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[PATCH v1] ASoC: tas2781: mark const variables tas2563_dvc_table as __maybe_unused
by Shenghao Ding 26 Aug '24

26 Aug '24

In case of tas2781, tas2563_dvc_table will be unused, so mark it as __maybe_unused. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2563-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h index faa3e194f73b..bb269b21f460 100644 --- a/include/sound/tas2563-tlv.h +++ b/include/sound/tas2563-tlv.h @@ -18,7 +18,7 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); /* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { +static const __maybe_unused unsigned char tas2563_dvc_table[][4] = { { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ -- 2.34.1

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[DISCUSSION] Why doesn't ALSA autogenerate/autoconfigure the mono profile by default?
by Xerox Slayer 26 Aug '24

26 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

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[PATCH v4] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 25 Aug '24

25 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. So far, the Software maintainer has not been confirmed. Once the maintainer was confimred, the guy will update his mail into audio converter section. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - Add Touch Screen support v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 34 +++++++++++++++++++++++++--------- 1 file changed, 25 insertions(+), 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 878dcd23b331..ff363eac55b1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22591,26 +22591,43 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22620,9 +22637,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

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[PATCH 0/2] Add SOF support for ACP7.0 based platform
by Vijendar Mukunda 24 Aug '24

24 Aug '24

This patch series adds SOF support for ACP7.0 based platform. Link: https://github.com/thesofproject/linux/pull/5139 Vijendar Mukunda (2): ASoC: amd: Add acpi machine id for acp7.0 version based platform ASoC: SOF: amd: add support for acp7.0 based platform sound/soc/amd/acp-config.c | 12 +++ sound/soc/amd/mach-config.h | 1 + sound/soc/sof/amd/Kconfig | 10 ++ sound/soc/sof/amd/Makefile | 4 +- sound/soc/sof/amd/acp-dsp-offset.h | 24 ++++- sound/soc/sof/amd/acp.c | 65 ++++++++++--- sound/soc/sof/amd/acp.h | 9 ++ sound/soc/sof/amd/acp70.c | 142 +++++++++++++++++++++++++++++ sound/soc/sof/amd/pci-acp70.c | 112 +++++++++++++++++++++++ 9 files changed, 365 insertions(+), 14 deletions(-) create mode 100644 sound/soc/sof/amd/acp70.c create mode 100644 sound/soc/sof/amd/pci-acp70.c -- 2.34.1

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[PATCH] ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict
by Hans de Goede 24 Aug '24

24 Aug '24

There are 2G and 4G RAM versions of the Lenovo Yoga Tab 3 X90F and it turns out that the 2G version has a DMI product name of "CHERRYVIEW D1 PLATFORM" where as the 4G version has "CHERRYVIEW C0 PLATFORM". The sys-vendor + product-version check are unique enough that the product-name check is not necessary. Drop the product-name check so that the existing DMI match for the 4G RAM version also matches the 2G RAM version. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 5e2ec60e2954..e4c3492a0c28 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -84,7 +84,6 @@ static const struct dmi_system_id lenovo_yoga_tab3_x90[] = { /* Lenovo Yoga Tab 3 Pro YT3-X90, codec missing from DSDT */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), - DMI_MATCH(DMI_PRODUCT_NAME, "CHERRYVIEW D1 PLATFORM"), DMI_MATCH(DMI_PRODUCT_VERSION, "Blade3-10A-001"), }, }, -- 2.46.0

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[PATCH 6.11 regression fix] ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards harder
by Hans de Goede 24 Aug '24

24 Aug '24

Since commit 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()") dummy codecs declared like this: SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); expand to: static struct snd_soc_dai_link_component dummy[] = { }; Which means that dummy is a zero sized array and thus dais[i].codecs should not be dereferenced *at all* since it points to the address of the next variable stored in the data section as the "dummy" variable has an address but no size, so even dereferencing dais[0] is already an out of bounds array reference. Which means that the if (dais[i].codecs->name) check added in commit 7d99a70b6595 ("ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards") relies on that the part of the next variable which the name member maps to just happens to be NULL. Which apparently so far it usually is, except when it isn't and then it results in crashes like this one: [ 28.795659] BUG: unable to handle page fault for address: 0000000000030011 ... [ 28.795780] Call Trace: [ 28.795787] <TASK> ... [ 28.795862] ? strcmp+0x18/0x40 [ 28.795872] 0xffffffffc150c605 [ 28.795887] platform_probe+0x40/0xa0 ... [ 28.795979] ? __pfx_init_module+0x10/0x10 [snd_soc_sst_bytcr_wm5102] Really fix things this time around by checking dais.num_codecs != 0. Fixes: 7d99a70b6595 ("ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards") Cc: stable(a)vger.kernel.org Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- 9 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index dce6a2086f2a..6da1517c53c6 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -605,7 +605,7 @@ static int broxton_audio_probe(struct platform_device *pdev) int i; for (i = 0; i < ARRAY_SIZE(broxton_rt298_dais); i++) { - if (card->dai_link[i].codecs->name && + if (card->dai_link[i].num_codecs && !strncmp(card->dai_link[i].codecs->name, "i2c-INT343A:00", I2C_NAME_SIZE)) { if (!strncmp(card->name, "broxton-rt298", diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index c014d85a08b2..df3c2a7b64d2 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -241,7 +241,7 @@ static int snd_byt_cht_cx2072x_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_cx2072x_dais); i++) { - if (byt_cht_cx2072x_dais[i].codecs->name && + if (byt_cht_cx2072x_dais[i].num_codecs && !strcmp(byt_cht_cx2072x_dais[i].codecs->name, "i2c-14F10720:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index f4ac3ddd148b..08c598b7e1ee 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -245,7 +245,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(dailink); i++) { - if (dailink[i].codecs->name && + if (dailink[i].num_codecs && !strcmp(dailink[i].codecs->name, "i2c-DLGS7213:00")) { dai_index = i; break; diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 2fcec2e02bb5..77b91ea4dc32 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -546,7 +546,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { - if (byt_cht_es8316_dais[i].codecs->name && + if (byt_cht_es8316_dais[i].num_codecs && !strcmp(byt_cht_es8316_dais[i].codecs->name, "i2c-ESSX8316:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a64d1989e28a..db4a33680d94 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1677,7 +1677,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_rt5640_dais); i++) { - if (byt_rt5640_dais[i].codecs->name && + if (byt_rt5640_dais[i].num_codecs && !strcmp(byt_rt5640_dais[i].codecs->name, "i2c-10EC5640:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 80c841b000a3..8514b79f389b 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -910,7 +910,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_rt5651_dais); i++) { - if (byt_rt5651_dais[i].codecs->name && + if (byt_rt5651_dais[i].num_codecs && !strcmp(byt_rt5651_dais[i].codecs->name, "i2c-10EC5651:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index cccb5e90c0fe..e5a7cc606aa9 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -605,7 +605,7 @@ static int snd_byt_wm5102_mc_probe(struct platform_device *pdev) /* find index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_wm5102_dais); i++) { - if (byt_wm5102_dais[i].codecs->name && + if (byt_wm5102_dais[i].num_codecs && !strcmp(byt_wm5102_dais[i].codecs->name, "wm5102-codec")) { dai_index = i; diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index eb41b7115d01..1da9ceee4d59 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -569,7 +569,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* set correct codec name */ for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) - if (cht_dailink[i].codecs->name && + if (cht_dailink[i].num_codecs && !strcmp(cht_dailink[i].codecs->name, "i2c-10EC5645:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index be2d1a8dbca8..d68e5bc755de 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -466,7 +466,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* find index of codec dai */ for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) { - if (cht_dailink[i].codecs->name && + if (cht_dailink[i].num_codecs && !strcmp(cht_dailink[i].codecs->name, RT5672_I2C_DEFAULT)) { dai_index = i; break; -- 2.46.0

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[GIT PULL] ASoC fixes for v6.11-rc3
by Mark Brown 23 Aug '24

23 Aug '24

The following changes since commit 72776774b55bb59b7b1b09117e915a5030110304: ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value (2024-08-08 20:34:56 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc3 for you to fetch changes up to bb4485562f5907708f1c218b5d70dce04165d1e1: ASoC: cs-amp-lib: Ignore empty UEFI calibration entries (2024-08-22 17:52:26 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A relatively large collection of fixes here, all driver specific and none of them particularly major, plus one MAINTAINERS update. There's been a bunch of work on module autoloading from several people. ---------------------------------------------------------------- Albert Jakieła (1): ASoC: SOF: mediatek: Add missing board compatible Dmitry Baryshkov (2): ASoC: codecs: lpass-macro: fix version strings returned for 1.x codecs ASoC: codecs: lpass-va-macro: warn on unknown version Hongbo Li (2): ASoC: allow module autoloading for table db1200_pids ASoC: allow module autoloading for table board_ids Krzysztof Kozlowski (2): ASoC: MAINTAINERS: Drop Banajit Goswami from Qualcomm sound drivers ASoC: codecs: wcd937x: Fix missing de-assert of reset GPIO Mark Brown (1): ASoC: codecs: warn on unknown codec version Richard Fitzgerald (2): ASoC: cs-amp-lib-test: Force test calibration blob entries to be valid ASoC: cs-amp-lib: Ignore empty UEFI calibration entries Vijendar Mukunda (3): ASoC: SOF: amd: move iram-dram fence register programming sequence ASoC: SOF: amd: Fix for incorrect acp error register offsets ASoC: SOF: amd: Fix for acp init sequence YR Yang (1): ASoC: mediatek: mt8188: Mark AFE_DAC_CON0 register as volatile Yuntao Liu (1): ASoC: amd: acp: fix module autoloading MAINTAINERS | 1 - sound/soc/amd/acp/acp-legacy-mach.c | 2 ++ sound/soc/amd/acp/acp-sof-mach.c | 2 ++ sound/soc/au1x/db1200.c | 1 + sound/soc/codecs/cs-amp-lib-test.c | 9 ++++++ sound/soc/codecs/cs-amp-lib.c | 7 +++- sound/soc/codecs/lpass-macro-common.h | 6 ++++ sound/soc/codecs/lpass-va-macro.c | 4 +++ sound/soc/codecs/wcd937x.c | 5 ++- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 1 + sound/soc/sof/amd/acp-dsp-offset.h | 6 ++-- sound/soc/sof/amd/acp.c | 52 ++++++++++++++++++++---------- sound/soc/sof/amd/acp.h | 9 ++++-- sound/soc/sof/amd/pci-acp63.c | 2 ++ sound/soc/sof/amd/pci-rmb.c | 2 ++ sound/soc/sof/amd/pci-rn.c | 2 ++ sound/soc/sof/mediatek/mt8195/mt8195.c | 3 ++ 17 files changed, 88 insertions(+), 26 deletions(-)

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[PATCH v8 0/4] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 23 Aug '24

23 Aug '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is v8 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.12 or v6.13 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871q5tnuok.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87bk4oqerx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/8734pctmte.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0ctwzr4.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87cymvlmki.wl-kuninori.morimoto.gx@renesas.com v8 -> v7 - rebase to latest ASoC branch (for-6.12) v6 -> v7 - based on latest ASoC branch (for-6.12) v5 -> v6 - based on latest ASoC branch v4 -> v5 - typo fix: limition -> limitation - typo fix: catpure -> capture - include Intel patch v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (3): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() ASoC: Intel: sof_sdw: use playback/capture_only flags include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/sdw_utils/soc_sdw_utils.c | 4 +- sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 13 files changed, 47 insertions(+), 88 deletions(-) -- 2.43.0

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ALC285 headphone no sound
by Kailang 22 Aug '24

22 Aug '24

Hi Takashi, I also changed with ALC215 ALC285 ALC289 ALC225 ALC295 ALC299 codec. It maybe has similar issue as ALC256. Let it has same procedure. Many Thanks. BR, Kailang > -----Original Message----- > From: Kailang > Sent: Thursday, August 22, 2024 11:14 AM > To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: ALC256 headphone no sound > > Hi Takashi, > > Attach patch will solve headphone issue for alc256. > Many Thanks. > > BR, > Kailang

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Re: [PATCH] [RFC] ASoC: Conditional PCM support
by Jaroslav Kysela 22 Aug '24

22 Aug '24

On 21. 08. 24 12:18, Cezary Rojewski wrote: > Conditional PCM (condpcm) helps facilitate modern audio usecases such as > Echo Cancellations and Noise Reduction. These are not invoked by the > means of userspace application opening an endpoint (FrontEnd) but are a > "side effect" of selected PCMs running simultaneously e.g.: if both > Speaker (source) and Microphone Array (sink) are running, reference > data from the Speaker and take it into account when processing capture > for better voice command detection ratio. > > Which PCMs are needed for given conditional PCM to be spawned is > determinated by the driver when registering the condpcm. Perhaps the naming may be changed to "sidepcm" or "monitorpcm"/"monpcm". It looks better in my eyes. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

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