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

[PATCH 00/11] ASoC: codecs: fix kernel-doc and W=1 warnings
by Pierre-Louis Bossart 03 Jul '20

03 Jul '20

Second batch of cleanups to remove all the warnings reported by buildbots/compile tests. Pierre-Louis Bossart (11): ASoC: codecs: ak4458: fix 'set but not used' warning ASoC: codecs: cros_ec_codec: fix 'defined but not used' warning ASoC: codecs: da7210: fix kernel-doc ASoC: codecs: da7219: fix 'defined but not used' warning ASoC: codecs: es8316: fix 'defined but not used' warning ASoC: codecs: max98390: fix 'defined but not used' warning ASoC: codecs: rt*: fix 'defined but not used' warning ASoC: codecs: rt5659: add _maybe_unused as needed ASoC: codecs: rt5631: fix kernel-doc ASoC: codecs: tlv320aic26: fix kernel-doc warning ASoC: codecs: jz4770: add _maybe_unused as needed sound/soc/codecs/ak4458.c | 2 +- sound/soc/codecs/cros_ec_codec.c | 2 ++ sound/soc/codecs/da7210.c | 12 +++++++----- sound/soc/codecs/da7219.c | 2 ++ sound/soc/codecs/es8316.c | 2 ++ sound/soc/codecs/jz4770.c | 2 +- sound/soc/codecs/max98390.c | 8 -------- sound/soc/codecs/rt274.c | 2 ++ sound/soc/codecs/rt286.c | 2 ++ sound/soc/codecs/rt298.c | 2 ++ sound/soc/codecs/rt5631.c | 8 ++++++-- sound/soc/codecs/rt5659.c | 14 +++++++------- sound/soc/codecs/rt5660.c | 2 ++ sound/soc/codecs/rt5677-spi.c | 2 ++ sound/soc/codecs/tlv320aic26.c | 2 +- 15 files changed, 39 insertions(+), 25 deletions(-) base-commit: a2b782d59c57cb6f673dbb4804ffc500336d7a54 -- 2.25.1

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[HDAudio][ALC295] speakers not working on Acer Travelmate P614
by Stanislav Kazmin 03 Jul '20

03 Jul '20

Hello, I have the following problem on my Acer TravelMate P614-51T-G2: I have all the audio sinks correctly identified (like HDMI, DMic, Headphones) but the internal speakers do not produce any sound. What I have tried so far: - hard/soft shutdown on Windows and reboot - disable/re-enable speakers and microphone in BIOS - uninstall pulseaudio and test alsa alone (so it is defeitely **not** a pulseaudio issue) - switch from sof-hda-dsp to snd-hda-intel driver (without Dmic support) - retask pins 0x14, 0x16, 0x1b to "Internal Speakers" through hdajackretask (only basic, without advanced features) - removed/reconnected the headphones `alsamixer` shows all needed sinks and nothing is muted. I already discussed the issue at sof github https://github.com/thesofproject/ sof/issues/3058 but since the same issue occurs on `snd-hda-intel` legacy river, I was advised to communicate with alsa-devel team. The alsa-info.sh result are linked at http://alsa-project.org/db/? f=252f92c7a1df3c755d16ee69353b26d2535a4d81 I have tested the kernel 5.8-rc1 but it does not make any difference. Let me know if I can do anything else to test the issue.

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[PATCH 0/2] ASoC: fsl-asoc-card: add support for generic codecs
by Arnaud Ferraris 03 Jul '20

03 Jul '20

fsl-asoc-card currently only works with AC97 or a selection of codecs, although the hardware is capable of more. Supporting generic codecs when acting as I2S slave (codec is master) would be useful, especially when using Bluetooth audio, as these are generally simple I2S devices not controlled by the sound subsystem. This will allow using simple/dummy codecs along with ASRC. Arnaud Ferraris (2): dt-bindings: sound: fsl-asoc-card: add new compatible for I2S slave ASoC: fsl-asoc-card: add support for generic I2S slave use-case Documentation/devicetree/bindings/sound/fsl-asoc-card.txt | 23 ++++++++++++++++++++++- sound/soc/fsl/fsl-asoc-card.c | 46 +++++++++++++++++++++++++++++++++++----------- 2 files changed, 57 insertions(+), 12 deletions(-)

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How to identify Alsa eLements?
by Tanjeff Moos 03 Jul '20

03 Jul '20

Hi alsa-devel, I'm programming a GUI for the mixer in my Focusrite Scarlett 6i6 interface. The Alsa driver supports that functionality, i.e. I can control the mixer using alsamixer, for example. Now I'm writing a Qt GUI for it. I wonder how I can reliably identify a mixer control element (e.g. the master volume control). Each element has an ID (struct snd_ctl_elem_id), consisting of multiple elements: - numid (seems to change at each boot) - iface - device - subdevice - name (is always constant) - index The docs says I can identify an element by the combination of name and index. My questions: 1) Are name and index enough to identify an element? 2) How can I obtain the index? The name and numid are shown by 'amixer contents', for example. Kind regards, Tanjeff

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sof-hda-dsp UCM doesn't work with the latest alsa-lib
by GitHub issues - opened 03 Jul '20

03 Jul '20

alsa-project/alsa-lib issue #67 was opened from libinyang: In the latest UCM for sof-hda-dsp, it has supported Syntax 3 and it uses $var for hdmi. In sof-hda-dsp/Hdmi.conf, It defines "HdmiNum 1", and later it will redefine "HdmiNum 2", then "HdmiNum 3". It will try to `Include.hdmi1.File "/codecs/hda/hdmi.conf"` to use these definitions. But actually, only "HdmiNum 3" will be used for HDMI1, HDMI2 and HDMI3. The old value of "1" and "2" will be overridden when parsing HDMI1 and HDMI2. This is the bug of alsa-lib initialization sequence issue. HDA-Intel/Hdmi.conf should have the same issue. Issue URL : https://github.com/alsa-project/alsa-lib/issues/67 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH RFC alsa-ucm-conf] sof-hda-dsp: Use Master mixer to control volume with dual stereo speakers
by Benjamin Poirier 03 Jul '20

03 Jul '20

The Lenovo ThinkPad X1 Carbon 7th is a laptop with two sets of stereo speakers driven from a single stereo output. The volume of each set of speakers can be controlled independently through two separate mixers. In that case, one of those mixers serves a dual purpose: it also controls headphone volume. With the current ucm profile, when outputting to speakers, pulseaudio uses only the mixer for the front speakers to control volume. This is ineffective; rear speaker volume must be controlled as well. In order to control both sets of speakers at once, indicate that the Master mixer should be used. Moreover, when switching between speaker and headphone outputs, the level of the dual purpose mixer should be restored. Since this kind of constraint cannot be described with ucm, indicate that the Master mixer should be used to control volume instead when outputting to headphones. This patch depends on a modification to the snd_hda_codec_realtek kernel driver used on that system to rename one of the controls. Signed-off-by: Benjamin Poirier <benjamin.poirier(a)gmail.com> --- Note that I plan to submit the related kernel patch upstream shortly, in the meantime, it is available here: https://github.com/gobenji/thinkpad-x1-gen7-sound Also note that this patch is based on commit 38e5906cd1b1 ("sof-hda-dsp: fix the device order (Hdmi devices)"). While testing with libasound built from alsa-lib git head 485930ea5dc8 ("ucm: substitution - remove duplicate allow_empty assignment"), alsa-ucm-conf versions including dcef48f13d4f ("HDA-Intel: add support for AMD acp microphone devices") and newer do not work for me. I've tested up to current head, ffe0cab5cfce ("sof-hda-dsp: use sof-hda-dsp/Hdmi.conf"). Looking at the output of `pulseaudio --log-level=debug`, things go wrong at: I: [pulseaudio] (alsa-lib)main.c: error: failed to import sof-hda-dsp use case configuration -17 ucm2/sof-hda-dsp/HiFi.conf | 32 ++++++++++++++++++++++++++++---- 1 file changed, 28 insertions(+), 4 deletions(-) diff --git a/ucm2/sof-hda-dsp/HiFi.conf b/ucm2/sof-hda-dsp/HiFi.conf index 62358e2..661a086 100644 --- a/ucm2/sof-hda-dsp/HiFi.conf +++ b/ucm2/sof-hda-dsp/HiFi.conf @@ -25,10 +25,22 @@ SectionDevice."Headphones" { Value { PlaybackPriority 200 PlaybackPCM "hw:${CardId}" - PlaybackMixerElem "Headphone" PlaybackMasterElem "Master" - PlaybackVolume "Headphone Playback Volume" PlaybackSwitch "Headphone Playback Switch" + If.hpmixer { + Condition { + Type ControlExists + Control "name='Headphone/Bass Speaker Playback Volume'" + } + True { + PlaybackMixerElem "Master" + PlaybackVolume "Master Playback Volume" + } + False { + PlaybackMixerElem "Headphone" + PlaybackVolume "Headphone Playback Volume" + } + } If.jack { Condition { Type ControlExists @@ -77,10 +89,22 @@ SectionDevice."Speaker" { Value { PlaybackPriority 100 PlaybackPCM "hw:${CardId}" - PlaybackMixerElem "Speaker" PlaybackMasterElem "Master" - PlaybackVolume "Speaker Playback Volume" PlaybackSwitch "Speaker Playback Switch" + If.spkmixer { + Condition { + Type ControlExists + Control "name='Headphone/Bass Speaker Playback Volume'" + } + True { + PlaybackMixerElem "Master" + PlaybackVolume "Master Playback Volume" + } + False { + PlaybackMixerElem "Speaker" + PlaybackVolume "Speaker Playback Volume" + } + } } } -- 2.27.0

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[PATCH 1/3] ALSA: hda/realtek - Enable audio jacks of Acer vCopperbox with ALC269VC
by Jian-Hong Pan 03 Jul '20

03 Jul '20

The Acer desktop vCopperbox with ALC269VC cannot detect the MIC of headset, the line out and internal speaker until ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS quirk applied. Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> Signed-off-by: Chris Chiu <chiu(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 737ef82a75fd..dfb4bca07d3f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6149,6 +6149,7 @@ enum { ALC236_FIXUP_HP_MUTE_LED, ALC298_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET, ALC295_FIXUP_ASUS_MIC_NO_PRESENCE, + ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS, }; static const struct hda_fixup alc269_fixups[] = { @@ -7327,6 +7328,17 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE }, + [ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90100120 }, /* use as internal speaker */ + { 0x18, 0x02a111f0 }, /* use as headset mic, without its own jack detect */ + { 0x1a, 0x01011020 }, /* use as line out */ + { }, + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -7346,6 +7358,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1099, "Acer Aspire E5-523G", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x110e, "Acer Aspire ES1-432", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1246, "Acer Predator Helios 500", ALC299_FIXUP_PREDATOR_SPK), + SND_PCI_QUIRK(0x1025, 0x1247, "Acer vCopperbox", ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS), SND_PCI_QUIRK(0x1025, 0x128f, "Acer Veriton Z6860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1290, "Acer Veriton Z4860G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1025, 0x1291, "Acer Veriton Z4660G", ALC286_FIXUP_ACER_AIO_HEADSET_MIC), -- 2.27.0

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[PATCH] ASoC: cros_ec_codec: Log results when EC commands fail
by Yu-Hsuan Hsu 03 Jul '20

03 Jul '20

Log results of failed EC commands to identify a problem more easily. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/cros_ec_codec.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cros_ec_codec.c b/sound/soc/codecs/cros_ec_codec.c index 8d45c628e988e..a4ab62f59efa6 100644 --- a/sound/soc/codecs/cros_ec_codec.c +++ b/sound/soc/codecs/cros_ec_codec.c @@ -90,10 +90,17 @@ static int send_ec_host_command(struct cros_ec_device *ec_dev, uint32_t cmd, if (outsize) memcpy(msg->data, out, outsize); - ret = cros_ec_cmd_xfer_status(ec_dev, msg); + ret = cros_ec_cmd_xfer(ec_dev, msg); if (ret < 0) goto error; + if (msg->result != EC_RES_SUCCESS) { + dev_err(ec_dev->dev, "Command %d failed: %d\n", cmd, + msg->result); + ret = -EPROTO; + goto error; + } + if (insize) memcpy(in, msg->data, insize); -- 2.27.0.212.ge8ba1cc988-goog

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alsa-lib: ctl: improve documentation for identifier of control element
by GitHub pull_request - opened 03 Jul '20

03 Jul '20

alsa-project/alsa-lib pull request #66 was opened from takaswie: In documentation, there are two ways relevant to the identifier of control element. However, the case of combination has the lack of parameters. This commit improves documentation in this point. Fixes: f3c24de8c0df ("ctl: add an overview for design of ALSA control interface") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-lib/pull/66 Patch URL : https://github.com/alsa-project/alsa-lib/pull/66.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] ASoC: dt-bindings: simple-card: Fix 'make dt_binding_check' warnings
by Fabio Estevam 03 Jul '20

03 Jul '20

The following build warnings are seen with 'make dt_binding_check': Documentation/devicetree/bindings/sound/simple-card.example.dts:209.46-211.15: Warning (unit_address_vs_reg): /example-4/sound/simple-audio-card,cpu@0: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:213.37-215.15: Warning (unit_address_vs_reg): /example-4/sound/simple-audio-card,cpu@1: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:250.42-261.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@0: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:263.42-288.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:270.32-272.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@0: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:273.23-275.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@1: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:276.23-278.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@2: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:279.23-281.19: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@1/cpu@3: node has a unit name, but no reg or ranges property Documentation/devicetree/bindings/sound/simple-card.example.dts:290.42-303.15: Warning (unit_address_vs_reg): /example-5/sound/simple-audio-card,dai-link@2: node has a unit name, but no reg or ranges property Fix them all. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- .../devicetree/bindings/sound/simple-card.yaml | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml index 8132d0c0f00a..35e669020296 100644 --- a/Documentation/devicetree/bindings/sound/simple-card.yaml +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -378,6 +378,8 @@ examples: - | sound { compatible = "simple-audio-card"; + #address-cells = <1>; + #size-cells = <0>; simple-audio-card,name = "rsnd-ak4643"; simple-audio-card,format = "left_j"; @@ -391,10 +393,12 @@ examples: "ak4642 Playback", "DAI1 Playback"; dpcmcpu: simple-audio-card,cpu@0 { + reg = <0>; sound-dai = <&rcar_sound 0>; }; simple-audio-card,cpu@1 { + reg = <1>; sound-dai = <&rcar_sound 1>; }; @@ -418,6 +422,8 @@ examples: - | sound { compatible = "simple-audio-card"; + #address-cells = <1>; + #size-cells = <0>; simple-audio-card,routing = "pcm3168a Playback", "DAI1 Playback", @@ -426,6 +432,7 @@ examples: "pcm3168a Playback", "DAI4 Playback"; simple-audio-card,dai-link@0 { + reg = <0>; format = "left_j"; bitclock-master = <&sndcpu0>; frame-master = <&sndcpu0>; @@ -439,22 +446,23 @@ examples: }; simple-audio-card,dai-link@1 { + reg = <1>; format = "i2s"; bitclock-master = <&sndcpu1>; frame-master = <&sndcpu1>; convert-channels = <8>; /* TDM Split */ - sndcpu1: cpu@0 { + sndcpu1: cpu0 { sound-dai = <&rcar_sound 1>; }; - cpu@1 { + cpu1 { sound-dai = <&rcar_sound 2>; }; - cpu@2 { + cpu2 { sound-dai = <&rcar_sound 3>; }; - cpu@3 { + cpu3 { sound-dai = <&rcar_sound 4>; }; codec { @@ -466,6 +474,7 @@ examples: }; simple-audio-card,dai-link@2 { + reg = <2>; format = "i2s"; bitclock-master = <&sndcpu2>; frame-master = <&sndcpu2>; -- 2.17.1

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