm.c +++ b/sound/usb/6fire/pcm.c @@ -582,6 +582,33 @@ static void usb6fire_pcm_init_urb(struct pcm_urb *urb, urb->instance.number_of_packets = PCM_N_PACKETS_PER_URB; } +static int usb6fire_pcm_buffers_init(struct pcm_runtime *rt) +{ + int i; + + for (i = 0; i < PCM_N_URBS; i++) { + rt->out_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB + * PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!rt->out_urbs[i].buffer) + return -ENOMEM; + rt->in_urbs[i].buffer = kzalloc(PCM_N_PACKETS_PER_URB + * PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!rt->in_urbs[i].buffer) + return -ENOMEM; + } + return 0; +} + +static void usb6fire_pcm_buffers_destroy(struct pcm_runtime *rt) +{ + int i; + + for (i = 0; i < PCM_N_URBS; i++) { + kfree(rt->out_urbs[i].buffer); + kfree(rt->in_urbs[i].buffer); + } +} + int usb6fire_pcm_init(struct sfire_chip *chip) { int i; @@ -593,6 +620,13 @@ int usb6fire_pcm_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + ret = usb6fire_pcm_buffers_init(rt); + if (ret) { + usb6fire_pcm_buffers_destroy(rt); + kfree(rt); + return ret; + } + rt->chip = chip; rt->stream_state = STREAM_DISABLED; rt->rate = ARRAY_SIZE(rates); @@ -614,6 +648,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip) ret = snd_pcm_new(chip->card, "DMX6FireUSB", 0, 1, 1, &pcm); if (ret < 0) { + usb6fire_pcm_buffers_destroy(rt); kfree(rt); snd_printk(KERN_ERR PREFIX "cannot create pcm instance.\n"); return ret; @@ -625,6 +660,7 @@ int usb6fire_pcm_init(struct sfire_chip *chip) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_ops); if (ret) { + usb6fire_pcm_buffers_destroy(rt); kfree(rt); snd_printk(KERN_ERR PREFIX "error preallocating pcm buffers.\n"); @@ -669,6 +705,9 @@ void usb6fire_pcm_abort(struct sfire_chip *chip) void usb6fire_pcm_destroy(struct sfire_chip *chip) { - kfree(chip->pcm); + struct pcm_runtime *rt = chip->pcm; + + usb6fire_pcm_buffers_destroy(rt); + kfree(rt); chip->pcm = NULL; } diff --git a/sound/usb/6fire/pcm.h b/sound/usb/6fire/pcm.h index 9b01133ee3f..f5779d6182c 100644 --- a/sound/usb/6fire/pcm.h +++ b/sound/usb/6fire/pcm.h @@ -32,7 +32,7 @@ struct pcm_urb { struct urb instance; struct usb_iso_packet_descriptor packets[PCM_N_PACKETS_PER_URB]; /* END DO NOT SEPARATE */ - u8 buffer[PCM_N_PACKETS_PER_URB * PCM_MAX_PACKET_SIZE]; + u8 *buffer; struct pcm_urb *peer; }; -- cgit v1.2.3-70-g09d2 From 4c2aee0032b70083dafebd733ed9c774633b2fa3 Mon Sep 17 00:00:00 2001 From: Torsten Schenk Date: Sun, 11 Aug 2013 11:11:35 +0200 Subject: ALSA: 6fire: make buffers DMA-able (midi) Patch makes midi output buffer DMA-able by allocating it separately. Signed-off-by: Torsten Schenk Cc: Signed-off-by: Takashi Iwai --- sound/usb/6fire/midi.c | 16 +++++++++++++++- sound/usb/6fire/midi.h | 6 +----- 2 files changed, 16 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/usb/6fire/midi.c b/sound/usb/6fire/midi.c index 26722423330..f3dd7266c39 100644 --- a/sound/usb/6fire/midi.c +++ b/sound/usb/6fire/midi.c @@ -19,6 +19,10 @@ #include "chip.h" #include "comm.h" +enum { + MIDI_BUFSIZE = 64 +}; + static void usb6fire_midi_out_handler(struct urb *urb) { struct midi_runtime *rt = urb->context; @@ -156,6 +160,12 @@ int usb6fire_midi_init(struct sfire_chip *chip) if (!rt) return -ENOMEM; + rt->out_buffer = kzalloc(MIDI_BUFSIZE, GFP_KERNEL); + if (!rt->out_buffer) { + kfree(rt); + return -ENOMEM; + } + rt->chip = chip; rt->in_received = usb6fire_midi_in_received; rt->out_buffer[0] = 0x80; /* 'send midi' command */ @@ -169,6 +179,7 @@ int usb6fire_midi_init(struct sfire_chip *chip) ret = snd_rawmidi_new(chip->card, "6FireUSB", 0, 1, 1, &rt->instance); if (ret < 0) { + kfree(rt->out_buffer); kfree(rt); snd_printk(KERN_ERR PREFIX "unable to create midi.\n"); return ret; @@ -197,6 +208,9 @@ void usb6fire_midi_abort(struct sfire_chip *chip) void usb6fire_midi_destroy(struct sfire_chip *chip) { - kfree(chip->midi); + struct midi_runtime *rt = chip->midi; + + kfree(rt->out_buffer); + kfree(rt); chip->midi = NULL; } diff --git a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h index c321006e543..84851b9f555 100644 --- a/sound/usb/6fire/midi.h +++ b/sound/usb/6fire/midi.h @@ -16,10 +16,6 @@ #include "common.h" -enum { - MIDI_BUFSIZE = 64 -}; - struct midi_runtime { struct sfire_chip *chip; struct snd_rawmidi *instance; @@ -32,7 +28,7 @@ struct midi_runtime { struct snd_rawmidi_substream *out; struct urb out_urb; u8 out_serial; /* serial number of out packet */ - u8 out_buffer[MIDI_BUFSIZE]; + u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length); -- cgit v1.2.3-70-g09d2 From aa773bfe8f860173752258c9ba4bf51060fb0d07 Mon Sep 17 00:00:00 2001 From: Clemens Ladisch Date: Sun, 11 Aug 2013 14:13:13 +0200 Subject: ALSA: usb-audio: fix automatic Roland/Yamaha MIDI detection Commit aafe77cc45a5 (ALSA: usb-audio: add support for many Roland/Yamaha devices) had several logic errors that prevented create_auto_midi_quirk from enumerating any MIDI ports. Reported-by: Keith A. Milner Signed-off-by: Clemens Ladisch Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 1bc45e71f1f..0df9ede99df 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -319,19 +319,19 @@ static int create_auto_midi_quirk(struct snd_usb_audio *chip, if (altsd->bNumEndpoints < 1) return -ENODEV; epd = get_endpoint(alts, 0); - if (!usb_endpoint_xfer_bulk(epd) || + if (!usb_endpoint_xfer_bulk(epd) && !usb_endpoint_xfer_int(epd)) return -ENODEV; switch (USB_ID_VENDOR(chip->usb_id)) { case 0x0499: /* Yamaha */ err = create_yamaha_midi_quirk(chip, iface, driver, alts); - if (err < 0 && err != -ENODEV) + if (err != -ENODEV) return err; break; case 0x0582: /* Roland */ err = create_roland_midi_quirk(chip, iface, driver, alts); - if (err < 0 && err != -ENODEV) + if (err != -ENODEV) return err; break; } -- cgit v1.2.3-70-g09d2 From f69910ddbd8c29391958cf82b598dd78fe5c8640 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Thu, 8 Aug 2013 09:32:37 +0200 Subject: ALSA: hda - Fix missing mute controls for CX5051 We've added a fake mute control (setting the amp volume to zero) for CX5051 at commit [3868137e: ALSA: hda - Add a fake mute feature], but this feature was overlooked in the generic parser implementation. Now the driver lacks of mute controls on these codecs. The fix is just to check both AC_AMPCAP_MUTE and AC_AMPCAP_MIN_MUTE bits in each place checking the amp capabilities. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59001 Cc: [v3.9+] Signed-off-by: Takashi Iwai --- sound/pci/hda/hda_generic.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound') diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 8e77cbbad87..e3c7ba8d758 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -522,7 +522,7 @@ static bool same_amp_caps(struct hda_codec *codec, hda_nid_t nid1, } #define nid_has_mute(codec, nid, dir) \ - check_amp_caps(codec, nid, dir, AC_AMPCAP_MUTE) + check_amp_caps(codec, nid, dir, (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) #define nid_has_volume(codec, nid, dir) \ check_amp_caps(codec, nid, dir, AC_AMPCAP_NUM_STEPS) @@ -624,7 +624,7 @@ static int get_amp_val_to_activate(struct hda_codec *codec, hda_nid_t nid, if (enable) val = (caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT; } - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (!enable) val |= HDA_AMP_MUTE; } @@ -648,7 +648,7 @@ static unsigned int get_amp_mask_to_modify(struct hda_codec *codec, { unsigned int mask = 0xff; - if (caps & AC_AMPCAP_MUTE) { + if (caps & (AC_AMPCAP_MUTE | AC_AMPCAP_MIN_MUTE)) { if (is_ctl_associated(codec, nid, dir, idx, NID_PATH_MUTE_CTL)) mask &= ~0x80; } -- cgit v1.2.3-70-g09d2 From 140d37de62ffe8405282a1d6498f3b4099006384 Mon Sep 17 00:00:00 2001 From: "Maksim A. Boyko" Date: Sat, 10 Aug 2013 12:20:02 +0400 Subject: ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam C525 Add the volume control quirk for avoiding the kernel warning for the Logitech HD Webcam C525 as in the similar commit 36691e1be6ec551eef4a5225f126a281f8c051c2 for the Logitech HD Webcam C310. Reported-by: Maksim Boyko Tested-by: Maksim Boyko Cc: # 3.10.5+ Signed-off-by: Maksim Boyko Signed-off-by: Takashi Iwai --- sound/usb/mixer.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound') diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index d5438083fd6..95558ef4a7a 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -888,6 +888,7 @@ static void volume_control_quirks(struct usb_mixer_elem_info *cval, case USB_ID(0x046d, 0x081b): /* HD Webcam c310 */ case USB_ID(0x046d, 0x081d): /* HD Webcam c510 */ case USB_ID(0x046d, 0x0825): /* HD Webcam c270 */ + case USB_ID(0x046d, 0x0826): /* HD Webcam c525 */ case USB_ID(0x046d, 0x0991): /* Most audio usb devices lie about volume resolution. * Most Logitech webcams have res = 384. -- cgit v1.2.3-70-g09d2 From c90c0d7a96e634a73ef1580f1d20993606545647 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 14 Aug 2013 14:24:16 -0600 Subject: ASoC: tegra: fix Tegra30 I2S capture parameter setup The Tegra30 I2S driver was writing the AHUB interface parameters to the playback path register rather than the capture path register. This caused the capture parameters not to be configured at all, so if capturing using non-HW-default parameters (e.g. 16-bit stereo rather than 8-bit mono) the audio would be corrupted. With this fixed, audio capture from an analog microphone works correctly on the Cardhu board. Cc: stable@vger.kernel.org Signed-off-by: Stephen Warren Signed-off-by: Mark Brown --- sound/soc/tegra/tegra30_i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound') diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index d04146cad61..47565fd0450 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -228,7 +228,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, reg = TEGRA30_I2S_CIF_RX_CTRL; } else { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - reg = TEGRA30_I2S_CIF_RX_CTRL; + reg = TEGRA30_I2S_CIF_TX_CTRL; } regmap_write(i2s->regmap, reg, val); -- cgit v1.2.3-70-g09d2 From 1801928e0f99d94c55e33c584c5eb2ff5e246ee6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 16 Aug 2013 08:17:05 +0200 Subject: ALSA: hda - Add a fixup for Gateway LT27 Gateway LT27 needs a fixup for the inverted digital mic. Reported-by: "Nathanael D. Noblet" Cc: 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 5b22bf95876..f303cd89851 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4339,6 +4339,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x0349, "eMachines eM250", ALC662_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1025, 0x034a, "Gateway LT27", ALC662_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), -- cgit v1.2.3-70-g09d2