May 2015 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] [PATCH v2 2/4] Documentation: devicetree: sound: Add documentation for gtm601 codec
by Mark Brown 07 May '15

07 May '15

On Wed, May 06, 2015 at 10:38:48PM +0200, Marek Belisko wrote: > Add small documentation for gtm601 UMTS modem audio interface. git am complains that this patch is corrupt so I can't apply it. Please also use subject lines reflecting the style for the subsystem.

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[alsa-devel] [PATCH 1/2] ASoC: dapm: Break out of widget search when source and sink are located
by Charles Keepax 07 May '15

07 May '15

Currently snd_soc_dapm_add_route will continue to search the widget list even after both the source and sink for the route have been located. This patch breaks out of the search when both are located giving a small improvement in probe time for drivers. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Note these patches depend on: commit 92fa12426741d52b39ec92ad77c9843d3fc2b3d6 Author: Lars-Peter Clausen <lars(a)metafoo.de> Date: Fri May 1 18:02:42 2015 +0200 ASoC: dapm: Add new widgets to the end of the widget list Thanks, Charles sound/soc/soc-dapm.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a2e5f22..ea3348e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2617,14 +2617,20 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, list_for_each_entry(w, &dapm->card->widgets, list) { if (!wsink && !(strcmp(w->name, sink))) { wtsink = w; - if (w->dapm == dapm) + if (w->dapm == dapm) { wsink = w; + if (wsource) + break; + } continue; } if (!wsource && !(strcmp(w->name, source))) { wtsource = w; - if (w->dapm == dapm) + if (w->dapm == dapm) { wsource = w; + if (wsink) + break; + } } } /* use widget from another DAPM context if not found from this */ -- 1.7.2.5

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[alsa-devel] Alsa triggering methods
by Andy Ng 07 May '15

07 May '15

Hi, I am working on a ALSA card driver that handles a DSP instance. The problem I have is that the triggering mechanism in ALSA to start a PCM has the interrupts disabled, and my dsp framework is actually acknowledging the start trigger with an event after 50 microseconds time that I can't handle because the IRQs are disabled. Is there any way when the trigger mechanism goes down to the card to be able to handle any acknowledgements what will happen within a sort period of time. I have seen bespoke triggering method but not sure if everyone is using it. Any ideas would be helpful. Best regards, Andreas

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Re: [alsa-devel] [PATCH v2 RESEND] ARM: dts: Support audio on Exynos5422-odroidxu3 using simple-audio-card
by Inha Song 07 May '15

07 May '15

Hi Kukjin, Any comments about this patch? Best Regards, Inha Song. On Fri, 10 Apr 2015 14:48:13 +0530 Anand Moon <linux.amoon(a)gmail.com> wrote: > Hi Inha, > > Tested-by: Anand Moon <linux.amoon(a)gmail.com> > > -Anand Moon > > On 10 April 2015 at 14:32, Markus Reichl <m.reichl(a)fivetechno.de> wrote: > > Hi Inha, > > > > I have tested this on XU3 (Headphone output). > > > > Tested-by: Markus Reichl <m.reichl(a)fivetechno.de> > > > > Best Regards > > -- > > Markus Reichl > > > > Am 10.04.2015 um 09:32 schrieb Inha Song: > >> Add MAX98090 audio codec, I2S interface and the sound nodes to support > >> audio on Exynos5422 SoC Based Odroid-XU3 board. Now we can support audio > >> in Odroid-XU3 board using simple-audio-card DT binding. > >> > >> Signed-off-by: Inha Song <ideal.song(a)samsung.com> > >> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> > >> --- > >> arch/arm/boot/dts/exynos5420.dtsi | 9 +++++ > >> arch/arm/boot/dts/exynos5422-odroidxu3.dts | 59 ++++++++++++++++++++++++++++++ > >> 2 files changed, 68 insertions(+) > >> > >> diff --git a/arch/arm/boot/dts/exynos5420.dtsi b/arch/arm/boot/dts/exynos5420.dtsi > >> index ac6f860..73886d6 100644 > >> --- a/arch/arm/boot/dts/exynos5420.dtsi > >> +++ b/arch/arm/boot/dts/exynos5420.dtsi > >> @@ -415,6 +415,9 @@ > >> <&clock_audss EXYNOS_I2S_BUS>, > >> <&clock_audss EXYNOS_SCLK_I2S>; > >> clock-names = "iis", "i2s_opclk0", "i2s_opclk1"; > >> + #clock-cells = <1>; > >> + clock-output-names = "i2s_cdclk0"; > >> + #sound-dai-cells = <1>; > >> samsung,idma-addr = <0x03000000>; > >> pinctrl-names = "default"; > >> pinctrl-0 = <&i2s0_bus>; > >> @@ -429,6 +432,9 @@ > >> dma-names = "tx", "rx"; > >> clocks = <&clock CLK_I2S1>, <&clock CLK_SCLK_I2S1>; > >> clock-names = "iis", "i2s_opclk0"; > >> + #clock-cells = <1>; > >> + clock-output-names = "i2s_cdclk1"; > >> + #sound-dai-cells = <1>; > >> pinctrl-names = "default"; > >> pinctrl-0 = <&i2s1_bus>; > >> status = "disabled"; > >> @@ -442,6 +448,9 @@ > >> dma-names = "tx", "rx"; > >> clocks = <&clock CLK_I2S2>, <&clock CLK_SCLK_I2S2>; > >> clock-names = "iis", "i2s_opclk0"; > >> + #clock-cells = <1>; > >> + clock-output-names = "i2s_cdclk2"; > >> + #sound-dai-cells = <1>; > >> pinctrl-names = "default"; > >> pinctrl-0 = <&i2s2_bus>; > >> status = "disabled"; > >> diff --git a/arch/arm/boot/dts/exynos5422-odroidxu3.dts b/arch/arm/boot/dts/exynos5422-odroidxu3.dts > >> index edc25cf..c038f0d 100644 > >> --- a/arch/arm/boot/dts/exynos5422-odroidxu3.dts > >> +++ b/arch/arm/boot/dts/exynos5422-odroidxu3.dts > >> @@ -11,6 +11,7 @@ > >> */ > >> > >> /dts-v1/; > >> +#include <dt-bindings/sound/samsung-i2s.h> > >> #include "exynos5800.dtsi" > >> > >> / { > >> @@ -285,6 +286,64 @@ > >> rtc@101E0000 { > >> status = "okay"; > >> }; > >> + > >> + sound: sound { > >> + compatible = "simple-audio-card"; > >> + > >> + simple-audio-card,name = "Odroid-XU3"; > >> + simple-audio-card,widgets = > >> + "Headphone", "Headphone Jack", > >> + "Speakers", "Speakers"; > >> + simple-audio-card,routing = > >> + "Headphone Jack", "HPL", > >> + "Headphone Jack", "HPR", > >> + "Headphone Jack", "MICBIAS", > >> + "IN1", "Headphone Jack", > >> + "Speakers", "SPKL", > >> + "Speakers", "SPKR"; > >> + > >> + simple-audio-card,format = "i2s"; > >> + simple-audio-card,bitclock-master = <&link0_codec>; > >> + simple-audio-card,frame-master = <&link0_codec>; > >> + > >> + simple-audio-card,cpu { > >> + sound-dai = <&i2s0 0>; > >> + system-clock-frequency = <19200000>; > >> + }; > >> + > >> + link0_codec: simple-audio-card,codec { > >> + sound-dai = <&max98090>; > >> + clocks = <&i2s0 CLK_I2S_CDCLK>; > >> + }; > >> + }; > >> +}; > >> + > >> +&clock_audss { > >> + assigned-clocks = <&clock_audss EXYNOS_MOUT_AUDSS>, > >> + <&clock_audss EXYNOS_MOUT_I2S>, > >> + <&clock_audss EXYNOS_DOUT_AUD_BUS>; > >> + assigned-clock-parents = <&clock CLK_FIN_PLL>, > >> + <&clock_audss EXYNOS_MOUT_AUDSS>; > >> + assigned-clock-rates = <0>, > >> + <0>, > >> + <19200000>; > >> +}; > >> + > >> +&hsi2c_5 { > >> + status = "okay"; > >> + max98090: max98090@10 { > >> + compatible = "maxim,max98090"; > >> + reg = <0x10>; > >> + interrupt-parent = <&gpx3>; > >> + interrupts = <2 0>; > >> + clocks = <&i2s0 CLK_I2S_CDCLK>; > >> + clock-names = "mclk"; > >> + #sound-dai-cells = <0>; > >> + }; > >> +}; > >> + > >> +&i2s0 { > >> + status = "okay"; > >> }; > >> > >> &hdmi { > >> > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-samsung-soc" 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 0/5] ASoC: Intel: let machines configure SSP configuration
by Vinod Koul 06 May '15

06 May '15

This series first add frame and data polarity to ssp config and then starts to use local values for ssp config and then eventually lets that be loaded by hw_params and updates by pcm format and tdm calls to allow machine t specify these parameters Vinod Koul (5): ASoC: Intel: add frame and data polarity to ssp config ASoC: Intel: use local values for ssp configuration ASoC: Intel: load hw_defaults in hw_params of ssp be ASoC: intel: add support for specifying PCM format ASoC: Intel: add support for configuring TDM slots for SSP sound/soc/intel/atom/sst-atom-controls.c | 167 +++++++++++++++++++++++---- sound/soc/intel/atom/sst-atom-controls.h | 7 ++ sound/soc/intel/atom/sst-mfld-platform-pcm.c | 47 +++++++- sound/soc/intel/atom/sst-mfld-platform.h | 2 + 4 files changed, 197 insertions(+), 26 deletions(-) -- 1.9.1

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Re: [alsa-devel] Front speakers doesn't work in multichannel output, regression in ALC888
by Raymond Yau 06 May '15

06 May '15

> > > > > > My problem is that when I set channel output to "Analog output > > > > surround > > > > > > > 5.1" > > > > > (or 2.1 or 4.1), I hear no sound in front speakers, but in case of > > > > 5.1 or > > > > > > > 4.1 > > > > You need pulseaudio lfe-filter for 2.1 > > > > http://cgit.freedesktop.org/pulseaudio/pulseaudio/log/?qt=grep&q=lfe-filter > > External 5.1 and 4.1 worked fine without lfe-filter. Fedora doesn't package lfe- > filter and I'm more interested to get my 5.1 configuration back. > > > do you mean surround21 need 6 channel instead of channels with 5.1 internal > > /external speaker ? > > > > http://git.alsa-project.org/?p=alsa-lib.git;a=commitdiff;h=48f1b308cc66152eb > > 6db66742dd0d08d888cda8d;hp=5c4cd46810cef8850b037fca9e38ffd43b0bff22 > > No. When in KDE channel output is set to 2.1 it makes no difference what is set > in alsamixer. When my external speakers are disconnected built-in front left > and front right works on my laptop, but built-in subwoofer doesn't work There > is no difference what channel output is set in alsamixer. > > My laptop has built-in front left, front right, rear left, rear right and > subwoofer and something is strange. > When I set channel output to 5.1 or 4.1 and external speakers are > disconnected, then front left and front right work as expected but built-in > subwoofer is heard only if I test rear left channel (wrong channel mapping?) https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… Seem only two pairs of stereo speakers and internal subwoofer at left channel before this patch ALSA: hda - Enable surround speakers (when line out is also present)

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[alsa-devel] [PATCH] ASoC: omap-twl4030: Use card DAPM context to access widgets
by Lars-Peter Clausen 06 May '15

06 May '15

Use the card DAPM context instead of the CODEC DAPM context since only card level widgets are accessed here. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/omap/omap-twl4030.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index 3673ada..7431314 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -159,9 +159,8 @@ static inline void twl4030_disconnect_pin(struct snd_soc_dapm_context *dapm, static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = rtd->card; - struct snd_soc_dapm_context *dapm = &codec->dapm; + struct snd_soc_dapm_context *dapm = &card->dapm; struct omap_tw4030_pdata *pdata = dev_get_platdata(card->dev); struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card); int ret = 0; -- 1.8.0

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[alsa-devel] [PATCH v3 0/7] ASoC: intel - add skylake PCM driver
by Vinod Koul 06 May '15

06 May '15

This patch series adds the hda codec wrapper first followed by asoc hda controller library, then SKL PCM driver and last decouples the controller for splitting the links Jeeja KP (7): ASoC: hda - add soc hda codec driver wrapper ALSA: hda - add new HDA registers ASoC: hda - adds SoC controller and stream operations ASoC: intel - add Skylake HDA platform driver ASoC: intel - add Skylake HDA audio driver ASoC: intel - add makefile support for SKL driver ASoC: intel - adds support for decoupled mode in skl driver include/sound/hda_register.h | 87 +++ include/sound/soc-hdaudio.h | 343 ++++++++++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/hda/Kconfig | 3 + sound/soc/hda/Makefile | 3 + sound/soc/hda/soc-hda-codec.c | 85 +++ sound/soc/hda/soc-hda-codec.h | 44 ++ sound/soc/hda/soc-hdac-controller.c | 289 ++++++++++ sound/soc/hda/soc-hdac-stream.c | 390 ++++++++++++++ sound/soc/intel/Kconfig | 20 + sound/soc/intel/Makefile | 1 + sound/soc/intel/skylake/Makefile | 3 + sound/soc/intel/skylake/hda-skl-pcm.c | 933 +++++++++++++++++++++++++++++++++ sound/soc/intel/skylake/hda-skl.c | 743 ++++++++++++++++++++++++++ sound/soc/intel/skylake/hda-skl.h | 74 +++ 16 files changed, 3020 insertions(+) create mode 100644 include/sound/soc-hdaudio.h create mode 100644 sound/soc/hda/Kconfig create mode 100644 sound/soc/hda/Makefile create mode 100644 sound/soc/hda/soc-hda-codec.c create mode 100644 sound/soc/hda/soc-hda-codec.h create mode 100644 sound/soc/hda/soc-hdac-controller.c create mode 100644 sound/soc/hda/soc-hdac-stream.c create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/hda-skl-pcm.c create mode 100644 sound/soc/intel/skylake/hda-skl.c create mode 100644 sound/soc/intel/skylake/hda-skl.h -- 1.7.9.5

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[alsa-devel] [PATCH 1/4] ASoC: dapm: Add new widgets to the end of the widget list
by Lars-Peter Clausen 06 May '15

06 May '15

Currently new widgets are appended to the beginning of the cards widget list. This has the effect that widgets that are created while iterating over the widget list in snd_soc_dapm_new_widgets() (like e.g. the auto-disable widgets) are not covered during that invocation of the function. If no further invocations of snd_soc_dapm_new_widgets() happen these widgets will not be fully initialized and e.g. no debugfs entries are created for them. By adding new widgets to the end of the widget list we make sure that widgets that are created in snd_soc_dapm_new_widgets() will still be handled during the same snd_soc_dapm_new_widgets() invocation and are always fully initialized. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- This series depends on Charles' mux auto-disable series. --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 79e6cf4..5c159f4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3252,7 +3252,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&w->sinks); INIT_LIST_HEAD(&w->list); INIT_LIST_HEAD(&w->dirty); - list_add(&w->list, &dapm->card->widgets); + list_add_tail(&w->list, &dapm->card->widgets); w->inputs = -1; w->outputs = -1; -- 1.8.0

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[alsa-devel] [PATCH v3 1/5] ASoC: Correct typo in SOC_VALUE_ENUM_SINGLE macro
by Charles Keepax 06 May '15

06 May '15

xnitmes is clearly intended to be xnitems, but all other macros just refer to this as xitems, so change it to that. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Use snd_soc_enum_item_to_val to get zero value from mux to account for non-value enums - Improve handling in snd_soc_dapm_put_enum_double of the situation where the register and widget views of the control values don't match Thanks, Charles include/sound/soc.h | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a73d855..45cfd69 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -190,8 +190,8 @@ #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} -#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ - SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xitems, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xitems, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ -- 1.7.2.5

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