October 2020 - Alsa-devel - mailman.alsa-project.org

alsamixer: add "restore defaults" keystroke
by GitHub issues - opened 18 Oct '20

18 Oct '20

alsa-project/alsa-utils issue #64 was opened from jidanni: There should be a way to "restore defaults" for all levels for this sound card. And perhaps for each level individually too. Issue URL : https://github.com/alsa-project/alsa-utils/issues/64 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] ALSA: hda/ca0132: make some const arrays static, makes object smaller
by Colin King 17 Oct '20

17 Oct '20

From: Colin Ian King <colin.king(a)canonical.com> Don't populate const arrays on the stack but instead make them static. Makes the object code smaller by 57 bytes. Before: text data bss dec hex filename 173256 38016 192 211464 33a08 sound/pci/hda/patch_ca0132.o After: text data bss dec hex filename 172879 38336 192 211407 339cf sound/pci/hda/patch_ca0132.o (gcc version 10.2.0) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/hda/patch_ca0132.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 2b38b2a716a1..e0c38f2735c6 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -7910,8 +7910,12 @@ static void ae7_post_dsp_asi_stream_setup(struct hda_codec *codec) static void ae7_post_dsp_pll_setup(struct hda_codec *codec) { - const unsigned int addr[] = { 0x41, 0x45, 0x40, 0x43, 0x51 }; - const unsigned int data[] = { 0xc8, 0xcc, 0xcb, 0xc7, 0x8d }; + static const unsigned int addr[] = { + 0x41, 0x45, 0x40, 0x43, 0x51 + }; + static const unsigned int data[] = { + 0xc8, 0xcc, 0xcb, 0xc7, 0x8d + }; unsigned int i; for (i = 0; i < ARRAY_SIZE(addr); i++) { @@ -7925,10 +7929,12 @@ static void ae7_post_dsp_pll_setup(struct hda_codec *codec) static void ae7_post_dsp_asi_setup_ports(struct hda_codec *codec) { struct ca0132_spec *spec = codec->spec; - const unsigned int target[] = { 0x0b, 0x04, 0x06, 0x0a, 0x0c, 0x11, - 0x12, 0x13, 0x14 }; - const unsigned int data[] = { 0x12, 0x00, 0x48, 0x05, 0x5f, 0xff, - 0xff, 0xff, 0x7f }; + static const unsigned int target[] = { + 0x0b, 0x04, 0x06, 0x0a, 0x0c, 0x11, 0x12, 0x13, 0x14 + }; + static const unsigned int data[] = { + 0x12, 0x00, 0x48, 0x05, 0x5f, 0xff, 0xff, 0xff, 0x7f + }; unsigned int i; mutex_lock(&spec->chipio_mutex); -- 2.27.0

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[PATCH v2] sound: sparc: dbri: fix repeated word 'the'
by Randy Dunlap 17 Oct '20

17 Oct '20

Change the duplicated word "the" to "Then the". Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: Joe Perches <joe(a)perches.com> --- v2: use "Then the" instead of just "The". (Joe Perches) sound/sparc/dbri.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20201016.orig/sound/sparc/dbri.c +++ linux-next-20201016/sound/sparc/dbri.c @@ -620,7 +620,7 @@ A circular command buffer is used here. while another can be executed. The scheme works by adding two WAIT commands after each sent batch of commands. When the next batch is prepared it is added after the WAIT commands then the WAITs are replaced with single JUMP -command to the new batch. The the DBRI is forced to reread the last WAIT +command to the new batch. Then the DBRI is forced to reread the last WAIT command (replaced by the JUMP by then). If the DBRI is still executing previous commands the request to reread the WAIT command is ignored.

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Re: [PATCH] ASoC: Intel: boards: Add CML_RT1015 m/c driver
by Mark Brown 16 Oct '20

16 Oct '20

On Sat, Oct 17, 2020 at 12:02:24AM +0800, matsufan(a)gmail.com wrote: > +<<<<<<< HEAD > obj-$(CONFIG_SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH) += snd-soc-cml_rt1011_rt5682.o > +======= > +obj-$(CONFIG_SND_SOC_INTEL_SOF_CML_RT1011_RT5682_MACH) += cml_rt1011_rt5682.o > +obj-$(CONFIG_SND_SOC_INTEL_SOF_CML_RT1015_RT5682_MACH) += cml_rt1015_rt5682.o > +>>>>>>> 955c356... CHROMIUM: ASoC: Intel: boards: Add CML_RT1015 m/c driver There's an unresolved conflict here :/

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Discover a microphone device, to later discover if it is receiving input
by info 16 Oct '20

16 Oct '20

Hi all, I am writing a program that aims to auto-discover the microphone device that the user is speaking into. I started off by querying device hints and collected all devices with the IO types of null, since I have not found any devices with the type of Input. I am puzzled how it is possible that there is no Input, where a microphone device should not be able to emit sound. This is the list on my particular machine: Name of device: sysdefault:CARD=SB Description of device: HDA ATI SB, ALC892 Analog Default Audio Device I/O type of device: (null) Name of device: front:CARD=SB,DEV=0 Description of device: HDA ATI SB, ALC892 Analog Front speakers I/O type of device: (null) Name of device: surround21:CARD=SB,DEV=0 Description of device: HDA ATI SB, ALC892 Analog 2.1 Surround output to Front and Subwoofer speakers I/O type of device: Output Name of device: surround41:CARD=SB,DEV=0 Description of device: HDA ATI SB, ALC892 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers I/O type of device: Output Name of device: surround50:CARD=SB,DEV=0 Description of device: HDA ATI SB, ALC892 Analog 5.0 Surround output to Front, Center and Rear speakers I/O type of device: Output Name of device: surround71:CARD=SB,DEV=0 Description of device: HDA ATI SB, ALC892 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers I/O type of device: Output Name of device: sysdefault:CARD=SB Description of device: HDA ATI SB, ALC892 Analog Default Audio Device I/O type of device: (null) Name of device: front:CARD=SB,DEV=0 Description of device: HDA ATI SB, ALC892 Analog Front speakers I/O type of device: (null) Now, I am more puzzled, because none of them is a microphone device explicitly. How can my program determine which of them is actually a microphone? Also, why are devices are duplicated in the output from snd_device_name_hint()? The very first device is also repeated as the 2nd last one... Regards, Alex

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[PATCH v3 00/13] Audio graph card updates and usage with Tegra210 audio
by Sameer Pujar 16 Oct '20

16 Oct '20

Summary of changes: * Support multiple instances of a component. For example there can be multiple I2S devices which can use the same component driver. * Support open platforms with empty Codec endpoint. Customers can plug their own HW and can populate codec endpoint. * In a component model there can be many components which can be connected togethe. In such cases Identify no-pcm DPCM DAI links which can be used in BE<->BE connections. * Add Tegra audio graph driver which is based on generic audio graph driver and specific customizations are done in Tegra driver. * This pushes DT support for Tegra210 based platforms which uses audio-graph card and above enhancements. The series is based on following references where DPCM usgae for Tegra Audio and simple-card driver proposal were discussed. * https://lkml.org/lkml/2020/4/30/519 (DPCM for Tegra) * https://lkml.org/lkml/2020/6/27/4 (simple-card driver) Changelog ========= v1 -> v2 -------- * Re-organized ports/endpoints description for ADMAIF and XBAR. Updated DT patches accordingly. * After above change, multiple Codec endpoint support is not required and hence dropped for now. This will be considered separately if at all required in future. * Re-ordered patches in the series. v2 -> v3 -------- * Dropped new compatible addition in generic graph driver after reviewing it with Morimoto-san. Instead added Tegra audio graph driver and new compatibles are added in the same. * Added new patches to expose new members for customization in audio graph driver. * Added new patch for Tegra audio graph driver and related documentation. * Minor change in below commit where mutex version of helper is used "ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM" * DT binding is updated to use the newly exposed compatibles * No changes in other patches Sameer Pujar (13): ASoC: soc-core: Fix component name_prefix parsing ASoC: soc-pcm: Get all BEs along DAPM path ASoC: audio-graph: Use of_node and DAI for DPCM DAI link names ASoC: audio-graph: Identify 'no_pcm' DAI links for DPCM ASoC: audio-graph: Support empty Codec endpoint ASoC: simple-card-utils: Expose new members for asoc_simple_priv ASoC: audio-graph: Update driver as per new exposed members ASoC: audio-graph: Expose helpers from audio graph ASoC: dt-bindings: tegra: Add schema for audio graph card ASoC: tegra: Add audio graph based card driver arm64: defconfig: Enable Tegra audio graph card driver arm64: tegra: Audio graph header for Tegra210 arm64: tegra: Audio graph sound card for Jetson Nano and TX1 .../sound/nvidia,tegra-audio-graph-card.yaml | 67 +++++ .../boot/dts/nvidia/tegra210-audio-graph.dtsi | 138 ++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 219 ++++++++++++++++ arch/arm64/boot/dts/nvidia/tegra210-p3450-0000.dts | 124 +++++++++ arch/arm64/configs/defconfig | 1 + include/sound/graph_card.h | 19 ++ include/sound/simple_card_utils.h | 4 + include/sound/soc.h | 1 + sound/soc/generic/audio-graph-card.c | 92 +++++-- sound/soc/soc-core.c | 3 +- sound/soc/soc-pcm.c | 3 +- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_audio_graph_card.c | 291 +++++++++++++++++++++ 14 files changed, 955 insertions(+), 18 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra-audio-graph-card.yaml create mode 100644 arch/arm64/boot/dts/nvidia/tegra210-audio-graph.dtsi create mode 100644 include/sound/graph_card.h create mode 100644 sound/soc/tegra/tegra_audio_graph_card.c -- 2.7.4

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[PATCH 1/2] MAINTAINERS: Move Kukjin Kim to credits
by Krzysztof Kozlowski 16 Oct '20

16 Oct '20

Kukjin Kim has been maintaining the Samsung ARM architectures since 2010 up to 2016. He contributed many patches for the S3C, S5P and Exynos support. However since 2016 there is little activity from him on the LKML [1] so move his name to the CREDITS. Dear Kukjin, thank you for all the effort you put in to the upstream Samsung support. [1] https://lore.kernel.org/lkml/?q=f%3A%22Kukjin+Kim%22 Cc: Kukjin Kim <kgene(a)kernel.org> Cc: Arnd Bergmann <arnd(a)arndb.de> Cc: Olof Johansson <olof(a)lixom.net> Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- CREDITS | 4 ++++ MAINTAINERS | 2 -- 2 files changed, 4 insertions(+), 2 deletions(-) diff --git a/CREDITS b/CREDITS index cb02b9923a52..5df027e12ff7 100644 --- a/CREDITS +++ b/CREDITS @@ -1910,6 +1910,10 @@ S: 660 Harvard Ave. #7 S: Santa Clara, CA 95051 S: USA +N: Kukjin Kim +E: kgene(a)kernel.org +D: Samsung S3C, S5P and Exynos ARM architectures + N: Russell King E: rmk(a)arm.linux.org.uk D: Linux/arm integrator, maintainer & hacker diff --git a/MAINTAINERS b/MAINTAINERS index 4538378de6f5..c3976803057c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -2375,7 +2375,6 @@ F: sound/soc/rockchip/ N: rockchip ARM/SAMSUNG EXYNOS ARM ARCHITECTURES -M: Kukjin Kim <kgene(a)kernel.org> M: Krzysztof Kozlowski <krzk(a)kernel.org> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) L: linux-samsung-soc(a)vger.kernel.org @@ -15497,7 +15496,6 @@ F: include/linux/clk/samsung.h F: include/linux/platform_data/clk-s3c2410.h SAMSUNG SPI DRIVERS -M: Kukjin Kim <kgene(a)kernel.org> M: Krzysztof Kozlowski <krzk(a)kernel.org> M: Andi Shyti <andi(a)etezian.org> L: linux-spi(a)vger.kernel.org -- 2.25.1

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Re: [alsa-devel] [PATCH] ALSA: usb-audio: Disable quirks for BOSS Katana amplifiers
by Takashi Iwai 16 Oct '20

16 Oct '20

On Fri, 11 Oct 2019 19:19:36 +0200, Szabolcs Szőke wrote: > > BOSS Katana amplifiers cannot be used for recording or playback if quirks > are applied > > BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=195223 > Signed-off-by: Szabolcs Szőke <szszoke.code(a)gmail.com> Applied now. Thanks. Takashi > > --- > sound/usb/pcm.c | 3 +++ > 1 file changed, 3 insertions(+) > > diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c > index 33cd26763c0e..daadb0c66eee 100644 > --- a/sound/usb/pcm.c > +++ b/sound/usb/pcm.c > @@ -348,6 +348,9 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, > ep = 0x84; > ifnum = 0; > goto add_sync_ep_from_ifnum; > + case USB_ID(0x0582, 0x01d8): /* BOSS Katana */ > + /* BOSS Katana amplifiers do not need quirks */ > + return 0; > } > > if (attr == USB_ENDPOINT_SYNC_ASYNC && > -- > 2.20.1 >

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More Generic Audio Graph Sound Card idea
by Kuninori Morimoto 16 Oct '20

16 Oct '20

Hi Mark Cc Pierre-Louis, Sameer Current audio-graph-card driver has DPCM support, but it is limited (= Mix/Mux/TDM-split/rate-convert/channel-convert). It was expanded forcibly expanded. Because of it, the connection judgement for normal vs DPCM is tricky. I know Pierre-Louis want to use it for SOF, but something is missing, thus, can't use (?). I know now Sameer is currently trying to expand it now. It is OK for me, but, more expansion in the future will be difficult/buggy. For example supporting many *generic* DPCM and normal connection in the same time will be difficult. I'm thinking below style as new audio-graph-card2. It is not tricky, thus connection judgement for normal vs DSP is easy. Then, code can be more readable (= no guaranty :P) ? I guss "audio-graph-card" and "audio-graph-card2" can have some (= not full) compatible for normal connection. But has no compatible for DPCM connection. I think I need to keep as-is for current audio-graph-card, and create new audio-graph-card2 driver. Because there is users, and card2 implementation will have huge different. Keeping compatible and update will be very difficult or buggy. But what do you think ? Note 1 is that this is just idea for now. I'm very happy to implement it, but it will be after that my all remaining patch was posted/accepted. Note 2 is that I'm not full DPCM analyst. Thus, maybe audio-graph-card2 can solve current connection issue, but might can't for unknown connection case. In such came we need ver3 (?). ------------ image --------------------------- | Front End PCMs | SoC DSP | Back End DAIs | Audio devices | ************* PCM0 <------------> * * <----DAI0-----> Codec Headset * * PCM1 <------------> * * <----DAI1-----> Codec Speakers * DSP * PCM2 <------------> * * <----DAI2-----> MODEM * * PCM3 <------------> * * <----DAI3-----> BT * * * * <----DAI4-----> DMIC * * * * <----DAI5-----> FM ************* ------------ DT sample --------------------------- sound { compatible = "audio-graph-card2"; dais = <&pcm0, &pcm1, &pcm2, &pcm3>; }; front-end { ports { pcm0: port@0 { pcm0_endpoint: endpoint { remote-endpoint = <&dsp_f0>; }; }; pcm1: port@1 { pcm1_endpoint: endpoint { remote-endpoint = <&dsp_f1>; }; }; pcm2: port@2 { pcm2_endpoint: endpoint { remote-endpoint = <&dsp_f2>; }; }; pcm3: port@3 { pcm3_endpoint: endpoint { remote-endpoint = <&dsp_f3>; }; }; }; }; dsp { compatible = "audio-graph-card2-dsp"; ports { /* Front End side */ port@0 { dsp_f0: endpoint { remote-endpoint = <&pcm0>; }; }; port@1 { dsp_f1: endpoint { remote-endpoint = <&pcm1>; }; }; port@2 { dsp_f2: endpoint { remote-endpoint = <&pcm2>; }; }; port@3 { dsp_f3: endpoint { remote-endpoint = <&pcm3>; }; }; /* Back End side */ port@4 { dsp_b0: endpoint { remote-endpoint = <&dai0>; }; }; port@5 { dsp_b1: endpoint { remote-endpoint = <&dai1>; }; }; port@6 { dsp_b2: endpoint { remote-endpoint = <&dai2>; }; }; port@7 { dsp_b3: endpoint { remote-endpoint = <&dai3>; }; }; port@8 { dsp_b4: endpoint { remote-endpoint = <&dai4>; }; }; port@9 { dsp_b5: endpoint { remote-endpoint = <&dai5>; }; }; }; }; back-end { ports { port@0 { dai0: endpoint { remote-endpoint = <&dsp_b0>; }; }; port@1 { dai1: endpoint { remote-endpoint = <&dsp_b1>; }; }; port@2 { dai2: endpoint { remote-endpoint = <&dsp_b2>; }; }; port@3 { dai3: endpoint { remote-endpoint = <&dsp_b3>; }; }; port@4 { dai4: endpoint { remote-endpoint = <&dsp_b4>; }; }; port@5 { dai5: endpoint { remote-endpoint = <&dsp_b5>; }; }; }; };

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amixer: cset: check control is readable before reading it
by GitHub pull_request - opened 15 Oct '20

15 Oct '20

alsa-project/alsa-utils pull request #63 was opened from ViorelSuman: In current implementation cset command will fail on controls which are not readable, ie just writable. Check readable state prior calling snd_ctl_elem_read. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Request URL : https://github.com/alsa-project/alsa-utils/pull/63 Patch URL : https://github.com/alsa-project/alsa-utils/pull/63.patch Repository URL: https://github.com/alsa-project/alsa-utils

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