January 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2] Fix thread safety issues
by Jerome Forissier 31 Jan '13

31 Jan '13

This is a set of two patches to fix thread safety issues I found when running a video application over ALSA. The application uses LibVLC to play several videos simultaneously. LibVLC calls snd_pcm_open() and snd_device_name_hint() from multiple threads, without locking--which is allowed according to http://www.alsa-project.org/main/index.php/SMP_Design. Without these patches, I would get random error messages like: ALSA lib confmisc.c:768:(parse_card) cannot find card '$CARD' ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device ALSA lib confmisc.c:392:(snd_func_concat) error evaluating strings ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_concat returned error: No such device ALSA lib confmisc.c:1251:(snd_func_refer) error evaluating name ALSA lib conf.c:4184:(_snd_config_evaluate) function snd_func_refer returned error: No such device ALSA lib conf.c:4663:(snd_config_expand) Evaluate error: No such device Jerome Forissier (2): Make snd_device_name_hint() thread-safe by locking a mutex snd_pcm_direct_parse_open_conf(): use thread-safe getgrnam_r() src/control/namehint.c | 31 +++++++++++++++++++++++++++++++ src/pcm/pcm_direct.c | 15 +++++++++++---- 2 files changed, 42 insertions(+), 4 deletions(-) -- 1.8.1.2

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[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Convert mic bias to a supply widget
by Hebbar Gururaja 31 Jan '13

31 Jan '13

Convert MicBias widgets to supply widget. On tlv320aic3x, Mic bias power on/off shares the same register bits with output mic bias voltage. So, when power on mic bias, we need reclaim it to voltage value. Provide a new platform data so that the micbias voltage can be sent according to board requirement. Now since tlc320aic3x codec driver is DT aware, update dt files and functions to handle this new "micbias-vg" platform data. Because of sharing of bits, when enabling the micbias, voltage also needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD macro to create an event to handle this. Signed-off-by: Hebbar Gururaja <gururaja.hebbar(a)ti.com> --- Mark, I have updated the codec driver as you suggested. Can you confirm if this is sufficient?. If yes, then I will send all the patch once again with new revision. :100644 100644 e7b98f4... f47c3f5... M Documentation/devicetree/bindings/sound/tlv320aic3x.txt :100644 100644 ffd9bc7... 9407fd0... M include/sound/tlv320aic3x.h :100644 100644 4989143... fe6dd27... M sound/soc/codecs/tlv320aic3x.c :100644 100644 6db3c41... e521ac3... M sound/soc/codecs/tlv320aic3x.h .../devicetree/bindings/sound/tlv320aic3x.txt | 6 ++ include/sound/tlv320aic3x.h | 10 +++ sound/soc/codecs/tlv320aic3x.c | 85 ++++++++++++++++++-- sound/soc/codecs/tlv320aic3x.h | 4 + 4 files changed, 99 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt index e7b98f4..f47c3f5 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic3x.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic3x.txt @@ -11,6 +11,12 @@ Optional properties: - gpio-reset - gpio pin number used for codec reset - ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality +- ai3x-micbias-vg - MicBias Voltage required. + 1 - MICBIAS output is powered to 2.0V, + 2 - MICBIAS output is powered to 2.5V, + 3 - MICBIAS output is connected to AVDD, + If this node is not mentioned or if the value is incorrect, then MicBias + is powered down. Example: diff --git a/include/sound/tlv320aic3x.h b/include/sound/tlv320aic3x.h index ffd9bc7..9407fd0 100644 --- a/include/sound/tlv320aic3x.h +++ b/include/sound/tlv320aic3x.h @@ -46,6 +46,13 @@ enum { AIC3X_GPIO2_FUNC_BUTTON_PRESS_IRQ = 15 }; +enum aic3x_micbias_voltage { + AIC3X_MICBIAS_OFF = 0, + AIC3X_MICBIAS_2_0V = 1, + AIC3X_MICBIAS_2_5V = 2, + AIC3X_MICBIAS_AVDDV = 3, +}; + struct aic3x_setup_data { unsigned int gpio_func[2]; }; @@ -53,6 +60,9 @@ struct aic3x_setup_data { struct aic3x_pdata { int gpio_reset; /* < 0 if not used */ struct aic3x_setup_data *setup; + + /* Selects the micbias voltage */ + enum aic3x_micbias_voltage micbias_vg; }; #endif diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 4989143..fe6dd27 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -85,6 +85,9 @@ struct aic3x_priv { #define AIC3X_MODEL_33 1 #define AIC3X_MODEL_3007 2 u16 model; + + /* Selects the micbias voltage */ + enum aic3x_micbias_voltage micbias_vg; }; /* @@ -195,6 +198,37 @@ static int snd_soc_dapm_put_volsw_aic3x(struct snd_kcontrol *kcontrol, return ret; } +/* + * mic bias power on/off share the same register bits with + * output voltage of mic bias. when power on mic bias, we + * need reclaim it to voltage value. + * 0x0 = Powered off + * 0x1 = MICBIAS output is powered to 2.0V, + * 0x2 = MICBIAS output is powered to 2.5V + * 0x3 = MICBIAS output is connected to AVDD + */ +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, + aic3x->micbias_vg << MICBIAS_LEVEL_SHIFT); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, 0); + break; + } + return 0; +} + static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; static const char *aic3x_left_hpcom_mux[] = @@ -596,12 +630,9 @@ static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = { AIC3X_ASD_INTF_CTRLA, 0, 3, 3, 0), /* Mic Bias */ - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2V", - MICBIAS_CTRL, 6, 3, 1, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias 2.5V", - MICBIAS_CTRL, 6, 3, 2, 0), - SND_SOC_DAPM_REG(snd_soc_dapm_micbias, "Mic Bias AVDD", - MICBIAS_CTRL, 6, 3, 3, 0), + SND_SOC_DAPM_SUPPLY("Mic Bias", MICBIAS_CTRL, 6, 0, + mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), /* Output mixers */ SND_SOC_DAPM_MIXER("Left Line Mixer", SND_SOC_NOPM, 0, 0, @@ -1386,6 +1417,24 @@ static int aic3x_probe(struct snd_soc_codec *codec) if (aic3x->model == AIC3X_MODEL_3007) snd_soc_add_codec_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); + /* set mic bias voltage */ + switch (aic3x->micbias_vg) { + case AIC3X_MICBIAS_2_0V: + case AIC3X_MICBIAS_2_5V: + case AIC3X_MICBIAS_AVDDV: + snd_soc_update_bits(codec, MICBIAS_CTRL, + MICBIAS_LEVEL_MASK, + (aic3x->micbias_vg) << MICBIAS_LEVEL_SHIFT); + break; + case AIC3X_MICBIAS_OFF: + /* + * noting to do. target won't enter here. This is just to avoid + * compile time warning "warning: enumeration value + * 'AIC3X_MICBIAS_OFF' not handled in switch" + */ + break; + } + aic3x_add_widgets(codec); list_add(&aic3x->list, &reset_list); @@ -1461,6 +1510,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, struct aic3x_setup_data *ai3x_setup; struct device_node *np = i2c->dev.of_node; int ret; + u32 value; aic3x = devm_kzalloc(&i2c->dev, sizeof(struct aic3x_priv), GFP_KERNEL); if (aic3x == NULL) { @@ -1474,6 +1524,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (pdata) { aic3x->gpio_reset = pdata->gpio_reset; aic3x->setup = pdata->setup; + aic3x->micbias_vg = pdata->micbias_vg; } else if (np) { ai3x_setup = devm_kzalloc(&i2c->dev, sizeof(*ai3x_setup), GFP_KERNEL); @@ -1493,6 +1544,28 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, aic3x->setup = ai3x_setup; } + if (!of_property_read_u32(np, "ai3x-micbias-vg", &value)) { + switch (value) { + case 1 : + aic3x->micbias_vg = AIC3X_MICBIAS_2_0V; + break; + case 2 : + aic3x->micbias_vg = AIC3X_MICBIAS_2_5V; + break; + case 3 : + aic3x->micbias_vg = AIC3X_MICBIAS_AVDDV; + break; + default : + aic3x->micbias_vg = AIC3X_MICBIAS_OFF; + dev_err(&i2c->dev, "Unsuitable MicBias voltage " + "found in DT\n"); + } + } else { + dev_warn(&i2c->dev, "No MicBias voltage found in DT - " + "using default\n"); + aic3x->micbias_vg = AIC3X_MICBIAS_OFF; + } + } else { aic3x->gpio_reset = -1; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6db3c41..e521ac3 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -238,6 +238,10 @@ /* Default input volume */ #define DEFAULT_GAIN 0x20 +/* MICBIAS Control Register */ +#define MICBIAS_LEVEL_SHIFT (6) +#define MICBIAS_LEVEL_MASK (3 << 6) + /* headset detection / button API */ /* The AIC3x supports detection of stereo headsets (GND + left + right signal) -- 1.7.9.5

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[alsa-devel] [PATCH] Asoc: Samsung: I2S: Add support for runtime S2R
by R. Chandrasekar 31 Jan '13

31 Jan '13

From: "R. Chandrasekar" <rcsekar(a)samsung.com> This patch adds runtime suspend to resume support for I2S. I2S clk is disabled at suspend and enabled at resume. Signed-off-by: R. Chandrasekar <rcsekar(a)samsung.com> Signed-off-by: Padmavathi Venna <padma.v(a)samsung.com> --- sound/soc/samsung/i2s.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 2fc42f9..d7231e3 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1071,6 +1071,26 @@ static inline int samsung_i2s_get_driver_data(struct platform_device *pdev) return platform_get_device_id(pdev)->driver_data; } +#ifdef CONFIG_PM_RUNTIME +static int i2s_runtime_suspend(struct device *dev) +{ + struct i2s_dai *i2s = dev_get_drvdata(dev); + + clk_disable_unprepare(i2s->clk); + + return 0; +} + +static int i2s_runtime_resume(struct device *dev) +{ + struct i2s_dai *i2s = dev_get_drvdata(dev); + + clk_prepare_enable(i2s->clk); + + return 0; +} +#endif /* CONFIG_PM_RUNTIME */ + static int samsung_i2s_probe(struct platform_device *pdev) { struct i2s_dai *pri_dai, *sec_dai = NULL; @@ -1288,6 +1308,11 @@ static const struct of_device_id exynos_i2s_match[] = { MODULE_DEVICE_TABLE(of, exynos_i2s_match); #endif +static const struct dev_pm_ops samsung_i2s_pm = { + SET_RUNTIME_PM_OPS(i2s_runtime_suspend, + i2s_runtime_resume, NULL) +}; + static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, .remove = samsung_i2s_remove, @@ -1296,6 +1321,7 @@ static struct platform_driver samsung_i2s_driver = { .name = "samsung-i2s", .owner = THIS_MODULE, .of_match_table = of_match_ptr(exynos_i2s_match), + .pm = &samsung_i2s_pm, }, }; -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: fsl: Fix multiple symbol definitions
by Thierry Reding 31 Jan '13

31 Jan '13

Commit 25b8d31 (ASoC: fsl: fix multiple definition of init_module) fixed a build error due to multiple symbol definitions when building as a module. However, it causes the build to break when the driver is builtin because the imx-pcm.o object is included multiple times. Solve the issue by adding imx-pcm.o to each of the modules that require it, but only add it to the kernel once when one or both drivers are builtin. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- sound/soc/fsl/Makefile | 21 ++++++++++++--------- 1 file changed, 12 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index ec14579..8446d49 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -30,21 +30,24 @@ obj-$(CONFIG_SND_MPC52xx_SOC_EFIKA) += efika-audio-fabric.o # i.MX Platform Support snd-soc-imx-ssi-objs := imx-ssi.o snd-soc-imx-audmux-objs := imx-audmux.o -snd-soc-imx-pcm-objs := imx-pcm.o -ifneq ($(CONFIG_SND_SOC_IMX_PCM_FIQ),) - snd-soc-imx-pcm-objs += imx-pcm-fiq.o -endif -ifneq ($(CONFIG_SND_SOC_IMX_PCM_DMA),) - snd-soc-imx-pcm-objs += imx-pcm-dma.o -endif obj-$(CONFIG_SND_SOC_IMX_SSI) += snd-soc-imx-ssi.o obj-$(CONFIG_SND_SOC_IMX_AUDMUX) += snd-soc-imx-audmux.o obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += snd-soc-imx-pcm-fiq.o -snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o imx-pcm.o +snd-soc-imx-pcm-fiq-y := imx-pcm-fiq.o +ifeq ($(CONFIG_SND_SOC_IMX_PCM_FIQ),m) +snd-soc-imx-pcm-fiq-y += imx-pcm.o +else +obj-$(CONFIG_SND_SOC_IMX_PCM_FIQ) += imx-pcm.o +endif obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += snd-soc-imx-pcm-dma.o -snd-soc-imx-pcm-dma-y := imx-pcm-dma.o imx-pcm.o +snd-soc-imx-pcm-dma-y := imx-pcm-dma.o +ifeq ($(CONFIG_SND_SOC_IMX_PCM_DMA),m) +snd-soc-imx-pcm-dma-y := imx-pcm.o +else +obj-$(CONFIG_SND_SOC_IMX_PCM_DMA) += imx-pcm.o +endif # i.MX Machine Support snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o -- 1.8.1.2

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[alsa-devel] volume adjust doesn't work after calling many times
by Keyon.Jey 31 Jan '13

31 Jan '13

hi, all, after I use UI interface with pavucontrol to tune output devices volume rapidly many times, the volume adjust seems doesn't work! error occur in dmesg: [ 568.807025] ALSA sound/pci/hda/hda_intel.c:716: azx_get_response timeout, switching to polling mode: last cmd=0x001f000a [ 569.808026] ALSA sound/pci/hda/hda_intel.c:755: hda_intel: azx_get_response timeout, switching to single_cmd mode: last cmd=0x001f000a realtek ALC888 codec. alsa version 1.0.23 attached the alsa-info output. any comment? BR, Keyon

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Re: [alsa-devel] [PATCH 3/4] ASOC: imx: add machine driver using wm8962 codec
by Mark Brown 31 Jan '13

31 Jan '13

On Thu, Jan 31, 2013 at 03:02:21AM +0000, Zhang Quan-B13634 wrote: > > > + if (!codec_dai->active) > > > + clk_enable(data->codec_clk); > > Should be clk_prepare_enable() but it's not clear to me why you need > > this... Also the clock API does refcounting so there should be no need > > to check for active, you should get matching startups and shutdowns. > > > > This will also fail for analogue bypass paths, set_bias_level() would be a good > > place to cover those. > [Gary-b13634] for power saving consideration. For now remove this feature to > Implement a basic feature I get why you're doing this at all but doing it in startup seems odd - like I say set_bias_level() is the most idiomatic place to implement this usually. There's a few machine drivers doing that you should be able to refer to for examples.

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Re: [alsa-devel] Sound/ARM/i.MX/SSI: fix SDMA starvation
by Shawn Guo 31 Jan '13

31 Jan '13

Thanks for the fix, Juergen. The patch subject could be written in a better form with a proper prefix added. "git log sound/soc/fsl/fsl_ssi.c" should give example. To have the patch get in sooner, you should copy Mark Brown on it. Also MAINTAINERS file gives the following, so you still do not have the right list copied. SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC) M: Liam Girdwood <lrg(a)ti.com> M: Mark Brown <broonie(a)opensource.wolfsonmicro.com> T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://alsa-project.org/main/index.php/ASoC S: Supported F: sound/soc/ F: include/sound/soc* On Wed, Jan 30, 2013 at 02:44:25PM +0100, Juergen Beisert wrote: > Juergen Beisert wrote: > > In cases when capturing is already running and someone enables also > > playback, the SDMA unit of the i.MX SoC does not see an edge on its TX DMA > > request line. This is due to the TX FIFOs are empty, and this request line > > is active all the time, when the CCSR_SSI_SIER_TDMAE bit is set. > > In this case also the TX FIFO underrun interrupt will flood the system > > forever. > > > > To ensure the first edge happens after enabling the TX side of the SSI unit > > and to trigger the SDMA unit successfully enable the CCSR_SSI_SIER_TDMAE on > > demand only. To be sure, this patch does the same for the > > CCSR_SSI_SIER_RDMAE (RX side) bit. > > > > Signed-off-by: Juergen Beisert <jbe(a)pengutronix.de> I can see the patch does fix the problem. A small comment below, otherwise Acked-by: Shawn Guo <shawn.guo(a)linaro.org> > > > > --- > > sound/soc/fsl/fsl_ssi.c | 24 ++++++++++++++++-------- > > 1 file changed, 16 insertions(+), 8 deletions(-) > > > > Index: linux.git/sound/soc/fsl/fsl_ssi.c > > =================================================================== > > --- linux.git.orig/sound/soc/fsl/fsl_ssi.c > > +++ linux.git/sound/soc/fsl/fsl_ssi.c > > @@ -86,10 +86,9 @@ static inline void write_ssi_mask(u32 __ > > #endif > > > > /* SIER bitflag of interrupts to enable */ > > -#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ > > - CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ > > - CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ > > - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ > > +#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TIE | \ > > + CCSR_SSI_SIER_TUE0_EN | CCSR_SSI_SIER_TUE1_EN | \ > > + CCSR_SSI_SIER_RFRC_EN | CCSR_SSI_SIER_RIE | \ It takes me sometime to see the real change here is to remove flags CCSR_SSI_SIER_TDMAE and CCSR_SSI_SIER_RDMAE. Then the changes below should look less dramatic and ease the reviewers a little bit, I think. Shawn --8<--- @@ -86,10 +86,9 @@ static inline void write_ssi_mask(u32 __iomem *addr, u32 clear, u32 set) #endif /* SIER bitflag of interrupts to enable */ -#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_TDMAE | \ +#define SIER_FLAGS (CCSR_SSI_SIER_TFRC_EN | CCSR_SSI_SIER_RIE | \ CCSR_SSI_SIER_TIE | CCSR_SSI_SIER_TUE0_EN | \ CCSR_SSI_SIER_TUE1_EN | CCSR_SSI_SIER_RFRC_EN | \ - CCSR_SSI_SIER_RDMAE | CCSR_SSI_SIER_RIE | \ CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) > > CCSR_SSI_SIER_ROE0_EN | CCSR_SSI_SIER_ROE1_EN) > > > > /** > > @@ -489,6 +488,9 @@ static int fsl_ssi_hw_params(struct snd_ > > * > > * The DMA channel is in external master start and pause mode, which > > * means the SSI completely controls the flow of data. > > + * > > + * To use the i.MX's SDMA unit it is important to create an edge on the > > + * corresponding request line. Otherwise the request will be ignored! > > */ > > static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, > > struct snd_soc_dai *dai) > > @@ -500,20 +502,26 @@ static int fsl_ssi_trigger(struct snd_pc > > switch (cmd) { > > case SNDRV_PCM_TRIGGER_START: > > case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > > - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) > > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > > write_ssi_mask(&ssi->scr, 0, > > CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); > > - else > > + write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_TDMAE); > > + } else { > > write_ssi_mask(&ssi->scr, 0, > > CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); > > + write_ssi_mask(&ssi->sier, 0, CCSR_SSI_SIER_RDMAE); > > + } > > break; > > > > case SNDRV_PCM_TRIGGER_STOP: > > case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > > - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) > > + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { > > write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_TE, 0); > > - else > > + write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_TDMAE, 0); > > + } else { > > write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); > > + write_ssi_mask(&ssi->sier, CCSR_SSI_SIER_RDMAE, 0); > > + } > > break; > > > > default: > > Any comments on this patch? > > jbe > > -- > Pengutronix e.K. | Juergen Beisert | > Linux Solutions for Science and Industry | Phone: +49-5121-206917-5128 | > Peiner Str. 6-8, 31137 Hildesheim, Germany | Fax: +49-5121-206917-5555 | > Amtsgericht Hildesheim, HRA 2686 | http://www.pengutronix.de/ | > > _______________________________________________ > linux-arm-kernel mailing list > linux-arm-kernel(a)lists.infradead.org > http://lists.infradead.org/mailman/listinfo/linux-arm-kernel -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] ASoC: wm2000: Expose ANC gain adjustment
by Mark Brown 30 Jan '13

30 Jan '13

No TLV information since it's not actually a direct gain control. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm2000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 4d809ab..f2ac38b 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -666,6 +666,7 @@ static int wm2000_speaker_put(struct snd_kcontrol *kcontrol, } static const struct snd_kcontrol_new wm2000_controls[] = { + SOC_SINGLE("ANC Volume", WM2000_REG_ANC_GAIN_CTRL, 0, 255, 0), SOC_SINGLE_BOOL_EXT("WM2000 ANC Switch", 0, wm2000_anc_mode_get, wm2000_anc_mode_put), @@ -778,6 +779,8 @@ static int wm2000_probe(struct snd_soc_codec *codec) { struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); + snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_REGMAP); + /* This will trigger a transition to standby mode by default */ wm2000_anc_set_mode(wm2000); -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Fix mono
by Benoît Thébaudeau 30 Jan '13

30 Jan '13

The TLV320AIC3x DAI supports only stereo. This patch fixes mono support for this codec by allowing only stereo at codec level. Tested with imx-ssi and TLV320AIC3104. Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: <alsa-devel(a)alsa-project.org> Signed-off-by: Benoît Thébaudeau <benoit.thebaudeau(a)advansee.com> --- sound/soc/codecs/tlv320aic3x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 5708a97..4989143 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1210,13 +1210,13 @@ static struct snd_soc_dai_driver aic3x_dai = { .name = "tlv320aic3x-hifi", .playback = { .stream_name = "Playback", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, .capture = { .stream_name = "Capture", - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = AIC3X_RATES, .formats = AIC3X_FORMATS,}, -- 1.7.10.4

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[alsa-devel] Thread safety of snd_pcm_open?
by Jérôme Forissier 29 Jan '13

29 Jan '13

Hello, I'm having problems using ALSA with LibVLC. The project page http://www.alsa-project.org/main/index.php/SMP_Design claims that the snd_*_open() calls are thread safe, yet it seems that when LibVLC calls them from several threads it can lead to errors like: ALSA lib conf.c:4571:(parse_args) Unknown parameter CARD ALSA lib conf.c:4571:(parse_args) Unknown parameter DEV ALSA lib conf.c:4704:(snd_config_expand) Parse arguments error: No such file or directory ALSA lib conf.c:4687:(snd_config_expand) Unknown parameters CARD=NVidia,DEV=0 ALSA lib conf.c:4571:(parse_args) Unknown parameter CARD ALSA lib conf.c:4687:(snd_config_expand) Unknown parameters CARD=PCH,DEV=0 ALSA lib conf.c:4687:(snd_config_expand) Unknown parameters CARD=NVidia,DEV=0 ALSA lib conf.c:4571:(parse_args) Unknown parameter DEV (Please see http://mailman.videolan.org/pipermail/vlc-devel/2013-January/091603.html and also http://mailman.videolan.org/pipermail/vlc-devel/2013-January/091633.html for a reproducer on top of LibVLC). I'm not familiar with the internals of ALSA lib, nor with the internals of LibVLC. The VLC devs claim that they respect the rules regarding ALSA access, but I can't tell for sure... I'm currently running my reproducer with valgrind, but before I dig too deeply, could you please give some advice? I have noticed that pcm_direct.c calls getgrnam() which is not reentrant (but replacing it with getgrnam_r is not enough to fix my problem). Also there does not seem to be any locking around snd_pcm_open[_noupdate]. Are those functions supposed to be reentrant? Thanks for your help, -- Jerome

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