August 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 0/3] Update the register list of MICFIL
by Chancel Liu 02 Aug '23

02 Aug '23

MICFIL IP is upgraded on i.MX93 platform. Add new registers and new bit definition. changes in v2: - rename check_version to use_verid to make it more explicit - rename fsl_micfil_check_version to fsl_micfil_use_verid Chancel Liu (3): ASoC: fsl_micfil: Add new registers and new bit definition ASoC: fsl_micfil: Add fsl_micfil_use_verid function ASoC: fsl_micfil: Use SET_SYSTEM_SLEEP_PM_OPS to simplify PM sound/soc/fsl/fsl_micfil.c | 100 ++++++++++++++++++++++++++++++------- sound/soc/fsl/fsl_micfil.h | 64 ++++++++++++++++++++++++ 2 files changed, 146 insertions(+), 18 deletions(-) -- 2.25.1

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[PATCH 0/2] ALSA: hda/realtek: Add headphone quirks for Aya devices
by ruinairas1992＠gmail.com 02 Aug '23

02 Aug '23

The content of this message was lost. It was probably cross-posted to multiple lists and previously handled on another list.

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[PATCH v1] ALSA: hda: cs35l41: Print amp configuration after bind
by Stefan Binding 02 Aug '23

02 Aug '23

Print amp configuration information to be able to confirm ACPI _DSD information (and other useful info) for each amp on each system using CS35L41, without having to get the acpidump. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index 98feb5ccd5866..825e551be9bb8 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1124,6 +1124,13 @@ static int cs35l41_hda_bind(struct device *dev, struct device *master, void *mas pm_runtime_mark_last_busy(dev); pm_runtime_put_autosuspend(dev); + dev_info(cs35l41->dev, + "CS35L41 Bound - SSID: %s, BST: %d, VSPK: %d, CH: %c, FW EN: %d, SPKID: %d\n", + cs35l41->acpi_subsystem_id, cs35l41->hw_cfg.bst_type, + cs35l41->hw_cfg.gpio1.func == CS35l41_VSPK_SWITCH, + cs35l41->hw_cfg.spk_pos ? 'R' : 'L', + cs35l41->firmware_running, cs35l41->speaker_id); + return ret; } -- 2.34.1

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[PATCH -next] ALSA: ac97: set variables dev_attr_vendor_id to static
by Yu Liao 02 Aug '23

02 Aug '23

sparse reports sound/ac97/bus.c:465:1: sparse: sparse: symbol 'dev_attr_vendor_id' was not declared. Should it be static? This variable is only used in its defining file, so it should be static. Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202307242300.Oy0Dp2QI-lkp@intel.com/ Signed-off-by: Yu Liao <liaoyu15(a)huawei.com> --- sound/ac97/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c index 6067c04ce4c0..3173e9d98927 100644 --- a/sound/ac97/bus.c +++ b/sound/ac97/bus.c @@ -462,7 +462,7 @@ static ssize_t vendor_id_show(struct device *dev, return sysfs_emit(buf, "%08x", codec->vendor_id); } -DEVICE_ATTR_RO(vendor_id); +static DEVICE_ATTR_RO(vendor_id); static struct attribute *ac97_dev_attrs[] = { &dev_attr_vendor_id.attr, -- 2.25.1

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[PATCH] ASoC: intel: use devm_snd_soc_register_dai()
by Kuninori Morimoto 02 Aug '23

02 Aug '23

It is still using snd_soc_{un}register_dai() manually. Let's use devm_snd_soc_register_dai(). Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/intel/avs/pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/avs/pcm.c b/sound/soc/intel/avs/pcm.c index 1fbb2c2fadb5..dd03db31c2d8 100644 --- a/sound/soc/intel/avs/pcm.c +++ b/sound/soc/intel/avs/pcm.c @@ -1309,8 +1309,6 @@ static void avs_component_hda_unregister_dais(struct snd_soc_component *componen for_each_pcm_streams(stream) snd_soc_dapm_free_widget(snd_soc_dai_get_widget(dai, stream)); - - snd_soc_unregister_dai(dai); } } @@ -1375,7 +1373,7 @@ static int avs_component_hda_probe(struct snd_soc_component *component) } } - dai = snd_soc_register_dai(component, &dais[i], false); + dai = devm_snd_soc_register_dai(component->dev, component, &dais[i], false); if (!dai) { dev_err(component->dev, "register dai for %s failed\n", pcm->name); -- 2.25.1

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How do I set up multiple codecs on one I2S - without TDM?
by John Watts 01 Aug '23

01 Aug '23

Hello there! I have an interesting problem that I'm trying to solve. It will require writing adding driver support but right now I'm struggling to understand how I should go about it all. As a background, this is with the Allwinner D1 or T113. It has a 16 channel I2S controller and each channel can be configured to use a specific pin and TDM time slice. Even channels are low LRCLK, odd are high LRCLK. So for my situation I want to have six channels: - Channel 0: Timeslot 0, Pin 0, LRCLK 0 -> ADC 1 - Channel 1: Timeslot 0, Pin 0, LRCLK 1 -> ADC 1 - Channel 2: Timeslot 0, Pin 1, LRCLK 0 -> ADC 2 - Channel 3: Timeslot 0, Pin 1, LRCLK 1 -> ADC 2 - Channel 4: Timeslot 0, Pin 2, LRCLK 0 -> ADC 3 - Channel 5: Timeslot 0, Pin 2, LRCLK 1 -> ADC 3 These ADCs are WM8782s, completely TDM unaware. If they were TDM aware then I could run even more ADCs by having multiple separate TDMs going on each pin. I have tried to look online for I2S controllers that do the same thing but haven't had much luck. The current driver for this controller only supports TDM. So I will have to add support for this. So my questions are: - How do I express this in terms of ALSA and sound card concepts? - How do I express this in the device tree? My immediate thought is that this is a situation where I have a card with 3 dai-links between the I2S controller and each separate codec. I have written this tree as a test: wm8782_1: stereo-adc { compatible = "wlf,wm8782"; Vdda-supply = <&reg_vcc>; Vdd-supply = <&reg_vcc>; #sound-dai-cells = <0>; }; wm8782_2: stereo-adc { compatible = "wlf,wm8782"; Vdda-supply = <&reg_vcc>; Vdd-supply = <&reg_vcc>; #sound-dai-cells = <0>; }; wm8782_3: stereo-adc { compatible = "wlf,wm8782"; Vdda-supply = <&reg_vcc>; Vdd-supply = <&reg_vcc>; #sound-dai-cells = <0>; }; sound { compatible = "simple-audio-card"; simple-audio-card,name = "Testcard In"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&sound_cpu>; simple-audio-card,frame-master = <&sound_cpu>; simple-audio-card,mclk-fs = <128>; #address-cells = <1>; #size-cells = <0>; simple-audio-card,dai-link@0 { reg = <0>; sound_cpu: cpu { sound-dai = <&i2s2>; }; codec { sound-dai = <&wm8782_1>; }; }; simple-audio-card,dai-link@1 { reg = <1>; cpu { sound-dai = <&i2s2>; }; codec { sound-dai = <&wm8782_2>; }; }; simple-audio-card,dai-link@2 { reg = <2>; cpu { sound-dai = <&i2s2>; }; codec { sound-dai = <&wm8782_3>; }; }; }; But I get this kernel error: kobject_add_internal failed for 2034000.i2s-wm8782 with -EEXIST, don't try to register things with the same name asoc-simple-card: probe of sound failed with error -12 The WM8782 doesn't support TDM or multiplexing so I'm not exactly sure if this is correct behaviour or my mistake configuring. I tried adding TDM nodes which didn't help. I also tried having a single link with multiple codecs on sound-dai which did create a sound card but it was unclear to me if this was ignoring the other codecs I had listed or not. I also can't test this at this point as the sunxi codec driver assumes that the number of channels is equal to TDM slots, which isn't true any more. Any pointers in the right direction would be a huge help. Surely someone else has had this problem. Thanks, John.

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[PATCH] ASoC: rt5665: add missed regulator_bulk_disable
by Zhang Shurong 01 Aug '23

01 Aug '23

The driver forgets to call regulator_bulk_disable() Add the missed call to fix it. Fixes: 33ada14a26c8 ("ASoC: add rt5665 codec driver") Signed-off-by: Zhang Shurong <zhang_shurong(a)foxmail.com> --- sound/soc/codecs/rt5665.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 83c367af91da..de36cad8a559 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -4756,6 +4756,13 @@ static void rt5665_calibrate_handler(struct work_struct *work) rt5665_calibrate(rt5665); } +static void rt5665_i2c_disable_regulators(void *data) +{ + struct rt5665_priv *rt5665 = data; + + regulator_bulk_disable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); +} + static int rt5665_i2c_probe(struct i2c_client *i2c) { struct rt5665_platform_data *pdata = dev_get_platdata(&i2c->dev); @@ -4786,6 +4793,11 @@ static int rt5665_i2c_probe(struct i2c_client *i2c) return ret; } + ret = devm_add_action_or_reset(&i2c->dev, rt5665_i2c_disable_regulators, + rt5665); + if (ret) + return ret; + ret = regulator_bulk_enable(ARRAY_SIZE(rt5665->supplies), rt5665->supplies); if (ret != 0) { -- 2.41.0

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[PATCH 00/23] ASoC: Intel: machine driver updates for 6.6
by Pierre-Louis Bossart 01 Aug '23

01 Aug '23

Lots of small cleanups coming from Bard Liao and Charles Keepax for SoundWire platforms, and minor additions for RVPs and Chromebooks. Balamurugan C (1): ASoC: Intel: soc-acpi: Add entry for rt711-sdca-sdw at link 0 in RPL match table Bard Liao (2): ASoC: Intel: sof_sdw: rename link_id to be_id ASoC: Intel: sof_sdw: break earlier when a adr link contains different codecs Chao Song (1): ASoC: Intel: sof_sdw: add cs35l56 codec info Charles Keepax (13): ASoC: intel: sof_sdw: Use consistent variable naming for links ASoC: intel: sof_sdw: Rename codec_idx to codec_dlc_index ASoC: intel: sof_sdw: Remove some extra line breaks ASoC: intel: sof_sdw: Use a module device table ASoC: intel: sof_sdw: Simplify find_codec_info_acpi ASoC: intel: sof_sdw: Constify parameter to find_codec_part_info ASoC: intel: sof_sdw: Minor tidy up of mc_probe ASoC: intel: sof_sdw: Remove redundant parameters in dai creation ASoC: intel: sof_sdw: Move amp_num initialisation to mc_probe ASoC: intel: sof_sdw: Allow direct specification of CODEC name ASoC: intel: sof_sdw: Clean up DAI link counting ASoC: intel: sof_sdw: Merge codec_conf_alloc into dailink_info ASoC: intel: sof_sdw: Move group_generated logic Pierre-Louis Bossart (3): ASoC: Intel: sof_sdw: reorder SoundWire codecs in Kconfig ASoC: Intel: sof_sdw: allow mockup amplifier to provide feedback ASoC: Intel: sof_sdw: add support for SKU 0AFE Terry Cheong (1): ASoC: Intel: sof_rt5682: add RPL support for MAX98357A speaker Uday M Bhat (1): ASoC: Intel: sof_sdw: Update BT offload config for soundwire config apoorv (1): ASoC: Intel: sof_rt5682 add support for HDMI_In capture sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/Makefile | 2 +- sound/soc/intel/boards/sof_rt5682.c | 55 +++ sound/soc/intel/boards/sof_sdw.c | 404 +++++++++--------- sound/soc/intel/boards/sof_sdw_common.h | 7 + sound/soc/intel/boards/sof_sdw_cs_amp.c | 47 ++ .../intel/common/soc-acpi-intel-adl-match.c | 7 + .../intel/common/soc-acpi-intel-rpl-match.c | 27 ++ 8 files changed, 342 insertions(+), 212 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw_cs_amp.c -- 2.39.2

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[PATCH -next] ALSA: usb-audio: Remove unused function declaration
by Yue Haibing 01 Aug '23

01 Aug '23

Commit 68e67f40b734 ("ALSA: snd-usb: move calls to usb_set_interface") leave this unused declaration. Signed-off-by: Yue Haibing <yuehaibing(a)huawei.com> --- sound/usb/endpoint.h | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index c09f68ce08b1..ba70f52f6860 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -44,7 +44,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep); void snd_usb_endpoint_stop(struct snd_usb_endpoint *ep, bool keep_pending); void snd_usb_endpoint_sync_pending_stop(struct snd_usb_endpoint *ep); void snd_usb_endpoint_suspend(struct snd_usb_endpoint *ep); -int snd_usb_endpoint_activate(struct snd_usb_endpoint *ep); void snd_usb_endpoint_release(struct snd_usb_endpoint *ep); void snd_usb_endpoint_free_all(struct snd_usb_audio *chip); -- 2.34.1

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[PATCH v1 2/2] MAINTAINERS: Add the MAINTAINERS entries for TEXAS INSTRUMENTS AUDIO DRIVERS
by Baojun.Xu 01 Aug '23

01 Aug '23

From: Baojun Xu <baojun.xu(a)ti.com> Add the MAINTAINERS entries for TEXAS INSTRUMENTS AUDIO DRIVERS. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- Change in v1: - Add the MAINTAINERS entries for TI. --- MAINTAINERS | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 53b7ca804465..2c16168167e2 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -21063,6 +21063,38 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ +TEXAS INSTRUMENTS AUDIO (ASoC/HDA) 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 +F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas2562.yaml +F: Documentation/devicetree/bindings/sound/tas2770.yaml +F: Documentation/devicetree/bindings/sound/tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm1681.txt +F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml +F: Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: include/sound/tas2*.h +F: include/sound/tlv320*.h +F: include/sound/tpa6130a2-plat.h +F: sound/soc/codecs/pcm1681.c +F: sound/soc/codecs/pcm1789*.* +F: sound/soc/codecs/pcm179x*.* +F: sound/soc/codecs/pcm186x*.* +F: sound/soc/codecs/pcm3008.* +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/tlv320*.* +F: sound/soc/codecs/tpa6130a2.* + TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> L: dmaengine(a)vger.kernel.org -- 2.34.1

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