October 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: remove component from snd_soc_pcm_runtime
by Kuninori Morimoto 24 Oct '16

24 Oct '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit f2ed6b07645e ("ASoC: Make aux_dev more like a generic component") removed its usecase. No one is using it now. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 091201a..cff9592 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1219,7 +1219,6 @@ struct snd_soc_pcm_runtime { struct snd_soc_platform *platform; struct snd_soc_dai *codec_dai; struct snd_soc_dai *cpu_dai; - struct snd_soc_component *component; /* Only valid for AUX dev rtds */ struct snd_soc_dai **codec_dais; unsigned int num_codecs; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: bunch up bit field for snd_soc_dai
by Kuninori Morimoto 24 Oct '16

24 Oct '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We can reduce struct size in certain environment. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc-dai.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 964b7de..d0ae96b 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -268,8 +268,9 @@ struct snd_soc_dai { unsigned int symmetric_rates:1; unsigned int symmetric_channels:1; unsigned int symmetric_samplebits:1; + unsigned int probed:1; + unsigned int active; - unsigned char probed:1; struct snd_soc_dapm_widget *playback_widget; struct snd_soc_dapm_widget *capture_widget; -- 1.9.1

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[alsa-devel] [PATCH] ASoC: bunch up bit field for snd_soc_pcm_runtime
by Kuninori Morimoto 24 Oct '16

24 Oct '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We can reduce struct size in certain environment. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index f9dc828..091201a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1206,14 +1206,11 @@ struct snd_soc_pcm_runtime { enum snd_soc_pcm_subclass pcm_subclass; struct snd_pcm_ops ops; - unsigned int dev_registered:1; - /* Dynamic PCM BE runtime data */ struct snd_soc_dpcm_runtime dpcm[2]; int fe_compr; long pmdown_time; - unsigned char pop_wait:1; /* runtime devices */ struct snd_pcm *pcm; @@ -1235,6 +1232,10 @@ struct snd_soc_pcm_runtime { unsigned int num; /* 0-based and monotonic increasing */ struct list_head list; /* rtd list of the soc card */ + + /* bit field */ + unsigned int dev_registered:1; + unsigned int pop_wait:1; }; /* mixer control */ -- 1.9.1

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[alsa-devel] [PATCH v2 0/6] DT: Fix spelling of standard properties
by Geert Uytterhoeven 24 Oct '16

24 Oct '16

Hi all, This patch series fixes misspellings of various standard DT properties in DT binding documentation, DTS files, and error messages. While most of these are harmless, some of them may cause hard-to-debug failures. Changes compared to v2: - Dropped patches that have been applied already, - Add Rob Herring's Acked-by, - Split off "[PATCH v2 4/6] ASoC: davinci-mcbsp: DT fix s/interrupts-names/interrupt-names/" from the unrelated "[PATCH 06/14] dmaengine: bcm2835: DT spelling s/interrupts-names/interrupt-names/", - Add new patch "[PATCH v2 3/6] arm64: dts: lg1313: DT fix s/#interrupts-cells/#interrupt-cells/". Please apply where appropriate. Thanks! P.S. I used the following to detect misspellings: words="(address|clock|cooling|dma|gpio|index|interrupt|mbox|msi|nvmem|phy|phys|power-domain|pwm|reset|size|sleep|sound-dai|thermal-sensor)" git grep -Ew "${words}s-names" git grep -E "[^-]\<${words}-name\>[^-]" git grep -Ew "#${words}s-cells" # false positive phys-cells git grep -E "#${words}-cell\>[^-]" git grep -w adress-cells git grep -Ew "interrupts-(map|parent)" Geert Uytterhoeven (6): ARM: dts: STiH407: DT fix s/interrupts-names/interrupt-names/ arm64: dts: lg1312: DT fix s/#interrupts-cells/#interrupt-cells/ arm64: dts: lg1313: DT fix s/#interrupts-cells/#interrupt-cells/ ASoC: davinci-mcbsp: DT fix s/interrupts-names/interrupt-names/ dt: booting-without-of: DT fix s/#interrupt-cell/#interrupt-cells/ powerpc: dts: acadia: DT fix s/#interrupts-parent/#interrupt-parent/ Documentation/devicetree/bindings/sound/davinci-mcbsp.txt | 2 +- Documentation/devicetree/booting-without-of.txt | 2 +- arch/arm/boot/dts/stih407-pinctrl.dtsi | 2 +- arch/arm64/boot/dts/lg/lg1312.dtsi | 2 +- arch/arm64/boot/dts/lg/lg1313.dtsi | 2 +- arch/powerpc/boot/dts/acadia.dts | 2 +- 6 files changed, 6 insertions(+), 6 deletions(-) -- 1.9.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[alsa-devel] [PATCH] ASoC: soc.h: use bit field for playback/capture_only
by Kuninori Morimoto 24 Oct '16

24 Oct '16

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current snd_soc_dai_link is already using many bit fields. Let's use it for playback_only/capture_only too. We can reduce struct size in certain environment. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/soc.h | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a1b65b2..f9dc828 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1029,13 +1029,13 @@ struct snd_soc_dai_link { const struct snd_soc_ops *ops; const struct snd_soc_compr_ops *compr_ops; - /* For unidirectional dai links */ - bool playback_only; - bool capture_only; - /* Mark this pcm with non atomic ops */ bool nonatomic; + /* For unidirectional dai links */ + unsigned int playback_only:1; + unsigned int capture_only:1; + /* Keep DAI active over suspend */ unsigned int ignore_suspend:1; -- 1.9.1

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[alsa-devel] [PATCH -next] ASoC: Intel: Atom: add terminate entry for dmi_system_id tables
by Wei Yongjun 23 Oct '16

23 Oct '16

From: Wei Yongjun <weiyongjun1(a)huawei.com> Make sure dmi_system_id tables are NULL terminated. Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index ba5c0d7..0a88537 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -416,6 +416,7 @@ static const struct dmi_system_id cht_table[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Surface 3"), }, }, + { } };

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[alsa-devel] [PATCH v2 1/1] sst: select DW_DMAC_CORE since it's mandatory
by Andy Shevchenko 22 Oct '16

22 Oct '16

Select DW_DMAC_CORE like the rest of glue drivers do, e.g. drivers/dma/dw/Kconfig. While here group selectors under SND_SOC_INTEL_HASWELL and SND_SOC_INTEL_BAYTRAIL. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/Kconfig | 20 ++++++++------------ 1 file changed, 8 insertions(+), 12 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index fd5d1e0..44d1cd4 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -47,16 +47,19 @@ config SND_SOC_INTEL_SST_MATCH config SND_SOC_INTEL_HASWELL tristate + select DW_DMAC_CORE + select SND_SOC_INTEL_SST select SND_SOC_INTEL_SST_FIRMWARE config SND_SOC_INTEL_BAYTRAIL tristate + select DW_DMAC_CORE + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SST_FIRMWARE config SND_SOC_INTEL_HASWELL_MACH tristate "ASoC Audio DSP support for Intel Haswell Lynxpoint" depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST select SND_SOC_INTEL_HASWELL select SND_SOC_RT5640 help @@ -99,9 +102,7 @@ config SND_SOC_INTEL_BXT_RT298_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "ASoC Audio driver for Intel Baytrail with RT5640 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE && (SND_SST_IPC_ACPI = n) - select SND_SOC_INTEL_SST - select SND_SOC_INTEL_SST_FIRMWARE + depends on SND_SST_IPC_ACPI = n select SND_SOC_INTEL_BAYTRAIL select SND_SOC_RT5640 help @@ -112,9 +113,7 @@ config SND_SOC_INTEL_BYT_RT5640_MACH config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "ASoC Audio driver for Intel Baytrail with MAX98090 codec" depends on X86_INTEL_LPSS && I2C - depends on DW_DMAC_CORE && (SND_SST_IPC_ACPI = n) - select SND_SOC_INTEL_SST - select SND_SOC_INTEL_SST_FIRMWARE + depends on SND_SST_IPC_ACPI = n select SND_SOC_INTEL_BAYTRAIL select SND_SOC_MAX98090 help @@ -134,10 +133,7 @@ config SND_SOC_INTEL_BDW_RT5677_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on X86_INTEL_LPSS && I2C && DW_DMAC && \ - I2C_DESIGNWARE_PLATFORM - depends on DW_DMAC_CORE - select SND_SOC_INTEL_SST + depends on X86_INTEL_LPSS && I2C && I2C_DESIGNWARE_PLATFORM select SND_SOC_INTEL_HASWELL select SND_SOC_RT286 help -- 2.9.3

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Re: [alsa-devel] M-audio Audiophile Firewire
by Takashi Sakamoto 22 Oct '16

22 Oct '16

Hi, (CCed to alsa-devel, ffado-devel) On Oct 22 2016 07:37, Peti Pap wrote: > Do you have any advice about this soundcard? I own the unit and work with it on Linux system. > .... I tried use on Linux Mint (Sarah) 64 bit, and I installed ffado packs, > and jack too... but still only the blue led lighting, and no any sound. (On > the Jack "panel" I tried all of the possibble settings) Please show your environment; at least, kernel version and libffado2. Regards Takashi Sakamoto

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[alsa-devel] [PATCH 0/7] ALSA: Add rewinds disabled, delays, max_inflight_bytes
by Subhransu S. Prusty 21 Oct '16

21 Oct '16

Set of patches to fix issues with delays, hw_ptr fuzziness [1] and increased buffering w/ DSPs 1. Rewinds can be disabled when data written in ring buffer will never be validated. This allow for new HDaudio SPIB DMA functionality(allow fetch up to the application pointer, no rewind supported) 2. Report max in-flight bytes to avoid problems with stale data (like late wake-ups, rewinds) 3. add new estimate for USB startup delay. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093646.html Pierre-Louis Bossart (6): ALSA: core: let low-level driver or userspace disable rewinds ALSA: core: add .update_appl_ptr callback for pcm ops ALSA: core: add report of max inflight bytes ALSA: hda: add default value for max_inflight_bytes ALSA: usb: no_period_wake and max_inflight_bytes report ALSA: usb: take startup delay into account Ramesh Babu (1): ALSA: pcm: avoid mmap of control data if .update_appl_ptr is implemented include/sound/pcm.h | 4 +++ include/uapi/sound/asound.h | 6 ++-- sound/core/pcm_lib.c | 26 ++++++++++++++++++ sound/core/pcm_native.c | 51 +++++++++++++++++++++++++++++++++- sound/pci/hda/hda_controller.c | 1 + sound/usb/card.h | 1 + sound/usb/pcm.c | 62 +++++++++++++++++++++++++++++++++++++----- 7 files changed, 141 insertions(+), 10 deletions(-) -- 1.9.1

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[alsa-devel] how to disable dynamic control of mic bias?
by Tim Harvey 21 Oct '16

21 Oct '16

I have a board design with a TLV320AIC3x codec + TS3A227E headset detect chip and am using the tlv320aic3x and ts3a227e drivers in mainline Linux. The TS3A227E serves a couple of purposes including support for multiple headset pinout standards and sensing buttons on the mic (ie see Android wired headset spec: https://source.android.com/devices/accessories/headset/specification.html). In order for the button detect to work mic bias from the codec needs to be enabled (at init time and regardless of playing or recording audio). What is the correct way to do this in either device-tree if using something like simple-audio-card or in a machine driver? I have not had success by setting the codec drivers 'idle_bias_off = false' - while this does keep mic bias on at init and after audio playback it kills it after record due to the 'Mic Bias' dapm widget in the case of the tlv320aic3x.c driver. I'm thinking I need to instead remove the routing or disable the 'Mic Bias' but I'm not clear how to do this? Regards, Tim

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