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[PATCH 0/6] ASoC: SOF: misc updates for 6.6
by Pierre-Louis Bossart 03 Aug '23

03 Aug '23

Cleanups, move of code to promote reuse and Mediatek minor updates. Curtis Malainey (1): ASoC: SOF: Deprecate invalid enums in IPC3 Pierre-Louis Bossart (3): ASoC: SOF: Intel: start simplify the signature of link_slaves_found() ASoC: soc-acpi: move link_slaves_found() ASoC: soc-acpi: improve log messagesin link_slaves_found() Trevor Wu (2): ASoC: SOF: ipc3: update dai_link_fixup for SOF_DAI_MEDIATEK_AFE ASoC: SOF: mediatek: mt8186 modify dram type as non-cache include/linux/soundwire/sdw.h | 5 ++ include/linux/soundwire/sdw_intel.h | 7 +-- include/sound/soc-acpi.h | 6 ++ include/sound/sof/topology.h | 4 +- sound/soc/soc-acpi.c | 73 +++++++++++++++++++++++ sound/soc/sof/intel/hda.c | 81 +------------------------- sound/soc/sof/ipc3-pcm.c | 17 ++++++ sound/soc/sof/mediatek/mt8186/mt8186.c | 40 +++++++------ 8 files changed, 130 insertions(+), 103 deletions(-) -- 2.39.2

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[PATCH 00/38] ASoC: merge DAI call back functions into ops
by Kuninori Morimoto 03 Aug '23

03 Aug '23

Hi Mark snd_soc_dai_driver has .ops for call back functions (A), but it also has other call back functions (B). It is duplicated and confusable. struct snd_soc_dai_driver { ... ^ int (*probe)(...); | int (*remove)(...); (B) int (*compress_new)(...); | int (*pcm_new)(...); v ... (A) const struct snd_soc_dai_ops *ops; ... } This patch merges (B) into (A). Kuninori Morimoto (38): ASoC: soc-dai.h: merge DAI call back functions into ops ASoC: ti: merge DAI call back functions into ops ASoC: adi: merge DAI call back functions into ops ASoC: adi: merge DAI call back functions into ops ASoC: amd: merge DAI call back functions into ops ASoC: dwc: merge DAI call back functions into ops ASoC: pxa: merge DAI call back functions into ops ASoC: bcm: merge DAI call back functions into ops ASoC: fsl: merge DAI call back functions into ops ASoC: img: merge DAI call back functions into ops ASoC: sof: merge DAI call back functions into ops ASoC: sti: merge DAI call back functions into ops ASoC: stm: merge DAI call back functions into ops ASoC: pxa: merge DAI call back functions into ops ASoC: rsnd: merge DAI call back functions into ops ASoC: qcom: merge DAI call back functions into ops ASoC: ux500: merge DAI call back functions into ops ASoC: sunxi: merge DAI call back functions into ops ASoC: tegra: merge DAI call back functions into ops ASoC: atmel: merge DAI call back functions into ops ASoC: intel: merge DAI call back functions into ops ASoC: meson: merge DAI call back functions into ops ASoC: jz4740: merge DAI call back functions into ops ASoC: cirrus: merge DAI call back functions into ops ASoC: drm/vc4: merge DAI call back functions into ops ASoC: samsung: merge DAI call back functions into ops ASoC: mediatek: merge DAI call back functions into ops ASoC: rockchip: merge DAI call back functions into ops ASoC: uniphier: merge DAI call back functions into ops ASoC: loongson: merge DAI call back functions into ops ASoC: starfive: merge DAI call back functions into ops ASoC: hisilicon: merge DAI call back functions into ops ASoC: codecs/wm*: merge DAI call back functions into ops ASoC: soc-topology: merge DAI call back functions into ops ASoC: codecs/cs47lxx: merge DAI call back functions into ops ASoC: codecs/cx2072x: merge DAI call back functions into ops ASoC: codecs/hdmi-codec: merge DAI call back functions into ops ASoC: soc-dai.h: remove unused call back functions drivers/gpu/drm/vc4/vc4_hdmi.c | 6 +- include/sound/soc-dai.h | 26 +-- sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp/acp-i2s.c | 24 +-- sound/soc/amd/acp/acp-rembrandt.c | 3 - sound/soc/amd/acp/acp-renoir.c | 2 - sound/soc/amd/acp/amd.h | 1 - sound/soc/atmel/atmel-i2s.c | 16 +- sound/soc/atmel/mchp-i2s-mcc.c | 24 +-- sound/soc/atmel/mchp-pdmc.c | 18 +-- sound/soc/atmel/mchp-spdifrx.c | 14 +- sound/soc/atmel/mchp-spdiftx.c | 18 +-- sound/soc/bcm/bcm2835-i2s.c | 24 +-- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/codecs/cs47l15.c | 6 +- sound/soc/codecs/cs47l24.c | 8 +- sound/soc/codecs/cs47l35.c | 8 +- sound/soc/codecs/cs47l85.c | 8 +- sound/soc/codecs/cs47l90.c | 8 +- sound/soc/codecs/cs47l92.c | 6 +- sound/soc/codecs/cx2072x.c | 11 +- sound/soc/codecs/hdmi-codec.c | 44 ++--- sound/soc/codecs/wm2200.c | 12 +- sound/soc/codecs/wm5102.c | 6 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/dwc/dwc-i2s.c | 18 +-- sound/soc/fsl/fsl_asrc.c | 16 +- sound/soc/fsl/fsl_aud2htx.c | 10 +- sound/soc/fsl/fsl_easrc.c | 16 +- sound/soc/fsl/fsl_esai.c | 20 +-- sound/soc/fsl/fsl_micfil.c | 14 +- sound/soc/fsl/fsl_sai.c | 24 +-- sound/soc/fsl/fsl_spdif.c | 17 +- sound/soc/fsl/fsl_ssi.c | 3 +- sound/soc/fsl/fsl_xcvr.c | 16 +- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/hisilicon/hi6210-i2s.c | 2 +- sound/soc/img/img-i2s-in.c | 14 +- sound/soc/img/img-i2s-out.c | 14 +- sound/soc/img/img-parallel-out.c | 14 +- sound/soc/img/img-spdif-in.c | 12 +- sound/soc/img/img-spdif-out.c | 12 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/avs/probes.c | 10 +- sound/soc/intel/catpt/pcm.c | 8 +- sound/soc/intel/keembay/kmb_platform.c | 4 +- sound/soc/jz4740/jz4740-i2s.c | 3 +- sound/soc/loongson/loongson_i2s.c | 16 +- sound/soc/mediatek/mt8195/mt8195-dai-etdm.c | 56 ++++--- sound/soc/meson/aiu-acodec-ctrl.c | 4 +- sound/soc/meson/aiu-codec-ctrl.c | 4 +- sound/soc/meson/aiu-fifo-i2s.c | 3 + sound/soc/meson/aiu-fifo-spdif.c | 3 + sound/soc/meson/aiu.c | 6 - sound/soc/meson/axg-frddr.c | 4 +- sound/soc/meson/axg-pdm.c | 18 +-- sound/soc/meson/axg-spdifin.c | 4 +- sound/soc/meson/axg-tdm-interface.c | 6 +- sound/soc/meson/axg-toddr.c | 4 +- sound/soc/meson/g12a-toacodec.c | 4 +- sound/soc/meson/g12a-tohdmitx.c | 4 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/pxa-ssp.c | 4 +- sound/soc/qcom/lpass-apq8016.c | 4 - sound/soc/qcom/lpass-cpu.c | 40 +++-- sound/soc/qcom/lpass-ipq806x.c | 1 - sound/soc/qcom/lpass-sc7180.c | 5 +- sound/soc/qcom/lpass-sc7280.c | 2 - sound/soc/qcom/lpass.h | 4 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 88 +++++----- sound/soc/qcom/qdsp6/q6asm-dai.c | 6 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 3 - sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/i2s.c | 5 +- sound/soc/samsung/pcm.c | 18 +-- sound/soc/sh/rcar/core.c | 27 ++-- sound/soc/soc-dai.c | 44 ++--- sound/soc/soc-topology.c | 6 +- sound/soc/sof/sof-client-probes.c | 6 +- sound/soc/starfive/jh7110_tdm.c | 16 +- sound/soc/sti/sti_uniperif.c | 6 +- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 18 ++- sound/soc/stm/stm32_spdifrx.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 6 +- sound/soc/sunxi/sun4i-i2s.c | 18 +-- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/tegra/tegra20_ac97.c | 10 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra210_admaif.c | 12 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/ti/davinci-i2s.c | 22 +-- sound/soc/ti/davinci-mcasp.c | 27 ++-- sound/soc/ti/omap-dmic.c | 22 +-- sound/soc/ti/omap-mcbsp.c | 28 ++-- sound/soc/ti/omap-mcpdm.c | 22 +-- sound/soc/uniphier/aio-cpu.c | 161 +++++++++++++++---- sound/soc/uniphier/aio-ld11.c | 62 ++----- sound/soc/uniphier/aio-pxs2.c | 55 +------ sound/soc/uniphier/aio.h | 10 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- 108 files changed, 792 insertions(+), 688 deletions(-) -- 2.25.1

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[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

2 1

[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

2 1

[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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[PATCH v1 1/2] ASoC: tas2783: Add source files for tas2783 soundwire driver
by Baojun.Xu 01 Aug '23

01 Aug '23

From: Baojun Xu <baojun.xu(a)ti.com> Add source file and header file for tas2783 driver. Update Kconfig and Makefile for tas2783 driver. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- Change in v1: - Add tas2783-sdw.c - Add tas2783.h - Update Makefile for tas2783 driver. - Update Kconfig for tas2783 driver. --- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 1166 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 178 +++++ 4 files changed, 1358 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c2de4ee72183..318ad867030c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -240,6 +240,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1762,6 +1763,17 @@ config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2781 speaker amplifier (sdw)" + depends on SOUNDWIRE + select REGMAP + select CRC8 + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + Note the TAS2783 driver implements a flexible and configurable + algo coff setting, for one, two, even multiple TAS2783 chips. + config SND_SOC_TAS571X tristate "Texas Instruments TAS571x power amplifiers" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b48a9a323b84..3298dfa3b9d5 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -276,6 +276,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -648,6 +649,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..93b1e7a1f311 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,1165 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2783 driver implements a flexible and configurable +// algo coefficient setting for single TAS2783 chips. +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/pm_runtime.h> +#include <linux/err.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/init.h> +#include <linux/pm.h> +#include <linux/of.h> +#include <linux/slab.h> +#include <linux/crc8.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/firmware.h> +#include <linux/of_gpio.h> +#include <linux/interrupt.h> +#include <sound/soc.h> +#include <sound/sdw.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/tas2781-tlv.h> + +#include "tas2783.h" + +static const unsigned int tas2783_calibration_reg[] = { + TAS2783_CALIBRATION_RE, + TAS2783_CALIBRATION_RE_LOW, + TAS2783_CALIBRATION_INV_RE, + TAS2783_CALIBRATION_POW, + TAS2783_CALIBRATION_TLIMIT, + 0, +}; + +static const struct reg_default tas2783_reg_defaults[] = { + // Default values for ROM mode. Activated. + { 0x8002, 0x1a}, + { 0x800e, 0x44}, + { 0x800f, 0x40}, + { 0x8097, 0xc8}, + { 0x80b5, 0x74}, + { 0x8099, 0x20}, + { 0xfe8d, 0x0d}, + { 0xfebe, 0x4a}, + { 0x8230, 0x00}, + { 0x8231, 0x00}, + { 0x8232, 0x00}, + { 0x8233, 0x01}, + { 0x8418, 0x00}, + { 0x8419, 0x00}, + { 0x841a, 0x00}, + { 0x841b, 0x00}, + { 0x8428, 0x40}, + { 0x8429, 0x00}, + { 0x842a, 0x00}, + { 0x842b, 0x00}, + { 0x8548, 0x00}, + { 0x8549, 0x00}, + { 0x854a, 0x00}, + { 0x854b, 0x00}, + { 0x8558, 0x40}, + { 0x8559, 0x00}, + { 0x855a, 0x00}, + { 0x855b, 0x00}, + { 0x800a, 0x3a}, + { 0x800e, 0x44}, + { 0x800f, 0x40}, + { 0x805c, 0xd9}, + { 0x8002, 0x00}, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0xc000: // Page 0 ~ 127. + case 0xfe80 ... 0xfeff: // Page 253. + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_UDMPU21, + TAS2783_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, CH_R): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE23, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, TAS2783_SDCA_ENT_PDE22, + TAS2783_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8001: + // Only reset register was volatiled. + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x41008000 + TASDEVICE_REG(0xa1, 0x60, 0x7f), + .reg_defaults = tas2783_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas2783_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .use_single_read = true, + .use_single_write = true, +}; + +static int tas2783_sdw_read(void *context, const void *reg_buf, + const size_t reg_size, void *val_buf, size_t val_size) +{ + struct sdw_slave *peripheral = context; + u8 *buf8 = val_buf; + unsigned int reg, bytes; + int ret; + + reg = le32_to_cpu(*(const __le32 *)reg_buf); + reg += TAS2783_SDW_ADDR_OFFSET; + + while (val_size) { + /* to end of page */ + bytes = SDW_REG_NO_PAGE - (reg & SDW_REGADDR); + if (bytes > val_size) + bytes = val_size; + + ret = sdw_nread_no_pm(peripheral, reg, bytes, buf8); + if (ret != 0) { + dev_err(&peripheral->dev, "Read @%#x..%#x:%d\n", + reg, reg + bytes - 1, ret); + return ret; + } + + swab32_array((u32 *)buf8, bytes / 4); + val_size -= bytes; + reg += bytes; + buf8 += bytes; + } + + return 0; +} + +/* Convert for LSB or MSB. */ +static inline void tas2783_swab_copy( + void *dest, const void *src, size_t nbytes) +{ + u32 *dest32 = dest; + const u32 *src32 = src; + + for (; nbytes > 0; nbytes -= 4) + *dest32++ = swab32(*src32++); +} + +static int tas2783_sdw_gather_write(void *context, const void *reg_buf, + size_t reg_size, const void *val_buf, size_t val_size) +{ + struct sdw_slave *peripheral = context; + const u8 *src_be = val_buf; + u32 val_le_buf[64]; /* Define u32 so it is 32-bit aligned */ + unsigned int reg, bytes; + int ret; + + reg = le32_to_cpu(*(const __le32 *)reg_buf); + + while (val_size) { + /* to end of page */ + bytes = SDW_REG_NO_PAGE - (reg & SDW_REGADDR); + if (bytes > val_size) + bytes = val_size; + if (bytes > sizeof(val_le_buf)) + bytes = sizeof(val_le_buf); + + tas2783_swab_copy(val_le_buf, src_be, bytes); + + ret = sdw_nwrite_no_pm(peripheral, reg, bytes, + (u8 *)val_le_buf); + if (ret != 0) { + dev_err(&peripheral->dev, "Write @%#x..%#x:%d\n", + reg, reg + bytes - 1, ret); + return ret; + } + + val_size -= bytes; + reg += bytes; + src_be += bytes; + } + + return 0; +} + +static int tas2783_sdw_write( + void *context, const void *val_buf, size_t val_size) +{ + const u8 *src_buf = val_buf; + + /* First word of val_buf contains the destination address */ + return tas2783_sdw_gather_write(context, &src_buf[0], 4, + &src_buf[4], val_size - 4); +} + +static int tasdevice_clamp(int val, int max, unsigned int invert) +{ + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + return val; +} + +/* + * Registers are big-endian on I2C and SPI but little-endian on SoundWire. + * Exported firmware controls are big-endian on I2C/SPI but little-endian + * on SoundWire. + * Firmware files are always big-endian and are opaque blobs. + * Present a big-endian regmap and hide the endianness swap, + * so that the ALSA byte controls always have the same byte order, + * and firmware file blobs can be written verbatim. + */ +static const struct regmap_bus tas2783_regmap_bus_sdw = { + .read = tas2783_sdw_read, + .write = tas2783_sdw_write, + .gather_write = tas2783_sdw_gather_write, + .reg_format_endian_default = REGMAP_ENDIAN_LITTLE, + .val_format_endian_default = REGMAP_ENDIAN_BIG, +}; + +static int tas2783_digital_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + unsigned int invert = mc->invert; + int max = mc->max; + int ret = 0; + + /* Read the primary device as the whole */ + ret = tasdevice_dev_read(tas_dev, &mc->reg, (unsigned int *)&val); + dev_dbg(tas_dev->dev, "%s, get digital vol %d from %x with %d\n", + __func__, val, mc->reg, ret); + if (ret) { + dev_err(tas_dev->dev, "%s, get digital vol error %x.\n", + __func__, ret); + goto out; + } + + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + ucontrol->value.integer.value[0] = val; + +out: + return ret; +} + +static int tas2783_digital_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val, ret = 0; + unsigned int invert = mc->invert; + int max = mc->max; + unsigned char value = 0; + + val = ucontrol->value.integer.value[0]; + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + + value = (u8)val; + ret = sdw_nwrite_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + dev_dbg(tas_dev->dev, "%s, Put digital vol %d into %x.\n", + __func__, val, mc->reg); + + return 1; +} + +static int tas2783_amp_getvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val; + unsigned char mask = 0; + int max = mc->max; + int ret = 0; + unsigned int invert = mc->invert; + unsigned char value = 0; + + /* Read the primary device */ + ret = sdw_nread_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + dev_dbg(tas_dev->dev, "%s, get AMP vol %d from %x with %d\n", + __func__, value, mc->reg, ret); + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = value; + val = (val & mask) >> mc->shift; + if (val > max) + val = max; + if (invert) + val = max - val; + if (val < 0) + val = 0; + ucontrol->value.integer.value[0] = val; + + return ret; +} + +static int tas2783_amp_putvol(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec + = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int val = 0, ret = 0, max = mc->max; + unsigned char mask = 0, value = 0; + unsigned int invert = mc->invert; + + mask = (1 << fls(mc->max)) - 1; + mask <<= mc->shift; + val = tasdevice_clamp(ucontrol->value.integer.value[0], max, + invert); + + ret = sdw_nread_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Read @%x.%x:%d\n", + mc->reg, value, ret); + return ret; + } + value &= (~TAS2783_AMP_LEVEL_MASK); + value |= (val << mc->shift); + ret = sdw_nwrite_no_pm(tas_dev->sdw_slave, mc->reg, 1, &value); + if (ret != 0) { + dev_err(tas_dev->dev, "Write @%#x..%#x:%d\n", + mc->reg, val, ret); + return ret; + } + dev_dbg(tas_dev->dev, "amp %x into 0x%08x by %lx shift %d.\n", + val, mc->reg, TAS2783_AMP_LEVEL_MASK, mc->shift); + + return ret; +} + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_EXT_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, 0, tas2783_amp_getvol, + tas2783_amp_putvol, amp_vol_tlv), + SOC_SINGLE_RANGE_EXT_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, tas2783_digital_getvol, + tas2783_digital_putvol, dvc_tlv), +}; + +static void tas2783_apply_calib( + struct tasdevice_priv *tas_priv, unsigned int *cali_data) +{ + unsigned int reg_address = 0, i = 0; + unsigned int *reg_start; + + if (!tas_priv->sdw_slave) + return; + if ((tas_priv->sdw_slave->id.unique_id < TAS2783_DEVICE_ID_START) + || (tas_priv->sdw_slave->id.unique_id > TAS2783_DEVICE_ID_END)) + return; + reg_start = cali_data+(tas_priv->sdw_slave->id.unique_id - + TAS2783_DEVICE_ID_START)*TAS2783_CALIBRATION_SIZE; + while (tas2783_calibration_reg[i] != 0) { + reg_address = tas2783_calibration_reg[i]; + tas2783_sdw_gather_write(tas_priv->sdw_slave, + &reg_address, 4, reg_start+i, 4); + } +} + +static int tas2783_calibration(struct tasdevice_priv *tas_priv) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct tm *tm = &tas_priv->tm; + unsigned int attr, crc; + unsigned int *tmp_val; + efi_status_t status; + + tas_priv->cali_data.total_sz = 128; + /* Get real size of UEFI variable */ + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes with result : %ld\n", + tas_priv->cali_data.total_sz, status); + if (status == EFI_BUFFER_TOO_SMALL) { + status = efi.get_variable(efi_name, &efi_guid, &attr, + &tas_priv->cali_data.total_sz, + tas_priv->cali_data.data); + dev_dbg(tas_priv->dev, "cali get %lx bytes result:%ld\n", + tas_priv->cali_data.total_sz, status); + } + for (int i = 0; i < tas_priv->cali_data.total_sz; i += 4) { + dev_dbg(tas_priv->dev, "cali get %02x %02x %02x %02x", + tas_priv->cali_data.data[i], + tas_priv->cali_data.data[i+1], + tas_priv->cali_data.data[i+2], + tas_priv->cali_data.data[i+3]); + } + + tmp_val = (unsigned int *)tas_priv->cali_data.data; + + crc = crc32(~0, tas_priv->cali_data.data, 84) ^ ~0; + dev_dbg(tas_priv->dev, "cali crc 0x%08x PK tmp_val 0x%08x\n", + crc, tmp_val[21]); + + if (crc == tmp_val[21]) { + time64_to_tm(tmp_val[20], 0, tm); + dev_dbg(tas_priv->dev, "%4ld-%2d-%2d, %2d:%2d:%2d\n", + tm->tm_year, tm->tm_mon, tm->tm_mday, + tm->tm_hour, tm->tm_min, tm->tm_sec); + tas2783_apply_calib(tas_priv, tmp_val); + } else { + dev_dbg(tas_priv->dev, "CRC error!\n"); + tas_priv->cali_data.total_sz = 0; + } + + return 0; +} + +static void tasdevice_rca_ready(const struct firmware *fmw, void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct MYSWFTFile *p; + unsigned char *buf = NULL, value_sdw; + int offset = 0, i = 0, num_files = 0, img_sz = 0; + int ret = 0; + + mutex_lock(&tas_dev->codec_lock); + + if (!fmw || !fmw->data) { + dev_err(tas_dev->dev, + "Failed to read %s, no side - effect on driver running\n", + tas_dev->rca_binaryname); + ret = -1; + goto out; + } + buf = (unsigned char *)fmw->data; + + img_sz = SMS_HTONL(buf[offset+3], buf[offset + 2], + buf[offset + 1], buf[offset]); + dev_dbg(tas_dev->dev, "Got %x:%lx.\n", img_sz, fmw->size); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_err(tas_dev->dev, + "Size not match, %d %u", (int)fmw->size, img_sz); + ret = -1; + goto out; + } + + while ((offset < img_sz) && (num_files < MAX_NO_FILES)) { + p = (struct MYSWFTFile *)(buf + offset); + tas_dev->m_swftFile[num_files].m_vendor_id = + p->m_vendor_id; + tas_dev->m_swftFile[num_files].m_file_id = p->m_file_id; + tas_dev->m_swftFile[num_files].m_version = p->m_version; + tas_dev->m_swftFile[num_files].m_Length = p->m_Length; + tas_dev->m_swftFile[num_files].m_downloadAddr = + p->m_downloadAddr; + tas_dev->m_swftFile[num_files].m_startAdress = + ((char *)p) + sizeof(unsigned int)*5; + + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + tas_dev->m_swftFile[num_files].m_downloadAddr, + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_startAdress); + dev_dbg(tas_dev->dev, "Write %x into %x result : %d\n", + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_downloadAddr, + ret); + + dev_dbg(tas_dev->dev, "Write %d :%x:%x:%x:%x:%x:%x.\n", + num_files, + tas_dev->m_swftFile[num_files].m_vendor_id, + tas_dev->m_swftFile[num_files].m_file_id, + tas_dev->m_swftFile[num_files].m_version, + tas_dev->m_swftFile[num_files].m_Length, + tas_dev->m_swftFile[num_files].m_downloadAddr, + tas_dev->m_swftFile[num_files].m_startAdress[0]); + + ret = sdw_nread_no_pm(tas_dev->sdw_peripheral, + tas_dev->m_swftFile[num_files].m_downloadAddr, + 1, &value_sdw); + dev_dbg(tas_dev->dev, "Got block %d :%x.\n", + num_files, value_sdw); + + offset += sizeof(unsigned int)*5 + p->m_Length; + num_files++; + } + + i = 0x50; + while (i < 0x56) { + ret = sdw_nread_no_pm(tas_dev->sdw_peripheral, i, 1, + &value_sdw); + dev_dbg(tas_dev->dev, "Got device id:%x\n", value_sdw); + i++; + } + tas2783_calibration(tas_dev); + +out: + mutex_unlock(&tas_dev->codec_lock); + if (fmw) + release_firmware(fmw); +} + +static void tasdevice_config_info_remove(void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + dev_dbg(tas_dev->dev, "%s was performed\n", __func__); +} + +static int tasdevice_dapm_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *codec = + snd_soc_dapm_to_component(w->dapm); + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int state = 1; + + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + dev_dbg(tas_dev->dev, "dapm event : %x\n", event); + if (event == SND_SOC_DAPM_PRE_PMD) + state = 0; + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + return 0; +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, + 0, 0, tasdevice_dapm_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"SPK", NULL, "ASI"}, + {"OUT", NULL, "SPK"}, + {"ASI OUT", NULL, "DMIC"} +}; + +static int tasdevice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + int ret = 0; + + return ret; +} + +static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct tasdevice_priv *tas_dev = + snd_soc_dai_get_drvdata(codec_dai); + + tas_dev->sysclk = freq; + + return 0; +} + +static int tasdevice_set_sdw_stream( + struct snd_soc_dai *dai, void *sdw_stream, int direction) +{ + struct sdw_stream_data *stream; + + if (!sdw_stream) + return 0; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = sdw_stream; + + /* Use tx_mask or rx_mask to set dma_data */ + snd_soc_dai_dma_data_set(dai, direction, stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_data *stream; + int retval; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -EINVAL; + + if (!tasdevice->sdw_slave) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + retval = sdw_stream_add_slave(tasdevice->sdw_slave, + &stream_config, &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tasdevice->sdw_slave) + return -EINVAL; + + sdw_stream_remove_slave(tasdevice->sdw_slave, stream->sdw_stream); + return 0; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tasdevice = + snd_soc_component_get_drvdata(component); + int ret = 0; + unsigned char value_sdw = 0; + + dev_dbg(tasdevice->dev, "%s: mute %d, %d\n", + __func__, mute, direction); + if (mute == 0) {// Unmute. + // FU21 Unmute + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 1, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %lx %x\n", + SDW_SDCA_CTL(1, 1, 1, 0), ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40402090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + + // FU23 Unmute + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 2, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %lx %x\n", + SDW_SDCA_CTL(1, 2, 1, 0), ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400108, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + value_sdw = 0; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x8002, 1, &value_sdw); + dev_dbg(tasdevice->dev, "Unmuted %x\n", ret); + } else { + // FU21 mute + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 1, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %lx %x\n", + SDW_SDCA_CTL(1, 1, 1, 0), ret); + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40402090, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + // FU23 mute + value_sdw = 1; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + SDW_SDCA_CTL(1, 2, 1, 0), 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %lx %x\n", + SDW_SDCA_CTL(1, 2, 1, 0), ret); + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x40400108, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + value_sdw = 0x1a; + ret = sdw_nwrite_no_pm(tasdevice->sdw_peripheral, + 0x8002, 1, &value_sdw); + dev_dbg(tasdevice->dev, "muted %x\n", ret); + } + + if (!tasdevice->sdw_slave) + return -EINVAL; + + return 0; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .startup = tasdevice_startup, + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .set_sysclk = tasdevice_set_dai_sysclk, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783_codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = TASDEVICE_RATES, + .formats = TASDEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static void tas2783_reset(struct tasdevice_priv *tas_dev) +{ + int ret = 0; + unsigned char value_sdw; + + value_sdw = TAS2873_REG_SWRESET_RESET; + ret = sdw_nwrite_no_pm(tas_dev->sdw_peripheral, + TAS2873_REG_SWRESET, 1, &value_sdw); + dev_dbg(tas_dev->dev, "%s TAS2783 was reseted.\n", __func__); + usleep_range(1000, 1050); +} + +static int tasdevice_codec_probe(struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + int ret = 0; + + dev_dbg(tas_dev->dev, "%s called for TAS2783 start.\n", + __func__); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + + crc8_populate_msb(tas_dev->crc8_lkp_tbl, + TASDEVICE_CRC8_POLYNOMIAL); + tas_dev->codec = codec; + + scnprintf(tas_dev->rca_binaryname, 64, "MY_SWFT_x%01x.bin", + tas_dev->sdw_peripheral->id.unique_id); + + ret = request_firmware_nowait(THIS_MODULE, FW_ACTION_UEVENT, + tas_dev->rca_binaryname, tas_dev->dev, GFP_KERNEL, + tas_dev, tasdevice_rca_ready); + dev_dbg(tas_dev->dev, + "%s: request_firmware %x open status: 0x%08x\n", + __func__, tas_dev->sdw_peripheral->id.unique_id, ret); + + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); + + tas2783_reset(tas_dev); + + dev_dbg(tas_dev->dev, "%s was called end.\n", __func__); + return ret; +} + +static void tasdevice_deinit(void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + + tasdevice_config_info_remove(tas_dev); +} + +static void tasdevice_codec_remove( + struct snd_soc_component *codec) +{ + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(codec); + /* Codec Lock Hold */ + mutex_lock(&tas_dev->codec_lock); + tasdevice_deinit(tas_dev); + /* Codec Lock Release*/ + mutex_unlock(&tas_dev->codec_lock); +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_codec_probe, + .remove = tasdevice_codec_remove, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +int tasdevice_dev_read(struct tasdevice_priv *tas_dev, unsigned int *reg, + unsigned int *val) +{ + int ret = 0; + + mutex_lock(&tas_dev->dev_lock); + ret = tas2783_sdw_read(tas_dev->sdw_peripheral, + reg, 4, (void *)val, 4); + dev_dbg(tas_dev->dev, "%s, read %x from reg %x. Result is (%d)\n", + __func__, *val, *reg, ret); + mutex_unlock(&tas_dev->dev_lock); + return ret; +} + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret = 0; + + tas_dev->cur_prog = -1; + tas_dev->cur_conf = -1; + + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + mutex_init(&tas_dev->dev_lock); + + dev_set_drvdata(tas_dev->dev, tas_dev); + + mutex_init(&tas_dev->codec_lock); + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, + tasdevice_dai_driver, ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:0x%08x\n", + __func__, ret); + } + dev_dbg(tas_dev->dev, "%s was called for TAS2783.\n", __func__); + + return ret; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = + SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x04; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + dev_dbg(&slave->dev, "%s was performed.\n", __func__); + + return 0; +} + +static int tasdevice_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(dev); + + if (tasdevice->hw_init) + return 0; + + if (tasdevice->first_hw_init) { + regcache_cache_only(tasdevice->regmap, false); + regcache_cache_bypass(tasdevice->regmap, true); + } else { + /* PM runtime is only enabled when + * a Slave reports as Attached + * set autosuspend parameters + */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + if (tasdevice->first_hw_init) { + regcache_cache_bypass(tasdevice->regmap, false); + regcache_mark_dirty(tasdevice->regmap); + } else + tasdevice->first_hw_init = true; + + /* Mark Slave initialization complete */ + tasdevice->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tasdevice = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tasdevice->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tasdevice->hw_init = false; + + /* Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tasdevice->hw_init || tasdevice->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static void tasdevice_remove(struct tasdevice_priv *tas_dev) +{ + snd_soc_unregister_component(tas_dev->dev); + if (gpio_is_valid(tas_dev->rst_gpio)) + gpio_free(tas_dev->rst_gpio); + + if (gpio_is_valid(tas_dev->irq_info.irq_gpio)) + gpio_free(tas_dev->irq_info.irq_gpio); + + mutex_destroy(&tas_dev->dev_lock); + mutex_destroy(&tas_dev->codec_lock); +} + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + int ret = 0; + + dev_dbg(dev, "%s was called for TAS2783.\n", __func__); + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) { + ret = -ENOMEM; + goto out; + } + tas_dev->dev = &peripheral->dev; + tas_dev->sdw_peripheral = peripheral; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_slave = peripheral; + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_dbg(dev, "%d chip id %x for TAS2783.\n", + peripheral->id.unique_id, tas_dev->chip_id); + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init(dev, &tas2783_regmap_bus_sdw, + peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) { + ret = PTR_ERR(tas_dev->regmap); + dev_err(dev, "Failed devm_regmap_init: %d\n", ret); + goto out; + } + ret = tasdevice_init(tas_dev); + +out: + if (ret < 0 && tas_dev != NULL) + tasdevice_remove(tas_dev); + + dev_dbg(dev, "%s was called for TAS2783.\n", __func__); + return ret; + +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = + dev_get_drvdata(&peripheral->dev); + + if (tas_dev) + tasdevice_remove(tas_dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..0fa5b3855b51 --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,178 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier + * + * Copyright (C) 2023 Texas Instruments Incorporated + * https://www.ti.com + * + * The TAS2783 driver implements a flexible and configurable + * algo coff setting for single TAS2783 chips. + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) +#define TAS2783_SDW_ADDR_OFFSET 0x8000 + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 +\ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1A) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x0d, 0x3c) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +#define TAS2783_CALIBRATION_SIZE 5 // int data number. + +#define TAS2783_DEVICE_ID_START 0x08 // Unique id start +#define TAS2783_DEVICE_ID_END 0x0F // Unique id end + +/* TAS2783 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_PDE23 0x0C +#define TAS2783_SDCA_ENT_PDE22 0x0B +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_UDMPU21 0x10 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define CH_L 1 +#define CH_R 2 + +#define TASDEVICE_CRC8_POLYNOMIAL 0x4d + +#define MAX_NO_FILES 100 +#define MAX_CALIBRATION_DATA_SIZE 252 + +#define SMS_HTONL(a, b, c, d) ((((a)&0x000000FF)<<24) | \ + (((b)&0x000000FF)<<16) | \ + (((c)&0x000000FF)<<8) | \ + ((d)&0x000000FF)) + +enum channel { + top_left_Chn, + top_right_chn, + bottom_left_Chn, + bottom_right_chn, + max_chn, +}; + +struct tasdevice { + unsigned int dev_addr; + unsigned int err_code; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; + bool is_calibrated_data_loaded; + unsigned char cur_book; + struct tasdevice_fw *cali_data_fmw; +}; + +struct MYSWFTFile { + unsigned int m_vendor_id; + unsigned int m_file_id; + unsigned int m_version; + unsigned int m_Length; + unsigned int m_downloadAddr; + unsigned char *m_startAdress; +}; + +/* + * This item is used to store the generic i2c address of + * all the tas2781 devices for I2C broadcast during the multi-device + * writes, useless in mono case. + */ +struct global_addr { + unsigned char cur_book; + unsigned int dev_addr; + int ref_cnt; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[MAX_CALIBRATION_DATA_SIZE]; +}; + +struct tas_control { + struct snd_kcontrol_new *tasdevice_profile_controls; + int nr_controls; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; +}; + +struct tasdevice_priv { + struct device *dev; + struct snd_soc_component *component; + struct sdw_slave *sdw_slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct mutex codec_lock; + struct mutex dev_lock; + int rst_gpio; + struct tasdevice tasdevice[max_chn]; + struct tasdevice_fw *fmw; + struct tasdevice_irqinfo irq_info; + struct tas_control tas_ctrl; + struct sdw_slave *sdw_peripheral; + struct global_addr glb_addr; + struct calibration_data cali_data; + struct tm tm; + int cur_prog; + int cur_conf; + unsigned int chip_id; + bool hw_init; + bool first_hw_init; + unsigned char *bq_params; + unsigned int magic_num; + unsigned char ndev; + unsigned char dev_name[32]; + unsigned char rca_binaryname[64]; + unsigned char coef_binaryname[64]; + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + + struct MYSWFTFile m_swftFile[MAX_NO_FILES]; + + void *codec; + unsigned int sysclk; + bool is_glb_calibrated_data_loaded; +}; + +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +int tasdevice_dev_read(struct tasdevice_priv *tasdevice, + unsigned int *reg, unsigned int *value); + +#endif /*__TAS2783_H__ */ -- 2.34.1

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[PATCH v4 0/1] Intel: sof_rt5682: remove quirk flag
by Brent Lu 01 Aug '23

01 Aug '23

We add a helper function to detect amplifier number according to device instance in ACPI table so the SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag and a dmi quirk for 4-amplifier configuration could be safely removed. Also refactor the max_98390_hw_params() function to use an array to handle the TDM parameter. Amplifier number detection and TDM parameter are tested on two Chromebooks. One with 2 MAX98390 and one with 4 MAX98390 amplifier. V2 Changes: - fix some typos in cover letter - use for_each_acpi_dev_match() macro to simplify the codec - use 'unsign int' for codec number variable V3 Changes: - replace pr_err() with dev_err() V4 Changes: - merge two patches to avoid build break - remove redundant assignment in max_98390_hw_params() - refactor MAX_98390_DEVX_NAME macros Brent Lu (2): ASoC: Intel: maxim-common: get codec number from ACPI ASoC: Intel: sof_rt5682: remove SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag Brent Lu (1): ASoC: Intel: maxim-common: get codec number from ACPI table sound/soc/intel/boards/sof_maxim_common.c | 170 +++++++++++++--------- sound/soc/intel/boards/sof_maxim_common.h | 21 ++- sound/soc/intel/boards/sof_rt5682.c | 37 +---- 3 files changed, 111 insertions(+), 117 deletions(-) -- 2.34.1

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[PATCH 0/8] ASoC/SOF/Intel/AMD: cleanups for GCC11 -fanalyzer checks
by Pierre-Louis Bossart 01 Aug '23

01 Aug '23

GCC11 provides an '-fanalyzer' static analysis option which does not provide too many false-positives. This patch cleans-up known problematic code paths to help enable this capability in CI. We've used this for about a month already. Pierre-Louis Bossart (8): ASoC: SOF: sof-client-probes-ipc4: add checks to prevent static analysis warnings ASoC: SOF: ipc3: add checks to prevent static analysis warnings ASoC: SOF: topology: simplify code to prevent static analysis warnings ASoC: SOF: imx: remove error checks on NULL ipc ASoC: SOF: mediatek: remove error checks on NULL ipc ASoC: Intel: bdw_rt286: add checks to avoid static analysis warnings ASoC: Intel: atom: remove static analysis false positive ASoC: amd: acp5x-mach:add checks to avoid static analysis warnings include/linux/firmware/imx/dsp.h | 6 ------ include/linux/firmware/mediatek/mtk-adsp-ipc.h | 6 ------ sound/soc/amd/vangogh/acp5x-mach.c | 3 +++ sound/soc/intel/atom/sst/sst_stream.c | 13 +++++++------ sound/soc/intel/boards/bdw_rt286.c | 6 ++++++ sound/soc/sof/ipc3.c | 2 +- sound/soc/sof/sof-client-probes-ipc4.c | 9 +++++++++ sound/soc/sof/topology.c | 5 +++-- 8 files changed, 29 insertions(+), 21 deletions(-) -- 2.39.2

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[PATCH v4] um/drivers: fix hostaudio build errors
by Randy Dunlap 01 Aug '23

01 Aug '23

Use "select"s to ensure that the required kconfig symbols are set as expected. Drop HOSTAUDIO since it is now equivalent to UML_SOUND. Allow SOUND with UML regardless of HAS_IOMEM. Otherwise there is a kconfig warning for unmet dependencies. (This was not an issue when SOUND was defined in arch/um/drivers/Kconfig. I have done 50 randconfig builds and didn't find any issues.) This fixes build errors when CONFIG_SOUND is not set: ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_cleanup_module': hostaudio_kern.c:(.exit.text+0xa): undefined reference to `unregister_sound_mixer' ld: hostaudio_kern.c:(.exit.text+0x15): undefined reference to `unregister_sound_dsp' ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_init_module': hostaudio_kern.c:(.init.text+0x19): undefined reference to `register_sound_dsp' ld: hostaudio_kern.c:(.init.text+0x31): undefined reference to `register_sound_mixer' ld: hostaudio_kern.c:(.init.text+0x49): undefined reference to `unregister_sound_dsp' and this kconfig warning: WARNING: unmet direct dependencies detected for SOUND Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Fixes: d886e87cb82b ("sound: make OSS sound core optional") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Closes: lore.kernel.org/r/202307141416.vxuRVpFv-lkp@intel.com Cc: Richard Weinberger <richard(a)nod.at> Cc: Anton Ivanov <anton.ivanov(a)cambridgegreys.com> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: linux-um(a)lists.infradead.org Cc: Tejun Heo <tj(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Nick Desaulniers <ndesaulniers(a)google.com> Cc: Nicolas Schier <nicolas(a)fjasle.eu> Cc: linux-kbuild(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- v2: don't delete the HOSTAUDIO Kconfig entry (Masahiro) v3: drop HOSTAUDIO and use CONFIG_UML_SOUND for it in Makefile (Takashi); add SOUND depends on "|| UML" to HAS_IOMEM v4: use depends on instead of select for SOUND (Masahiro); use Closes: instead of Link: arch/um/drivers/Kconfig | 16 +++------------- arch/um/drivers/Makefile | 2 +- sound/Kconfig | 2 +- 3 files changed, 5 insertions(+), 15 deletions(-) diff -- a/arch/um/drivers/Kconfig b/arch/um/drivers/Kconfig --- a/arch/um/drivers/Kconfig +++ b/arch/um/drivers/Kconfig @@ -111,24 +111,14 @@ config SSL_CHAN config UML_SOUND tristate "Sound support" + depends on SOUND + select SOUND_OSS_CORE help This option enables UML sound support. If enabled, it will pull in - soundcore and the UML hostaudio relay, which acts as a intermediary + the UML hostaudio relay, which acts as a intermediary between the host's dsp and mixer devices and the UML sound system. It is safe to say 'Y' here. -config SOUND - tristate - default UML_SOUND - -config SOUND_OSS_CORE - bool - default UML_SOUND - -config HOSTAUDIO - tristate - default UML_SOUND - endmenu menu "UML Network Devices" diff -- a/sound/Kconfig b/sound/Kconfig --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only menuconfig SOUND tristate "Sound card support" - depends on HAS_IOMEM + depends on HAS_IOMEM || UML help If you have a sound card in your computer, i.e. if it can say more than an occasional beep, say Y. diff -- a/arch/um/drivers/Makefile b/arch/um/drivers/Makefile --- a/arch/um/drivers/Makefile +++ b/arch/um/drivers/Makefile @@ -54,7 +54,7 @@ obj-$(CONFIG_UML_NET) += net.o obj-$(CONFIG_MCONSOLE) += mconsole.o obj-$(CONFIG_MMAPPER) += mmapper_kern.o obj-$(CONFIG_BLK_DEV_UBD) += ubd.o -obj-$(CONFIG_HOSTAUDIO) += hostaudio.o +obj-$(CONFIG_UML_SOUND) += hostaudio.o obj-$(CONFIG_NULL_CHAN) += null.o obj-$(CONFIG_PORT_CHAN) += port.o obj-$(CONFIG_PTY_CHAN) += pty.o

2 1

Audio output heavily distorted on Xen PV dom0 if not recording (Intel ADL)
by Marek Marczykowski-Górecki 01 Aug '23

01 Aug '23

Hi, I have a rather interesting issue on HP EliteBook 1040, with Intel ADL HDA. Sometimes (often) audio output is heavily distorted, but then opening pavucontrol or just launching `parecord /dev/null` fixes the issue instantly. But, when the mixer or recording is closed, then few seconds later issue comes back. You can hear this at https://cloud.marmarek.net/s/46prqZnwrqDkBdD Initially recording was running, then about 10s stopped and you can hear the issue few seconds later. Then I starter recording again, and the issue is gone. This is how it should sound: https://youtube.com/watch?v=4Tr0otuiQuU I do not see any kernel messages when issue happens. The issue happens on Linux 6.1.26, 6.2.10, and 6.3.2. I haven't tested other versions. It happens only when running under Xen (in PV dom0). The very same Linux started directly works fine. Full dmesg of both native and Xen boots is at https://gist.github.com/marmarek/548ee79282446344a172109bcd943930 When running Linux 6.3.2 there, I noticed that just after boot first playback is okay, but if I start and stop recording it breaks. And then after some time of idle it fixes itself again. I have not observed this behavior on 6.2.10 (it was broken initially too), but I could be also doing some things differently. I have few other ADL-based systems (but not HP), and they do not show this issue, only this HP laptop is affected. Example dmesg from another ADL-based system: https://gist.github.com/marmarek/38c5ba3ec58f9300fbc1842ddc9fa4f5 I'm not sure if that's the relevant difference, but comparing dmesg from working and not working systems, I see those on boot for the problematic case: snd_hda_intel 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver sof-audio-pci-intel-tgl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if info 0x040100 sof-audio-pci-intel-tgl 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver sof-audio-pci-intel-tgl 0000:00:1f.3: enabling device (0000 -> 0002) xen: registering gsi 16 triggering 0 polarity 1 Already setup the GSI :16 sof-audio-pci-intel-tgl 0000:00:1f.3: DSP detected with PCI class/subclass/prog-if 0x040100 sof-audio-pci-intel-tgl 0000:00:1f.3: bound 0000:00:02.0 (ops i915_audio_component_bind_ops [i915]) sof-audio-pci-intel-tgl 0000:00:1f.3: use msi interrupt mode sof-audio-pci-intel-tgl 0000:00:1f.3: hda codecs found, mask 5 sof-audio-pci-intel-tgl 0000:00:1f.3: using HDA machine driver skl_hda_dsp_generic now sof-audio-pci-intel-tgl 0000:00:1f.3: DMICs detected in NHLT tables: 2 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:0 sof-audio-pci-intel-tgl 0000:00:1f.3: unknown sof_ext_man header type 3 size 0x30 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware info: version 2:2:0-57864 sof-audio-pci-intel-tgl 0000:00:1f.3: Firmware: ABI 3:22:1 Kernel ABI 3:23:0 sof-audio-pci-intel-tgl 0000:00:1f.3: Topology: ABI 3:22:1 Kernel ABI 3:23:0 skl_hda_dsp_generic skl_hda_dsp_generic: ASoC: Parent card not yet available, widget card binding deferred cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: Firmware version: 3 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: cirrus/cs35l41-dsp1-spk-prot-103c8974.wmfw: Fri 27 Aug 2021 14:58:19 W. Europe Daylight Time cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: Firmware: 400a4 vendor: 0x2 v0.43.1, 2 algorithms cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: 0: ID cd v29.63.1 XM@94 YM@e cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: 1: ID f20b v0.1.0 XM@176 YM@0 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.0: DSP1: spk-prot: E:\Amp Tuning\HP\840\0527\103C8972_220527.bin snd_hda_codec_realtek ehdaudio0D0: bound spi0-CSC3551:00-cs35l41-hda.0 (ops cs35l41_hda_comp_ops [snd_hda_scodec_cs35l41]) cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: Firmware version: 3 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: cirrus/cs35l41-dsp1-spk-prot-103c8974.wmfw: Fri 27 Aug 2021 14:58:19 W. Europe Daylight Time cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: Firmware: 400a4 vendor: 0x2 v0.43.1, 2 algorithms cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: 0: ID cd v29.63.1 XM@94 YM@e cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: 1: ID f20b v0.1.0 XM@176 YM@0 cs35l41-hda spi0-CSC3551:00-cs35l41-hda.1: DSP1: spk-prot: E:\Amp Tuning\HP\840\0527\103C8972_220527.bin snd_hda_codec_realtek ehdaudio0D0: bound spi0-CSC3551:00-cs35l41-hda.1 (ops cs35l41_hda_comp_ops [snd_hda_scodec_cs35l41]) Any ideas? -- Best Regards, Marek Marczykowski-Górecki Invisible Things Lab

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Re: alsa-project-Domains updated and renewal (Urgent to Principal/CEO)(Case#220119)
by auditing 01 Aug '23

01 Aug '23

Dear Principal, Sorry to trouble you! We did not receive your reply until now. Do you mean that you want to give up the registration? If so, we will sign the registration agreement with that company. If you have any questions, contact me freely. Best Regards, Joe Luo ______________________________________________________________________ Dear Principal, I'm Joe Luo from HK TLD Creating Limited. The domains "alsa-project.cn/.hk......" you registered with us had been expired, would you like to renew them ? If we don't renew in time, these domains will be available for re-registration by anyone.If you choose to renew, we will send you the renewal price/documents and arrange the next step for you in time. Since this matter is very urgent and important. I hope you give me a reply asap. Best Regards, Joe Luo Manager of Auditing Department ______________________________________________________________________ HK TLD Creating Limited Web: www.tldservice.org Tel: +852 3069 7574 Fax: +852 3069 7409 E: joe(a)hkcreating.com Add: Commercial Building,17-19Prat Auenue,Tsimshatsui,Kowloon,HongKong.

1 0

[PATCH 0/9] ALSA: hda/cs35l56: Various bugfixes
by Richard Fitzgerald 01 Aug '23

01 Aug '23

A collection of various bugfixes to the cs35l56 driver. Richard Fitzgerald (9): ALSA: hda/cs35l56: Complete firmware reboot before calling cs_dsp_run() ALSA: hda/cs35l56: Do not mark cache dirty after REINIT ALSA: hda/cs35l56: Call cs_dsp_power_down() before reloading firmware ALSA: hda/cs35l56: Always power-up and start cs_dsp ALSA: hda/cs35l56: Call cs_dsp_power_down() before calling cs_dsp_remove() ALSA: hda/cs35l56: cs_dsp_power_down() on cs35l56_hda_fw_load() error path ALSA: hda/cs35l56: Do not download firmware over existing RAM firmware ALSA: hda/cs35l56: Fail if .bin not found and firmware not patched ALSA: hda/cs35l56: Reject I2C alias addresses sound/pci/hda/cs35l56_hda.c | 91 ++++++++++++++++++++++++++----------- 1 file changed, 65 insertions(+), 26 deletions(-) -- 2.30.2

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[PATCH 0/7] usb: Add USB MIDI 2.0 Gadget Function Driver
by Takashi Iwai 01 Aug '23

01 Aug '23

Hi, This is a patch set for adding a USB gadget function driver for USB MIDI 2.0 interface. The MIDI 2.0 support in the host side was already enabled since 6.5 kernel in snd-usb-audio driver. Now, this provides a similar gadget functionality using the existing ALSA UMP helper functions. The gadget driver provides both MIDI 2.0 and MIDI 1.0 fallback interfaces with one or more loopback UMP rawmidi devices. The configfs implementation is flexible enough to allow users to set up multiple UMP Endpoints and Function Blocks. The first patch adds the bare f_midi2 driver without configfs support, followed by the configfs support, MIDI 1.0 interface support and a few other extensions and documentation. Takashi === Takashi Iwai (7): usb: gadget: Add support for USB MIDI 2.0 function driver usb: gadget: midi2: Add configfs support usb: gadget: midi2: Dynamically create MIDI 1.0 altset descriptors usb: gadget: midi2: MIDI 1.0 interface (altset 0) support usb: gadget: midi2: Add testing documentation usb: gadget: midi2: Add "Operation Mode" control usb: gadget: midi2: More flexible MIDI 1.0 configuration .../ABI/testing/configfs-usb-gadget-midi2 | 54 + Documentation/usb/gadget-testing.rst | 154 + drivers/usb/gadget/Kconfig | 18 + drivers/usb/gadget/function/Makefile | 2 + drivers/usb/gadget/function/f_midi2.c | 2883 +++++++++++++++++ drivers/usb/gadget/function/u_midi2.h | 81 + 6 files changed, 3192 insertions(+) create mode 100644 Documentation/ABI/testing/configfs-usb-gadget-midi2 create mode 100644 drivers/usb/gadget/function/f_midi2.c create mode 100644 drivers/usb/gadget/function/u_midi2.h -- 2.35.3

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[PATCH] ASoC: soc-core.c: use devm_snd_soc_register_dai()
by Kuninori Morimoto 01 Aug '23

01 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/soc-core.c | 42 ++++++++++-------------------------------- 1 file changed, 10 insertions(+), 32 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a5b96c17633a..7284363b9088 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2552,48 +2552,28 @@ struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, EXPORT_SYMBOL_GPL(snd_soc_register_dai); /** - * snd_soc_unregister_dais - Unregister DAIs from the ASoC core - * - * @component: The component for which the DAIs should be unregistered - */ -static void snd_soc_unregister_dais(struct snd_soc_component *component) -{ - struct snd_soc_dai *dai, *_dai; - - for_each_component_dais_safe(component, dai, _dai) - snd_soc_unregister_dai(dai); -} - -/** - * snd_soc_register_dais - Register a DAI with the ASoC core + * devm_snd_soc_register_dais - Register a DAI with the ASoC core * * @component: The component the DAIs are registered for * @dai_drv: DAI driver to use for the DAIs * @count: Number of DAIs */ -static int snd_soc_register_dais(struct snd_soc_component *component, - struct snd_soc_dai_driver *dai_drv, - size_t count) +static int devm_snd_soc_register_dais(struct snd_soc_component *component, + struct snd_soc_dai_driver *dai_drv, + size_t count) { struct snd_soc_dai *dai; unsigned int i; - int ret; for (i = 0; i < count; i++) { - dai = snd_soc_register_dai(component, dai_drv + i, count == 1 && - component->driver->legacy_dai_naming); - if (dai == NULL) { - ret = -ENOMEM; - goto err; - } + dai = devm_snd_soc_register_dai(component->dev, + component, dai_drv + i, count == 1 && + component->driver->legacy_dai_naming); + if (!dai) + return -ENOMEM; } return 0; - -err: - snd_soc_unregister_dais(component); - - return ret; } #define ENDIANNESS_MAP(name) \ @@ -2644,8 +2624,6 @@ static void snd_soc_del_component_unlocked(struct snd_soc_component *component) { struct snd_soc_card *card = component->card; - snd_soc_unregister_dais(component); - if (card) snd_soc_unbind_card(card, false); @@ -2696,7 +2674,7 @@ int snd_soc_add_component(struct snd_soc_component *component, } } - ret = snd_soc_register_dais(component, dai_drv, num_dai); + ret = devm_snd_soc_register_dais(component, dai_drv, num_dai); if (ret < 0) { dev_err(component->dev, "ASoC: Failed to register DAIs: %d\n", ret); -- 2.25.1

1 0

[PATCH] ASoC: codecs/hda.c: use devm_snd_soc_register_dai()
by Kuninori Morimoto 01 Aug '23

01 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/codecs/hda.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/hda.c b/sound/soc/codecs/hda.c index d57b043d6bfe..551bd0cbfc44 100644 --- a/sound/soc/codecs/hda.c +++ b/sound/soc/codecs/hda.c @@ -99,7 +99,7 @@ static int hda_codec_register_dais(struct hda_codec *codec, struct snd_soc_compo list_for_each_entry(pcm, &codec->pcm_list_head, list) { struct snd_soc_dai *dai; - dai = snd_soc_register_dai(component, drvs, false); + dai = devm_snd_soc_register_dai(component->dev, component, drvs, false); if (!dai) { dev_err(component->dev, "register dai for %s failed\n", pcm->name); return -EINVAL; @@ -108,7 +108,6 @@ static int hda_codec_register_dais(struct hda_codec *codec, struct snd_soc_compo ret = snd_soc_dapm_new_dai_widgets(dapm, dai); if (ret < 0) { dev_err(component->dev, "create widgets failed: %d\n", ret); - snd_soc_unregister_dai(dai); return ret; } @@ -135,7 +134,6 @@ static void hda_codec_unregister_dais(struct hda_codec *codec, for_each_pcm_streams(stream) snd_soc_dapm_free_widget(snd_soc_dai_get_widget(dai, stream)); - snd_soc_unregister_dai(dai); break; } } -- 2.25.1

1 0

[PATCH 0/2] ASoC: wm8960: Clocking tweaks
by Mark Brown 01 Aug '23

01 Aug '23

A couple of minor clocking improvements that make the wm8960 driver work a bit better with generic cards. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (2): ASoC: wm8960: Read initial MCLK rate from clock API ASoC: wm8960: Make automatic the default clocking mode sound/soc/codecs/wm8960.c | 8 ++++++++ sound/soc/codecs/wm8960.h | 4 ++-- 2 files changed, 10 insertions(+), 2 deletions(-) --- base-commit: 5d0c230f1de8c7515b6567d9afba1f196fb4e2f4 change-id: 20230730-asoc-wm8960-clk-6c7415ee22b4 Best regards, -- Mark Brown <broonie(a)kernel.org>

2 5

params_free functions don't work
by GitHub issues - opened 01 Aug '23

01 Aug '23

alsa-project/alsa-lib issue #341 was opened from imyxh: The following code segfaults on the call to `snd_pcm_hw_params_free` on my machine: ```c #include <alsa/asoundlib.h> void main() { snd_pcm_hw_params_t *params; snd_pcm_hw_params_alloca(&params); snd_pcm_hw_params_free(params); } ``` The same goes for `s/hw/sw/g`. Am I doing something wrong? In more complex examples it seems to fail with munmap_chunk complaining about an invalid pointer, or free complaining about an invalid size. How are the `*_params_free` functions meant to be used? Issue URL : https://github.com/alsa-project/alsa-lib/issues/341 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: codecs: rx-macro: Enable RX path clock to match POST_PMD
by Krzysztof Kozlowski 31 Jul '23

31 Jul '23

The driver disables RX path clock in SND_SOC_DAPM_POST_PMD of rx_macro_enable_interp_clk() and rx_macro_enable_mix_path(). Make the code symmetrical by enabling the clock in SND_SOC_DAPM_PRE_PMU. This also matches downstream Qualcomm driver for Qualcomm SM8550 SoC. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-rx-macro.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 685ca95ef4a9..a758ef4f575a 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -2529,6 +2529,9 @@ static int rx_macro_enable_interp_clk(struct snd_soc_component *component, CDC_RX_PATH_PGA_MUTE_MASK, 0x1); snd_soc_component_write_field(component, dsm_reg, CDC_RX_RXn_DSM_CLK_EN_MASK, 0x1); + snd_soc_component_update_bits(component, main_reg, + CDC_RX_PATH_CLK_EN_MASK, + CDC_RX_PATH_CLK_ENABLE); snd_soc_component_update_bits(component, rx_cfg2_reg, CDC_RX_RXn_HPF_CUT_FREQ_MASK, 0x03); rx_macro_load_compander_coeff(component, rx, interp_idx, event); @@ -2595,6 +2598,9 @@ static int rx_macro_enable_mix_path(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: rx_macro_enable_interp_clk(component, event, w->shift); + snd_soc_component_update_bits(component, mix_reg, + CDC_RX_RXn_MIX_CLK_EN_MASK, + CDC_RX_RXn_MIX_CLK_EN_MASK); break; case SND_SOC_DAPM_POST_PMU: snd_soc_component_write(component, gain_reg, -- 2.34.1

2 1

[PATCH 0/5] ASoC: rsnd: add multi Component support
by Kuninori Morimoto 31 Jul '23

31 Jul '23

Hi Mark These are part of "ASoC: add multi Component support" patch-set. The patch-set supports below case as "2 Cards". +-- basic board --------+ |+--------+ | || CPU ch0| <--> CodecA | || ch1| <-+ | |+--------+ | | +-------------|---------+ +-- expansion board ----+ | | | | +-> CodecB| +-----------------------+ Renesas sound driver and its Doc part were held for observation. Rob mentioned that "definitions" vs "$defs". But I got error on "$defs", no error on "definitions". I believe this change is not mandatory. He also mentioned that "reg" is missing, but I also believe that "reg" is automatically handled somehow/somewhere (I'm not sure detail, but other reviewer indicated it before). He also mentioned that "ports" and "port", but added new "ports" needs special handling. Using "ports" vs "port" are different, not compatible on this driver. This means we need both on Doc. Thus, there is no update for these. Kuninori Morimoto (5): ASoC: rsnd: use DAI driver ID instead of DAI ID ASoC: rsnd: cleanup rsnd_dai_of_node() ASoC: rsnd: enable multi Component support for Audio Graph Card/Card2 ASoC: dt-bindings: renesas,rsnd.yaml: add common port-def ASoC: dt-bindings: renesas,rsnd.yaml: enable multi ports for multi Component support .../bindings/sound/renesas,rsnd.yaml | 60 +++---- sound/soc/sh/rcar/core.c | 146 ++++++++++++------ sound/soc/sh/rcar/rsnd.h | 4 + 3 files changed, 134 insertions(+), 76 deletions(-) -- 2.25.1

2 6

Re: [PATCH v3 5/6] ASoC: dt-bindings: mediatek,mt7986-wm8960: add mt7986-wm8960 document
by Krzysztof Kozlowski 31 Jul '23

31 Jul '23

On 31/07/2023 09:31, Maso Huang (黃加竹) wrote: > On Fri, 2023-07-28 at 14:49 +0200, Krzysztof Kozlowski wrote: >> >> External email : Please do not click links or open attachments until >> you have verified the sender or the content. >> On 28/07/2023 11:08, Maso Huang wrote: >>> Add document for mt7986 board with wm8960. >>> >>> Signed-off-by: Maso Huang <maso.huang(a)mediatek.com> >>> --- >>> .../sound/mediatek,mt7986-wm8960.yaml | 53 >> +++++++++++++++++++ >>> 1 file changed, 53 insertions(+) >>> create mode 100644 >> Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml >>> >>> diff --git >> a/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml >> b/Documentation/devicetree/bindings/sound/mediatek,mt7986-wm8960.yaml >>> new file mode 100644 >>> index 000000000000..76394f7e5502 >>> --- /dev/null >>> +++ b/Documentation/devicetree/bindings/sound/mediatek,mt7986- >> wm8960.yaml >>> @@ -0,0 +1,53 @@ >>> +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) >>> +%YAML 1.2 >>> +--- >>> +$id: >> http://devicetree.org/schemas/sound/mediatek,mt7986-wm8960.yaml# >>> +$schema: http://devicetree.org/meta-schemas/core.yaml# >>> + >>> +title: MediaTek MT7986 sound card with WM8960 codec >>> + >>> +maintainers: >>> + - Maso Huang <maso.huang(a)mediatek.com> >>> + >>> +properties: >>> + compatible: >>> + const: mediatek,mt7986-wm8960-machine >>> + >>> + mediatek,platform: >>> + $ref: /schemas/types.yaml#/definitions/phandle >>> + description: The phandle of MT7986 platform. >>> + >>> + audio-routing: >>> + $ref: /schemas/types.yaml#/definitions/non-unique-string-array >>> + description: >>> + A list of the connections between audio components. Each >> entry is a >>> + sink/source pair of strings. Valid names could be the input >> or output >>> + widgets of audio components, power supplies, MicBias of >> codec and the >>> + software switch. >>> + >>> + mediatek,audio-codec: >>> + $ref: /schemas/types.yaml#/definitions/phandle >>> + description: The phandle of wm8960 codec. >>> + >> >> How did you implement Rob's comment? Or did you just ignore it? >> >> Best regards, >> Krzysztof >> > > Hi Krzysztof, > > Sorry, I did not mean to ignore Rob's comment. > I waited for some suggestion in mail below, but it seems Rob was a > little busy. > > https://lore.kernel.org/lkml/8c6316e79e40406e4d46709f602dcb14a4c00562.camel… > > After gentle ping last week and receiving your advice, I thought that > means to send the v3 patch and might discuss dtbingding in v3 series. > > So sorry for misunderstanding it, I'll check the details with Rob in v3 > series then refine it in v4. The problem is that you did not reference in this patch any ongoing discussion and further questions, so comment looks like addressed, while it was not. Rob said: "in a common schema and reference them " You said: "common part yaml and reference to it" so I think you both agreed on the same. The advice would be to create common binding which is then referenced by other and your bindings. However if you start doing it, you will notice that it is impossible, because you have conflicting types for "audio-codec", so you cannot have one definition. This leads to the point - property is probably wrong and you need dai-link with sound-dai property, just like most cards are doing. Best regards, Krzysztof

1 1

[PATCH 0/3] ASoC: mt8188-mt6359: add SOF support
by Trevor Wu 31 Jul '23

31 Jul '23

This series introduces dynamic pinctrl and adds support for the SOF on the mt8188-mt6359 machine driver. Trevor Wu (3): ASoC: mediatek: mt8188-mt6359: support dynamic pinctrl ASoC: mediatek: common: revise SOF common code ASoC: mediatek: mt8188-mt6359: add SOF support .../soc/mediatek/common/mtk-dsp-sof-common.c | 106 ++++++-- .../soc/mediatek/common/mtk-dsp-sof-common.h | 8 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 238 +++++++++++++++++- 3 files changed, 325 insertions(+), 27 deletions(-) -- 2.18.0

3 12

[PATCH v4] dt-bindings: sound: gtm601: convert to YAML
by David Heidelberg 31 Jul '23

31 Jul '23

Convert GTM601 binding to the YAML format. Signed-off-by: David Heidelberg <david(a)ixit.cz> --- v4: - added missing required property v3: - incorporated feedback from Krzysztof v2: - add #sound-dai-cells - put kernel(a)puri.sm into maintainers .../devicetree/bindings/sound/gtm601.txt | 19 --------- .../bindings/sound/option,gtm601.yaml | 40 +++++++++++++++++++ 2 files changed, 40 insertions(+), 19 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/gtm601.txt create mode 100644 Documentation/devicetree/bindings/sound/option,gtm601.yaml diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt deleted file mode 100644 index efa32a486c4a..000000000000 --- a/Documentation/devicetree/bindings/sound/gtm601.txt +++ /dev/null @@ -1,19 +0,0 @@ -GTM601 UMTS modem audio interface CODEC - -This device has no configuration interface. The sample rate and channels are -based on the compatible string - "option,gtm601" = 8kHz mono - "broadmobi,bm818" = 48KHz stereo - -Required properties: - - - compatible : one of - "option,gtm601" - "broadmobi,bm818" - - -Example: - -codec: gtm601_codec { - compatible = "option,gtm601"; -}; diff --git a/Documentation/devicetree/bindings/sound/option,gtm601.yaml b/Documentation/devicetree/bindings/sound/option,gtm601.yaml new file mode 100644 index 000000000000..69c2ccc79dc5 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/option,gtm601.yaml @@ -0,0 +1,40 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/option,gtm601.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: GTM601 UMTS modem audio interface CODEC + +maintainers: + - kernel(a)puri.sm + +description: > + This device has no configuration interface. The sample rate and channels are + based on the compatible string + +properties: + compatible: + oneOf: + - items: # 48 kHz stereo + - const: broadmobi,bm818 + - const: option,gtm601 + - const: option,gtm601 # 8 kHz mono + + '#sound-dai-cells': + const: 0 + +required: + - compatible + +allOf: + - $ref: dai-common.yaml# + +additionalProperties: false + +examples: + - | + codec { + compatible = "option,gtm601"; + #sound-dai-cells = <0>; + }; -- 2.40.1

4 3

[PATCH v6 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 31 Jul '23

31 Jul '23

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. This series is mostly just a resend keeping pace with the kernel under it, except for a minor fixup in the spi stuff. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 313 +++ MAINTAINERS | 4 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 98 + drivers/mfd/cs42l43-sdw.c | 239 ++ drivers/mfd/cs42l43.c | 1189 +++++++++ drivers/mfd/cs42l43.h | 28 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 32 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 284 ++ include/linux/mfd/cs42l43-regs.h | 1184 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 969 +++++++ sound/soc/codecs/cs42l43-sdw.c | 74 + sound/soc/codecs/cs42l43.c | 2278 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 131 + 26 files changed, 7639 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

5 10

Re: [PATCH v3 5/6] ASoC: dt-bindings: mediatek,mt7986-wm8960: add mt7986-wm8960 document
by AngeloGioacchino Del Regno 31 Jul '23

31 Jul '23

Il 31/07/23 08:42, Maso Huang (黃加竹) ha scritto: > On Fri, 2023-07-28 at 18:16 +0800, Maso Huang wrote: >> On Fri, 2023-07-28 at 11:55 +0200, AngeloGioacchino Del Regno wrote: >>> Il 28/07/23 11:08, Maso Huang ha scritto: >>>> Add document for mt7986 board with wm8960. >>>> >>>> Signed-off-by: Maso Huang <maso.huang(a)mediatek.com> >>>> --- >>>> .../sound/mediatek,mt7986-wm8960.yaml | 53 >>>> +++++++++++++++++++ >>>> 1 file changed, 53 insertions(+) >>>> create mode 100644 >>>> Documentation/devicetree/bindings/sound/mediatek,mt7986- >>>> wm8960.yaml >>>> >>>> diff --git >>>> a/Documentation/devicetree/bindings/sound/mediatek,mt7986- >>>> wm8960.yaml >>>> b/Documentation/devicetree/bindings/sound/mediatek,mt7986- >>>> wm8960.yaml >>>> new file mode 100644 >>>> index 000000000000..76394f7e5502 >>>> --- /dev/null >>>> +++ b/Documentation/devicetree/bindings/sound/mediatek,mt7986- >>>> wm8960.yaml >>>> @@ -0,0 +1,53 @@ >>>> +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) >>>> +%YAML 1.2 >>>> +--- >>>> +$id: >>>> > https://urldefense.com/v3/__http://devicetree.org/schemas/sound/mediatek,mt… >>>> >>>> >>>> +$schema: >>>> > https://urldefense.com/v3/__http://devicetree.org/meta-schemas/core.yaml*__… >>>> >>>> >>>> + >>>> +title: MediaTek MT7986 sound card with WM8960 codec >>>> + >>>> +maintainers: >>>> + - Maso Huang <maso.huang(a)mediatek.com> >>>> + >>>> +properties: >>>> + compatible: >>>> + const: mediatek,mt7986-wm8960-machine >>> >>> mediatek,mt7986-wm8960-sound looks better. >>> >>> After which, >>> >>> Reviewed-by: AngeloGioacchino Del Regno < >>> angelogioacchino.delregno(a)collabora.com> >>> >>> >> >> Hi Angelo, >> >> Thanks for your review. >> I'll change the compatible to "mediatek,mt7986-wm8960-sound" in v4 >> patch. >> >> Best regards, >> Maso >> > > Hi Angelo, > > One more question for this compatible. > > The suffix "machine" means alsa machine driver for mt7986-wm8960. It > might be better to use "machine" here. > > Or you prefer "sound" than "machine"? > > Best regards, > Maso I prefer "sound" because of consistency with other MediaTek machine driver compatible strings. Regards, Angelo

1 0

[PATCH V3 0/5] ASoC: codecs: Add awinic AW88261 audio amplifier driver
by wangweidong.a＠awinic.com 31 Jul '23

31 Jul '23

From: Weidong Wang <wangweidong.a(a)awinic.com> The awinic AW88261 is an I2S/TDM input, high efficiency digital Smart K audio amplifier v1 -> v2: Submit the yaml file as a separate patch file Modify word capitalization in the Kconfig file Delete unused macro definitions Weidong Wang (5): ASoC: dt-bindings: Add schema for "awinic,aw88261" ASoC: codecs: Add code for bin parsing compatible with aw88261 ASoC: codecs: Add aw88261 amplifier driver ASoC: codecs: aw88261 device related operation functions ASoC: codecs: aw88261 chip register file, Kconfig and Makefile .../bindings/sound/awinic,aw88395.yaml | 4 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/aw88261/aw88261.c | 517 +++++++++++ sound/soc/codecs/aw88261/aw88261.h | 52 ++ sound/soc/codecs/aw88261/aw88261_device.c | 877 ++++++++++++++++++ sound/soc/codecs/aw88261/aw88261_device.h | 79 ++ sound/soc/codecs/aw88261/aw88261_reg.h | 374 ++++++++ sound/soc/codecs/aw88395/aw88395_lib.c | 193 +++- sound/soc/codecs/aw88395/aw88395_reg.h | 1 + 10 files changed, 2097 insertions(+), 18 deletions(-) create mode 100644 sound/soc/codecs/aw88261/aw88261.c create mode 100644 sound/soc/codecs/aw88261/aw88261.h create mode 100644 sound/soc/codecs/aw88261/aw88261_device.c create mode 100644 sound/soc/codecs/aw88261/aw88261_device.h create mode 100644 sound/soc/codecs/aw88261/aw88261_reg.h base-commit: ffabf7c731765da3dbfaffa4ed58b51ae9c2e650 -- 2.41.0

2 14

[PATCH v3] um/drivers: fix hostaudio build errors
by Randy Dunlap 31 Jul '23

31 Jul '23

Use "select"s to ensure that the required kconfig symbols are set as expected. Drop HOSTAUDIO since it is now equivalent to UML_SOUND. Allow SOUND with UML regardless of HAS_IOMEM. Otherwise there is a kconfig warning for unmet dependencies. (This was not an issue when SOUND was defined in arch/um/drivers/Kconfig. I have done 50 randconfig builds and didn't find any issues.) This fixes build errors when CONFIG_SOUND is not set: ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_cleanup_module': hostaudio_kern.c:(.exit.text+0xa): undefined reference to `unregister_sound_mixer' ld: hostaudio_kern.c:(.exit.text+0x15): undefined reference to `unregister_sound_dsp' ld: arch/um/drivers/hostaudio_kern.o: in function `hostaudio_init_module': hostaudio_kern.c:(.init.text+0x19): undefined reference to `register_sound_dsp' ld: hostaudio_kern.c:(.init.text+0x31): undefined reference to `register_sound_mixer' ld: hostaudio_kern.c:(.init.text+0x49): undefined reference to `unregister_sound_dsp' and this kconfig warning: WARNING: unmet direct dependencies detected for SOUND Fixes: 1da177e4c3f4 ("Linux-2.6.12-rc2") Fixes: d886e87cb82b ("sound: make OSS sound core optional") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Link: lore.kernel.org/r/202307141416.vxuRVpFv-lkp@intel.com Cc: Richard Weinberger <richard(a)nod.at> Cc: Anton Ivanov <anton.ivanov(a)cambridgegreys.com> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: linux-um(a)lists.infradead.org Cc: Tejun Heo <tj(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Nick Desaulniers <ndesaulniers(a)google.com> Cc: Nicolas Schier <nicolas(a)fjasle.eu> Cc: linux-kbuild(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- v2: don't delete the HOSTAUDIO Kconfig entry (Masahiro) v3: drop HOSTAUDIO and use CONFIG_UML_SOUND for it in Makefile (Takashi); add SOUND depends on "|| UML" to HAS_IOMEM arch/um/drivers/Kconfig | 14 ++------------ arch/um/drivers/Makefile | 2 +- sound/Kconfig | 2 +- 3 files changed, 4 insertions(+), 14 deletions(-) diff -- a/arch/um/drivers/Kconfig b/arch/um/drivers/Kconfig --- a/arch/um/drivers/Kconfig +++ b/arch/um/drivers/Kconfig @@ -111,24 +111,14 @@ config SSL_CHAN config UML_SOUND tristate "Sound support" + select SOUND + select SOUND_OSS_CORE help This option enables UML sound support. If enabled, it will pull in soundcore and the UML hostaudio relay, which acts as a intermediary between the host's dsp and mixer devices and the UML sound system. It is safe to say 'Y' here. -config SOUND - tristate - default UML_SOUND - -config SOUND_OSS_CORE - bool - default UML_SOUND - -config HOSTAUDIO - tristate - default UML_SOUND - endmenu menu "UML Network Devices" diff -- a/sound/Kconfig b/sound/Kconfig --- a/sound/Kconfig +++ b/sound/Kconfig @@ -1,7 +1,7 @@ # SPDX-License-Identifier: GPL-2.0-only menuconfig SOUND tristate "Sound card support" - depends on HAS_IOMEM + depends on HAS_IOMEM || UML help If you have a sound card in your computer, i.e. if it can say more than an occasional beep, say Y. diff -- a/arch/um/drivers/Makefile b/arch/um/drivers/Makefile --- a/arch/um/drivers/Makefile +++ b/arch/um/drivers/Makefile @@ -54,7 +54,7 @@ obj-$(CONFIG_UML_NET) += net.o obj-$(CONFIG_MCONSOLE) += mconsole.o obj-$(CONFIG_MMAPPER) += mmapper_kern.o obj-$(CONFIG_BLK_DEV_UBD) += ubd.o -obj-$(CONFIG_HOSTAUDIO) += hostaudio.o +obj-$(CONFIG_UML_SOUND) += hostaudio.o obj-$(CONFIG_NULL_CHAN) += null.o obj-$(CONFIG_PORT_CHAN) += port.o obj-$(CONFIG_PTY_CHAN) += pty.o

4 4

[PATCH] ASoC: dt-bindings: Convert maxim,max98925 to DT schema
by Krzysztof Kozlowski 30 Jul '23

30 Jul '23

Convert the Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier bindings to DT schema format. Changes during conversion: add sound-dai-cells, already used by DTS. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Ryan, As original author, I added you as bindings maintainer. Is that okay or maybe someone else from Maxim should take care about the bindings? --- .../devicetree/bindings/sound/max9892x.txt | 44 -------- .../bindings/sound/maxim,max98925.yaml | 100 ++++++++++++++++++ 2 files changed, 100 insertions(+), 44 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/max9892x.txt create mode 100644 Documentation/devicetree/bindings/sound/maxim,max98925.yaml diff --git a/Documentation/devicetree/bindings/sound/max9892x.txt b/Documentation/devicetree/bindings/sound/max9892x.txt deleted file mode 100644 index 98cb9ba5b328..000000000000 --- a/Documentation/devicetree/bindings/sound/max9892x.txt +++ /dev/null @@ -1,44 +0,0 @@ -Maxim Integrated MAX98925/MAX98926/MAX98927 Speaker Amplifier - -This device supports I2C. - -Required properties: - - - compatible : should be one of the following - - "maxim,max98925" - - "maxim,max98926" - - "maxim,max98927" - - - vmon-slot-no : slot number used to send voltage information - or in inteleave mode this will be used as - interleave slot. - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - imon-slot-no : slot number used to send current information - MAX98925/MAX98926 slot range : 0 ~ 30, Default : 0 - MAX98927 slot range : 0 ~ 15, Default : 0 - - - interleave-mode : When using two MAX9892X in a system it is - possible to create ADC data that that will - overflow the frame size. Digital Audio Interleave - mode provides a means to output VMON and IMON data - from two devices on a single DOUT line when running - smaller frames sizes such as 32 BCLKS per LRCLK or - 48 BCLKS per LRCLK. - Range : 0 (off), 1 (on), Default : 0 - - - reg : the I2C address of the device for I2C - -Optional properties: - - reset-gpios : GPIO to reset the device - -Example: - -codec: max98927@3a { - compatible = "maxim,max98927"; - vmon-slot-no = <0>; - imon-slot-no = <1>; - interleave-mode = <0>; - reg = <0x3a>; -}; diff --git a/Documentation/devicetree/bindings/sound/maxim,max98925.yaml b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml new file mode 100644 index 000000000000..a01a130ef686 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/maxim,max98925.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/maxim,max98925.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Maxim Integrated MAX98925/MAX98926/MAX98927 speaker amplifier + +maintainers: + - Ryan Lee <ryans.lee(a)maximintegrated.com> + +properties: + compatible: + enum: + - maxim,max98925 + - maxim,max98926 + - maxim,max98927 + + reg: + maxItems: 1 + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vmon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send voltage information or in inteleave mode this + will be used as interleave slot. + + imon-slot-no: + $ref: /schemas/types.yaml#/definitions/uint32 + minimum: 0 + maximum: 30 + default: 0 + description: + Slot number used to send current information. + + interleave-mode: + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [0, 1] + default: 0 + description: + When using two MAX9892X in a system it is possible to create ADC data + that will overflow the frame size. Digital Audio Interleave mode provides + a means to output VMON and IMON data from two devices on a single DOUT + line when running smaller frames sizes such as 32 BCLKS per LRCLK or 48 + BCLKS per LRCLK. + +required: + - compatible + - reg + +allOf: + - $ref: dai-common.yaml# + - if: + properties: + compatible: + contains: + enum: + - maxim,max98927 + then: + properties: + vmon-slot-no: + minimum: 0 + maximum: 15 + + imon-slot-no: + minimum: 0 + maximum: 15 + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + #include <dt-bindings/gpio/gpio.h> + audio-codec@3a { + compatible = "maxim,max98927"; + reg = <0x3a>; + #sound-dai-cells = <0>; + + pinctrl-0 = <&speaker_default>; + pinctrl-names = "default"; + + reset-gpios = <&tlmm 69 GPIO_ACTIVE_LOW>; + + vmon-slot-no = <1>; + imon-slot-no = <0>; + }; + }; -- 2.34.1

1 1

[PATCH v3] dt-bindings: sound: gtm601: convert to YAML
by David Heidelberg 30 Jul '23

30 Jul '23

Convert GTM601 binding to the YAML format. Signed-off-by: David Heidelberg <david(a)ixit.cz> --- v3: - incorporated feedback from Krzysztof v2: - add #sound-dai-cells - put kernel(a)puri.sm into maintainers .../devicetree/bindings/sound/gtm601.txt | 19 ---------- .../bindings/sound/option,gtm601.yaml | 37 +++++++++++++++++++ 2 files changed, 37 insertions(+), 19 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/gtm601.txt create mode 100644 Documentation/devicetree/bindings/sound/option,gtm601.yaml diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt deleted file mode 100644 index efa32a486c4a..000000000000 --- a/Documentation/devicetree/bindings/sound/gtm601.txt +++ /dev/null @@ -1,19 +0,0 @@ -GTM601 UMTS modem audio interface CODEC - -This device has no configuration interface. The sample rate and channels are -based on the compatible string - "option,gtm601" = 8kHz mono - "broadmobi,bm818" = 48KHz stereo - -Required properties: - - - compatible : one of - "option,gtm601" - "broadmobi,bm818" - - -Example: - -codec: gtm601_codec { - compatible = "option,gtm601"; -}; diff --git a/Documentation/devicetree/bindings/sound/option,gtm601.yaml b/Documentation/devicetree/bindings/sound/option,gtm601.yaml new file mode 100644 index 000000000000..9a70d20eb30d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/option,gtm601.yaml @@ -0,0 +1,37 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/option,gtm601.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: GTM601 UMTS modem audio interface CODEC + +maintainers: + - kernel(a)puri.sm + +description: > + This device has no configuration interface. The sample rate and channels are + based on the compatible string + +properties: + compatible: + oneOf: + - items: # 48 kHz stereo + - const: broadmobi,bm818 + - const: option,gtm601 + - const: option,gtm601 # 8 kHz mono + + '#sound-dai-cells': + const: 0 + +allOf: + - $ref: dai-common.yaml# + +additionalProperties: false + +examples: + - | + codec { + compatible = "option,gtm601"; + #sound-dai-cells = <0>; + }; -- 2.40.1

2 1

[PATCH RESEND v2] dt-bindings: sound: gtm601: convert to YAML
by David Heidelberg 30 Jul '23

30 Jul '23

Convert GTM601 binding to the YAML format. Signed-off-by: David Heidelberg <david(a)ixit.cz> Cc: kernel(a)puri.sm --- v2: - add #sound-dai-cells - put kernel(a)puri.sm into maintainers .../devicetree/bindings/sound/gtm601.txt | 19 ---------- .../devicetree/bindings/sound/gtm601.yaml | 36 +++++++++++++++++++ 2 files changed, 36 insertions(+), 19 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/gtm601.txt create mode 100644 Documentation/devicetree/bindings/sound/gtm601.yaml diff --git a/Documentation/devicetree/bindings/sound/gtm601.txt b/Documentation/devicetree/bindings/sound/gtm601.txt deleted file mode 100644 index efa32a486c4a..000000000000 --- a/Documentation/devicetree/bindings/sound/gtm601.txt +++ /dev/null @@ -1,19 +0,0 @@ -GTM601 UMTS modem audio interface CODEC - -This device has no configuration interface. The sample rate and channels are -based on the compatible string - "option,gtm601" = 8kHz mono - "broadmobi,bm818" = 48KHz stereo - -Required properties: - - - compatible : one of - "option,gtm601" - "broadmobi,bm818" - - -Example: - -codec: gtm601_codec { - compatible = "option,gtm601"; -}; diff --git a/Documentation/devicetree/bindings/sound/gtm601.yaml b/Documentation/devicetree/bindings/sound/gtm601.yaml new file mode 100644 index 000000000000..e81a6aa75522 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/gtm601.yaml @@ -0,0 +1,36 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/gtm601.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: GTM601 UMTS modem audio interface CODEC + +maintainers: + - kernel(a)puri.sm + +description: > + This device has no configuration interface. The sample rate and channels are + based on the compatible string + +properties: + compatible: + oneOf: + - items: + - const: broadmobi,bm818 + - const: option,gtm601 + - items: + - enum: + - broadmobi,bm818 # 48 kHz stereo + - option,gtm601 # 8 kHz mono + + '#sound-dai-cells': + const: 0 + +additionalProperties: false + +examples: + - | + gtm601_codec { + compatible = "option,gtm601"; + }; -- 2.40.1

2 2

Problem with Rog Delta S Wireless
by GitHub issues - opened 30 Jul '23

30 Jul '23

alsa-project/alsa-utils issue #227 was opened from bagou4502: Greetings, I am posting this issue here since I haven't found a solution in forums yet. I am facing an problem with my Rog Delta S Wireless headset when using Linux with Gnome. The volume control doesn't function correctly. Setting the volume to < 95% keeps the sound at 0 in alsamixer. When I set it to 95%, alsamixer shows 50%, and 100% corresponds to 100% in alsamixer. This issue occurs specifically with this headset in both 2.4GHz and Bluetooth modes. I have already tried using another Rog Delta S Wireless headset, but the same problem persists. Additionally, I tested it on multiple computers running Fedora, Ubuntu 22.04 and 23.04, Debian 11 and 12, as well as ZorinOs 16.2 and 16.3, and encountered the same problem on all of them. Here is a video demonstrating the issue: https://www.youtube.com/watch?v=-OuKtLt5U1k Unfortunately, I am uncertain whether this problem originates from the Gnome volume system or AlsaMixer. I am hopeful that someone can assist me with this matter. Thank you for your help. Issue URL : https://github.com/alsa-project/alsa-utils/issues/227 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Documentation cleanup
by GitHub pull_request - opened 30 Jul '23

30 Jul '23

alsa-project/alsa-lib pull request #340 was opened from borine: Fixes missing doxygen tags and macros so that now doxygen picks up all of the documentation comments and completes cleanly without any errors or warnings. Request URL : https://github.com/alsa-project/alsa-lib/pull/340 Patch URL : https://github.com/alsa-project/alsa-lib/pull/340.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fwd: Lenovo Thinkpad T490 Microphone Noise
by Bagas Sanjaya 30 Jul '23

30 Jul '23

Hi, I notice a bug report on Bugzilla [1]. Quoting from it: > I have reported this issue to the Arch but they redirected me to here. > https://bugs.archlinux.org/task/79245 > > > Description: > When I use microphone for online meetings, my sound reaches them with a high noise that prevent it from being understood. I tested the microphone on my own and yes the noise is there. I tried pulseaudio and pipewire, but neither has solved the problem. This problem is not stemming from hardware since I do not have this issue with Windows 10. > > Additional info: > * pulseaudio: 16.1-6, piprewire: 1:0.3.75-2, kernel: 6.4.6-arch1-1 > > Steps to reproduce: > 1 - Open a chat app/website or online meeting tool like, Whatsapp, Signal, Zoom, etc. > 2 - Speak to the microphone. You can use Whatsapp and Signal to listen your microphone sound. See Bugzilla for the full thread. The reporter chose the incorrect component when this bug was being filed on Bugzilla (only generic `Kernel`), hence missing sound maintainers and I had to forward it to the mailing list instead. Thanks. [1]: https://bugzilla.kernel.org/show_bug.cgi?id=217730 -- An old man doll... just what I always wanted! - Clara

1 0

ALC3306 beeing registered as ALC287 on Lenovo Yoga
by GitHub issues - opened 29 Jul '23

29 Jul '23

alsa-project/alsa-lib issue #339 was opened from InfinityRed-Code: I have a Lenovo Yoga 7 16IAP7 with Fedora Workstation 38 installed. Kernel Verion: Linux 6.4.6-200.fc38.x86_64 alsa-lib version: alsa-lib-1.2.9-1.fc38.x86_64 I installed alsa-tools and went into hdajackretask to see what is going on. Then I noticed that my System is telling me that I have an ALC287 installed but actually it is a ALC3306. Sound is horrible and several fixes with ALC 287 for Legion Models did nit work. In the Settings I can see as well, that only 2 of the 4 integrated speakers are getting noticed. Is there any known fix? Issue URL : https://github.com/alsa-project/alsa-lib/issues/339 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ALSA: bcd2000: refactor deprecated strncpy
by Justin Stitt 29 Jul '23

29 Jul '23

`strncpy` is deprecated for use on NUL-terminated destination strings [1]. A suitable replacement is `strscpy` [2] due to the fact that it guarantees NUL-termination on its destination buffer argument which is _not_ always the case for `strncpy`! It should be noted that, in this case, the destination buffer has a length strictly greater than the source string. Moreover, the source string is NUL-terminated (and so is the destination) which means there was no real bug happening here. Nonetheless, this patch would get us one step closer to eliminating the `strncpy` API in the kernel, as its use is too ambiguous. We need to favor less ambiguous replacements such as: strscpy, strscpy_pad, strtomem and strtomem_pad (amongst others). Technically, my patch yields subtly different behavior. The original implementation with `strncpy` would fill the entire destination buffer with null bytes [3] while `strscpy` will leave the junk, uninitialized bytes trailing after the _mandatory_ NUL-termination. So, if somehow `card->driver` or `card->shortname` require this NUL-padding behavior then `strscpy_pad` should be used. My interpretation, though, is that the aforementioned fields are just fine as NUL-terminated strings. Please correct my assumptions if needed and I'll send in a v2. [1]: www.kernel.org/doc/html/latest/process/deprecated.html#strncpy-on-nul-termi… [2]: manpages.debian.org/testing/linux-manual-4.8/strscpy.9.en.html [3]: https://linux.die.net/man/3/strncpy Link: https://github.com/KSPP/linux/issues/90 Link: https://lore.kernel.org/r/20230727-sound-xen-v1-1-89dd161351f1@google.com (related ALSA patch) Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- sound/usb/bcd2000/bcd2000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c index 7aec0a95c609..392b4d8e9e76 100644 --- a/sound/usb/bcd2000/bcd2000.c +++ b/sound/usb/bcd2000/bcd2000.c @@ -395,8 +395,8 @@ static int bcd2000_probe(struct usb_interface *interface, snd_card_set_dev(card, &interface->dev); - strncpy(card->driver, "snd-bcd2000", sizeof(card->driver)); - strncpy(card->shortname, "BCD2000", sizeof(card->shortname)); + strscpy(card->driver, "snd-bcd2000", sizeof(card->driver)); + strscpy(card->shortname, "BCD2000", sizeof(card->shortname)); usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), "Behringer BCD2000 at %s", --- base-commit: 57012c57536f8814dec92e74197ee96c3498d24e change-id: 20230727-sound-usb-bcd2000-400b3060a9f9 Best regards, -- Justin Stitt <justinstitt(a)google.com>

2 1

[PATCH v3 31/50] ASoC: dt-bindings: microchip: add sam9x7
by Varshini Rajendran 29 Jul '23

29 Jul '23

Add sam9x7 compatible in the DT documentation. Signed-off-by: Varshini Rajendran <varshini.rajendran(a)microchip.com> --- .../bindings/sound/microchip,sama7g5-i2smcc.yaml | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml index 651f61c7c25a..fb630a184350 100644 --- a/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-i2smcc.yaml @@ -24,9 +24,14 @@ properties: const: 0 compatible: - enum: - - microchip,sam9x60-i2smcc - - microchip,sama7g5-i2smcc + oneOf: + - enum: + - microchip,sam9x60-i2smcc + - microchip,sama7g5-i2smcc + - items: + - enum: + - microchip,sam9x7-i2smcc + - const: microchip,sam9x60-i2smcc reg: maxItems: 1 -- 2.25.1

2 1

[tiwai-sound:test/pcm-sockptr 18/26] drivers/media/pci/solo6x10/solo6x10-g723.c:242:11: error: use of undeclared identifier 'snd_solo_pcm_copy_user'; did you mean 'snd_solo_pcm_copy'?
by kernel test robot 29 Jul '23

29 Jul '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/pcm-sockptr head: 3fffa74b00add561981357c76375e1829059d4e5 commit: 79cf0bf97368998dd44c1863d6f54057846b68c9 [18/26] media: solo6x10: Convert to generic PCM copy ops config: x86_64-randconfig-x004-20230728 (https://download.01.org/0day-ci/archive/20230729/202307291526.AOnKZKB1-lkp@…) compiler: clang version 16.0.4 (https://github.com/llvm/llvm-project.git ae42196bc493ffe877a7e3dff8be32035dea4d07) reproduce: (https://download.01.org/0day-ci/archive/20230729/202307291526.AOnKZKB1-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307291526.AOnKZKB1-lkp@intel.com/ All errors (new ones prefixed by >>): >> drivers/media/pci/solo6x10/solo6x10-g723.c:242:11: error: use of undeclared identifier 'snd_solo_pcm_copy_user'; did you mean 'snd_solo_pcm_copy'? .copye = snd_solo_pcm_copy_user, ^~~~~~~~~~~~~~~~~~~~~~ snd_solo_pcm_copy drivers/media/pci/solo6x10/solo6x10-g723.c:207:12: note: 'snd_solo_pcm_copy' declared here static int snd_solo_pcm_copy(struct snd_pcm_substream *ss, int channel, ^ >> drivers/media/pci/solo6x10/solo6x10-g723.c:242:3: error: field designator 'copye' does not refer to any field in type 'const struct snd_pcm_ops'; did you mean 'copy'? .copye = snd_solo_pcm_copy_user, ^~~~~ copy include/sound/pcm.h:72:8: note: 'copy' declared here int (*copy)(struct snd_pcm_substream *substream, int channel, ^ 2 errors generated. vim +242 drivers/media/pci/solo6x10/solo6x10-g723.c 235 236 static const struct snd_pcm_ops snd_solo_pcm_ops = { 237 .open = snd_solo_pcm_open, 238 .close = snd_solo_pcm_close, 239 .prepare = snd_solo_pcm_prepare, 240 .trigger = snd_solo_pcm_trigger, 241 .pointer = snd_solo_pcm_pointer, > 242 .copye = snd_solo_pcm_copy_user, 243 }; 244 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:test/pcm-sockptr 18/26] drivers/media/pci/solo6x10/solo6x10-g723.c:207:12: warning: 'snd_solo_pcm_copy' defined but not used
by kernel test robot 29 Jul '23

29 Jul '23

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git test/pcm-sockptr head: 3fffa74b00add561981357c76375e1829059d4e5 commit: 79cf0bf97368998dd44c1863d6f54057846b68c9 [18/26] media: solo6x10: Convert to generic PCM copy ops config: i386-buildonly-randconfig-r004-20230728 (https://download.01.org/0day-ci/archive/20230729/202307291119.03yget13-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce: (https://download.01.org/0day-ci/archive/20230729/202307291119.03yget13-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202307291119.03yget13-lkp@intel.com/ All warnings (new ones prefixed by >>): drivers/media/pci/solo6x10/solo6x10-g723.c:242:10: error: 'const struct snd_pcm_ops' has no member named 'copye'; did you mean 'copy'? 242 | .copye = snd_solo_pcm_copy_user, | ^~~~~ | copy drivers/media/pci/solo6x10/solo6x10-g723.c:242:18: error: 'snd_solo_pcm_copy_user' undeclared here (not in a function); did you mean 'snd_solo_pcm_copy'? 242 | .copye = snd_solo_pcm_copy_user, | ^~~~~~~~~~~~~~~~~~~~~~ | snd_solo_pcm_copy drivers/media/pci/solo6x10/solo6x10-g723.c:242:18: error: positional initialization of field in 'struct' declared with 'designated_init' attribute [-Werror=designated-init] drivers/media/pci/solo6x10/solo6x10-g723.c:242:18: note: (near initialization for 'snd_solo_pcm_ops') >> drivers/media/pci/solo6x10/solo6x10-g723.c:207:12: warning: 'snd_solo_pcm_copy' defined but not used [-Wunused-function] 207 | static int snd_solo_pcm_copy(struct snd_pcm_substream *ss, int channel, | ^~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/snd_solo_pcm_copy +207 drivers/media/pci/solo6x10/solo6x10-g723.c 206 > 207 static int snd_solo_pcm_copy(struct snd_pcm_substream *ss, int channel, 208 unsigned long pos, sockptr_t dst, 209 unsigned long count) 210 { 211 struct solo_snd_pcm *solo_pcm = snd_pcm_substream_chip(ss); 212 struct solo_dev *solo_dev = solo_pcm->solo_dev; 213 unsigned int off = 0; 214 int err, i; 215 216 for (i = 0; i < (count / G723_FRAMES_PER_PAGE); i++) { 217 int page = (pos / G723_FRAMES_PER_PAGE) + i; 218 219 err = solo_p2m_dma_t(solo_dev, 0, solo_pcm->g723_dma, 220 SOLO_G723_EXT_ADDR(solo_dev) + 221 (page * G723_PERIOD_BLOCK) + 222 (ss->number * G723_PERIOD_BYTES), 223 G723_PERIOD_BYTES, 0, 0); 224 if (err) 225 return err; 226 227 if (copy_to_sockptr_offset(dst, off, 228 solo_pcm->g723_buf, G723_PERIOD_BYTES)) 229 return -EFAULT; 230 off += G723_PERIOD_BYTES; 231 } 232 233 return 0; 234 } 235 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH] ASoC: 88pm860x: refactor deprecated strncpy
by Justin Stitt 29 Jul '23

29 Jul '23

`strncpy` is deprecated for use on NUL-terminated destination strings [1]. A suitable replacement is `strscpy` [2] due to the fact that it guarantees NUL-termination on its destination buffer argument which is _not_ always the case for `strncpy`! In this case, though, there was care taken to ensure that the destination buffer would be NUL-terminated. The destination buffer is zero-initialized and each `pm860x->name[i]` has a size of `MAX_NAME_LENGTH + 1`. This means that there is unlikely to be a bug here. However, in an attempt to eliminate the usage of the `strncpy` API as well as disambiguate implementations, replacements such as: `strscpy`, `strscpy_pad`, `strtomem` and `strtomem_pad` should be preferred. We are able to eliminate the need for `len + 1` since `strscpy` guarantees NUL-termination for its destination buffer as per its implementation [3]: | /* Hit buffer length without finding a NUL; force NUL-termination. */ | if (res) | dest[res-1] = '\0'; [1]: www.kernel.org/doc/html/latest/process/deprecated.html#strncpy-on-nul-termi… [2]: manpages.debian.org/testing/linux-manual-4.8/strscpy.9.en.html [3]: https://elixir.bootlin.com/linux/v6.3/source/lib/string.c#L183 Link: https://github.com/KSPP/linux/issues/90 Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- sound/soc/codecs/88pm860x-codec.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 3574c68e0dda..d99b674d574b 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -143,7 +143,7 @@ struct pm860x_priv { struct pm860x_det det; int irq[4]; - unsigned char name[4][MAX_NAME_LEN+1]; + unsigned char name[4][MAX_NAME_LEN]; }; /* -9450dB to 0dB in 150dB steps ( mute instead of -9450dB) */ @@ -1373,7 +1373,7 @@ static int pm860x_codec_probe(struct platform_device *pdev) return -EINVAL; } pm860x->irq[i] = res->start + chip->irq_base; - strncpy(pm860x->name[i], res->name, MAX_NAME_LEN); + strscpy(pm860x->name[i], res->name, MAX_NAME_LEN); } ret = devm_snd_soc_register_component(&pdev->dev, --- base-commit: 57012c57536f8814dec92e74197ee96c3498d24e change-id: 20230727-sound-soc-codecs-947fcb9536a7 Best regards, -- Justin Stitt <justinstitt(a)google.com>

3 2

[PATCH] ASoC: Intel: Skylake: replace deprecated strncpy with strscpy
by justinstitt＠google.com 28 Jul '23

28 Jul '23

`strncpy` is deprecated for use on NUL-terminated destination strings [1]. A suitable replacement is `strscpy` [2] due to the fact that it guarantees NUL-termination on its destination buffer argument which is _not_ the case for `strncpy`! It was pretty difficult, in this case, to try and figure out whether or not the destination buffer was zero-initialized. If it is and this behavior is relied on then perhaps `strscpy_pad` is the preferred option here. Kees was able to help me out and identify the following code snippet which seems to show that the destination buffer is zero-initialized. | skl = devm_kzalloc(&pci->dev, sizeof(*skl), GFP_KERNEL); With this information, I opted for `strscpy` since padding is seemingly not required. Also within this patch is a change to an instance of `x > y - 1` to `x >= y` which tends to be more robust and readable. Consider, for instance, if `y` was somehow `INT_MIN`. [1]: www.kernel.org/doc/html/latest/process/deprecated.html#strncpy-on-nul-termi… [2]: manpages.debian.org/testing/linux-manual-4.8/strscpy.9.en.html Link: https://github.com/KSPP/linux/issues/90 Suggested-by: Kees Cook <keescook(a)chromium.org> Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- sound/soc/intel/skylake/skl-topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 96cfebded072..67f08ec3a2ea 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3154,12 +3154,12 @@ static int skl_tplg_fill_str_mfest_tkn(struct device *dev, switch (str_elem->token) { case SKL_TKN_STR_LIB_NAME: - if (ref_count > skl->lib_count - 1) { + if (ref_count >= skl->lib_count) { ref_count = 0; return -EINVAL; } - strncpy(skl->lib_info[ref_count].name, + strscpy(skl->lib_info[ref_count].name, str_elem->string, ARRAY_SIZE(skl->lib_info[ref_count].name)); ref_count++; --- base-commit: 0b4a9fdc9317440a71d4d4c264a5650bf4a90f3c change-id: 20230726-asoc-intel-skylake-remove-deprecated-strncpy-9dbcfc26040c Best regards, -- Justin Stitt <justinstitt(a)google.com>

5 9

[PATCH] ALSA: xen-front: refactor deprecated strncpy
by Justin Stitt 28 Jul '23

28 Jul '23

`strncpy` is deprecated for use on NUL-terminated destination strings [1]. A suitable replacement is `strscpy` [2] due to the fact that it guarantees NUL-termination on its destination buffer argument which is _not_ always the case for `strncpy`! It should be noted that, in this case, the destination buffer has a length strictly greater than the source string. Moreover, the source string is NUL-terminated (and so is the destination) which means there was no real bug happening here. Nonetheless, this patch would get us one step closer to eliminating the `strncpy` API in the kernel, as its use is too ambiguous. We need to favor less ambiguous replacements such as: strscpy, strscpy_pad, strtomem and strtomem_pad (amongst others). Technically, my patch yields subtly different behavior. The original implementation with `strncpy` would fill the entire destination buffer with null bytes [3] while `strscpy` will leave the junk, uninitialized bytes trailing after the _mandatory_ NUL-termination. So, if somehow `pcm->name` or `card->driver/shortname/longname` require this NUL-padding behavior then `strscpy_pad` should be used. My interpretation, though, is that the aforementioned fields are just fine as NUL-terminated strings. Please correct my assumptions if needed and I'll send in a v2. [1]: www.kernel.org/doc/html/latest/process/deprecated.html#strncpy-on-nul-termi… [2]: manpages.debian.org/testing/linux-manual-4.8/strscpy.9.en.html [3]: https://linux.die.net/man/3/strncpy Link: https://github.com/KSPP/linux/issues/90 Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- sound/xen/xen_snd_front_alsa.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/xen/xen_snd_front_alsa.c b/sound/xen/xen_snd_front_alsa.c index db917453a473..7a3dfce97c15 100644 --- a/sound/xen/xen_snd_front_alsa.c +++ b/sound/xen/xen_snd_front_alsa.c @@ -783,7 +783,7 @@ static int new_pcm_instance(struct xen_snd_front_card_info *card_info, pcm->info_flags = 0; /* we want to handle all PCM operations in non-atomic context */ pcm->nonatomic = true; - strncpy(pcm->name, "Virtual card PCM", sizeof(pcm->name)); + strscpy(pcm->name, "Virtual card PCM", sizeof(pcm->name)); if (instance_cfg->num_streams_pb) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, @@ -835,9 +835,9 @@ int xen_snd_front_alsa_init(struct xen_snd_front_info *front_info) goto fail; } - strncpy(card->driver, XENSND_DRIVER_NAME, sizeof(card->driver)); - strncpy(card->shortname, cfg->name_short, sizeof(card->shortname)); - strncpy(card->longname, cfg->name_long, sizeof(card->longname)); + strscpy(card->driver, XENSND_DRIVER_NAME, sizeof(card->driver)); + strscpy(card->shortname, cfg->name_short, sizeof(card->shortname)); + strscpy(card->longname, cfg->name_long, sizeof(card->longname)); ret = snd_card_register(card); if (ret < 0) --- base-commit: 57012c57536f8814dec92e74197ee96c3498d24e change-id: 20230727-sound-xen-398c9927b3d8 Best regards, -- Justin Stitt <justinstitt(a)google.com>

4 3

[PATCH v3 00/50] Add support for sam9x7 SoC family
by Varshini Rajendran 28 Jul '23

28 Jul '23

This patch series adds support for the new SoC family - sam9x7. - The device tree, configs and drivers are added - Clock driver for sam9x7 is added - Support for basic peripherals is added - Target board SAM9X75 Curiosity is added Changes in v3: -------------- - Fixed the DT documentation errors pointed out in v2. - Dropped Acked-by tag in tcb DT doc patch as it had to be adapted according to sam9x7 correctly. - Picked by the previously missed tags. - Dropped this patch "dt-bindings: usb: generic-ehci: Document clock-names property" as the warning was not found while validating DT-schema for at91-sam9x75_curiosity.dtb. - Dropped redundant words in the commit message. - Fixed the CHECK_DTBS warnings validated against at91-sam9x75_curiosity.dtb. - Renamed dt nodes according to naming convention. - Dropped unwanted status property in dts. - Removed nodes that are not in use from the board dts. - Removed spi DT doc patch from the series as it was already applied and a fix patch was applied subsequently. Added a patch to remove the compatible to adapt sam9x7. - Added sam9x7 compatibles in usb dt documentation. Changes in v2: -------------- - Added sam9x7 specific compatibles in DT with fallbacks - Documented all the newly added DT compatible strings - Added device tree for the target board sam9x75 curiosity and documented the same in the DT bindings documentation - Removed the dt nodes that are not supported at the moment - Removed the configs added by previous version that are not supported at the moment - Fixed all the corrections in the commit message - Changed all the instances of copyright year to 2023 - Added sam9x7 flag in PIT64B configuration - Moved macro definitions to header file - Added another divider in mck characteristics in the pmc driver - Fixed the memory leak in the pmc driver - Dropped patches that are no longer needed - Picked up Acked-by and Reviewed-by tags Hari Prasath (1): irqchip/atmel-aic5: Add support for sam9x7 aic Varshini Rajendran (49): dt-bindings: microchip: atmel,at91rm9200-tcb: add sam9x60, sam9x7 compatible dt-bindings: usb: ehci: Add atmel at91sam9g45-ehci compatible dt-bindings: net: cdns,macb: add sam9x7 ethernet interface dt-bindings: clk: at91: add sam9x7 dt-bindings: clk: at91: add sam9x7 clock controller dt-bindings: reset: atmel,at91sam9260-reset: add sam9x7 dt-bindings: power: reset: atmel,sama5d2-shdwc: add sam9x7 dt-bindings: atmel-sysreg: add sam9x7 dt-bindings: crypto: add sam9x7 in Atmel AES dt-bindings: crypto: add sam9x7 in Atmel SHA dt-bindings: crypto: add sam9x7 in Atmel TDES dt-bindings: dmaengine: at_xdmac: add compatible with microchip,sam9x7 dt-bindings: i2c: at91: Add sam9x7 compatible string dt-bindings: mfd: at91: Add SAM9X7 compatible string dt-bindings: atmel-gpbr: add microchip,sam9x7-gpbr dt-bindings: atmel-matrix: add microchip,sam9x7-matrix dt-bindings: atmel-smc: add microchip,sam9x7-smc dt-bindings: atmel-ssc: add microchip,sam9x7-ssc dt-bindings: sdhci-of-at91: add microchip,sam9x7-sdhci dt-bindings: atmel-nand: add microchip,sam9x7-pmecc dt-bindings: pinctrl: at91: add sam9x7 dt-bindings: rng: atmel,at91-trng: add sam9x7 TRNG dt-bindings: rtc: at91rm9200: add sam9x7 compatible dt-bindings: rtt: at91rm9260: add sam9x7 compatible dt-bindings: serial: atmel,at91-usart: add compatible for sam9x7 dt-bindings: atmel-classd: add sam9x7 compatible dt-bindings: usb: atmel: add sam9x7 dt-bindings: watchdog: sama5d4-wdt: add compatible for sam9x7-wdt dt-bindings: irqchip/atmel-aic5: Add support for sam9x7 aic spi: dt-bindings: atmel,at91rm9200-spi: remove 9x60 compatible from list ASoC: dt-bindings: microchip: add sam9x7 dt-bindings: usb: ehci: Add sam9x7 dt-bindings: usb: add sam9x7 ARM: at91: pm: add support for sam9x7 SoC family ARM: at91: pm: add sam9x7 SoC init config ARM: at91: add support in SoC driver for new sam9x7 clk: at91: clk-sam9x60-pll: re-factor to support individual core freq outputs clk: at91: sam9x7: add support for HW PLL freq dividers clk: at91: sama7g5: move mux table macros to header file clk: at91: sam9x7: Allow PLLs to be exported and referenced in DT clk: at91: sam9x7: add sam9x7 pmc driver power: reset: at91-poweroff: lookup for proper pmc dt node for sam9x7 power: reset: at91-reset: add reset support for sam9x7 SoC power: reset: at91-reset: add sdhwc support for sam9x7 SoC ARM: at91: Kconfig: add config flag for SAM9X7 SoC ARM: configs: at91: enable config flags for sam9x7 SoC family ARM: dts: at91: sam9x7: add device tree for SoC dt-bindings: arm: add sam9x75 curiosity board ARM: dts: at91: sam9x75_curiosity: add sam9x75 curiosity board .../devicetree/bindings/arm/atmel-at91.yaml | 6 + .../devicetree/bindings/arm/atmel-sysregs.txt | 7 +- .../bindings/clock/atmel,at91rm9200-pmc.yaml | 1 + .../bindings/clock/atmel,at91sam9x5-sckc.yaml | 3 + .../crypto/atmel,at91sam9g46-aes.yaml | 6 +- .../crypto/atmel,at91sam9g46-sha.yaml | 6 +- .../crypto/atmel,at91sam9g46-tdes.yaml | 6 +- .../devicetree/bindings/dma/atmel-xdma.txt | 3 +- .../bindings/i2c/atmel,at91sam-i2c.yaml | 4 + .../interrupt-controller/atmel,aic.txt | 2 +- .../devicetree/bindings/mfd/atmel-flexcom.txt | 1 + .../devicetree/bindings/mfd/atmel-gpbr.txt | 1 + .../devicetree/bindings/mfd/atmel-matrix.txt | 1 + .../devicetree/bindings/mfd/atmel-smc.txt | 1 + .../devicetree/bindings/misc/atmel-ssc.txt | 1 + .../devicetree/bindings/mmc/sdhci-atmel.txt | 4 +- .../devicetree/bindings/mtd/atmel-nand.txt | 1 + .../devicetree/bindings/net/cdns,macb.yaml | 6 + .../bindings/pinctrl/atmel,at91-pinctrl.txt | 2 + .../power/reset/atmel,sama5d2-shdwc.yaml | 3 + .../reset/atmel,at91sam9260-reset.yaml | 4 + .../bindings/rng/atmel,at91-trng.yaml | 4 + .../bindings/rtc/atmel,at91rm9200-rtc.yaml | 18 +- .../bindings/rtc/atmel,at91sam9260-rtt.yaml | 3 + .../bindings/serial/atmel,at91-usart.yaml | 9 + .../soc/microchip/atmel,at91rm9200-tcb.yaml | 21 +- .../bindings/sound/atmel,sama5d2-classd.yaml | 7 +- .../sound/microchip,sama7g5-i2smcc.yaml | 11 +- .../bindings/spi/atmel,at91rm9200-spi.yaml | 1 - .../devicetree/bindings/usb/atmel-usb.txt | 11 +- .../devicetree/bindings/usb/generic-ehci.yaml | 7 + .../devicetree/bindings/usb/generic-ohci.yaml | 6 + .../bindings/watchdog/atmel,sama5d4-wdt.yaml | 15 +- arch/arm/boot/dts/microchip/Makefile | 5 + .../dts/microchip/at91-sam9x75_curiosity.dts | 311 +++++ arch/arm/boot/dts/microchip/sam9x7.dtsi | 1238 +++++++++++++++++ arch/arm/configs/at91_dt_defconfig | 1 + arch/arm/mach-at91/Kconfig | 23 +- arch/arm/mach-at91/Makefile | 1 + arch/arm/mach-at91/generic.h | 2 + arch/arm/mach-at91/pm.c | 35 + arch/arm/mach-at91/sam9x7.c | 34 + drivers/clk/at91/Makefile | 1 + drivers/clk/at91/clk-sam9x60-pll.c | 50 +- drivers/clk/at91/pmc.h | 18 + drivers/clk/at91/sam9x60.c | 7 + drivers/clk/at91/sam9x7.c | 946 +++++++++++++ drivers/clk/at91/sama7g5.c | 42 +- drivers/irqchip/irq-atmel-aic5.c | 10 + drivers/power/reset/Kconfig | 4 +- drivers/power/reset/at91-sama5d2_shdwc.c | 1 + drivers/soc/atmel/soc.c | 23 + drivers/soc/atmel/soc.h | 9 + include/dt-bindings/clock/at91.h | 4 + 54 files changed, 2871 insertions(+), 76 deletions(-) create mode 100644 arch/arm/boot/dts/microchip/at91-sam9x75_curiosity.dts create mode 100644 arch/arm/boot/dts/microchip/sam9x7.dtsi create mode 100644 arch/arm/mach-at91/sam9x7.c create mode 100644 drivers/clk/at91/sam9x7.c -- 2.25.1

5 7

[PATCH] ASoC: fsl_micfil: refactor deprecated strncpy
by Justin Stitt 28 Jul '23

28 Jul '23

`strncpy` is deprecated for use on NUL-terminated destination strings [1]. A suitable replacement is `strscpy` [2] due to the fact that it guarantees NUL-termination on its destination buffer argument which is _not_ always the case for `strncpy`! In this case, though, there was great care taken to ensure that the destination buffer would be NUL-terminated through the use of `len - 1` ensuring that the previously zero-initialized buffer would not overwrite the last NUL byte. This means that there's no bug here. However, `strscpy` will add a mandatory NUL byte to the destination buffer as promised by the following `strscpy` implementation [3]: | /* Hit buffer length without finding a NUL; force NUL-termination. */ | if (res) | dest[res-1] = '\0'; This means we can lose the `- 1` which clears up whats happening here. All the while, we get one step closer to eliminating the ambiguous `strncpy` api in favor of its less ambiguous replacement like `strscpy`, `strscpy_pad`, `strtomem` and `strtomem_pad` amongst others. [1]: www.kernel.org/doc/html/latest/process/deprecated.html#strncpy-on-nul-termi… [2]: manpages.debian.org/testing/linux-manual-4.8/strscpy.9.en.html [3]: https://elixir.bootlin.com/linux/v6.3/source/lib/string.c#L183 Link: https://github.com/KSPP/linux/issues/90 Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- sound/soc/fsl/fsl_micfil.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 3f08082a55be..fe28b27e50d0 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -1044,7 +1044,7 @@ static int fsl_micfil_probe(struct platform_device *pdev) return -ENOMEM; micfil->pdev = pdev; - strncpy(micfil->name, np->name, sizeof(micfil->name) - 1); + strscpy(micfil->name, np->name, sizeof(micfil->name)); micfil->soc = of_device_get_match_data(&pdev->dev); --- base-commit: 57012c57536f8814dec92e74197ee96c3498d24e change-id: 20230727-sound-soc-fsl-4fc5569d771e Best regards, -- Justin Stitt <justinstitt(a)google.com>

3 2

[PATCH v2 00/11] Fix support for System Suspend for CS35L41 HDA
by Stefan Binding 28 Jul '23

28 Jul '23

There have been a couple of customer reports of intermittant issues after system resume, where sometimes the DSP firmware stops responding. Investigations into this issue show that there is a race between receiving a prepare from the HDA core, and the firmware reload which is started by the system resume. This can causes the Global Enable on the CS35L41 to be enabled during the firmware load, which can sometimes cause issues in the DSP. The existing system resume behaviour also did not resume the audio, if audio was previously playing when it was suspended. In addition, during investigation, it was found there were additional problems in the System Resume sequence, as well as the Playback sequence with External Boost, where the driver does not correctly follow its enable sequence for this mode. This can cause additional issues such as pops and clicks. This chain intends to correct the sequences for playback and system suspend/resume so that the driver: obeys the external boost enable sequence; resumes audio on system resume; and avoids the race condition on firmware load and playback during system resume. Changes since v1: - Split patch 1 into 2 separate patches - Combine Patches 6 and 9 Stefan Binding (11): ALSA: cs35l41: Use mbox command to enable speaker output for external boost ALSA: cs35l41: Poll for Power Up/Down rather than waiting a fixed delay ALSA: hda: cs35l41: Check mailbox status of pause command after firmware load ALSA: hda: cs35l41: Ensure we correctly re-sync regmap before system suspending. ALSA: hda: cs35l41: Ensure we pass up any errors during system suspend. ALSA: hda: cs35l41: Move Play and Pause into separate functions ALSA: hda: hda_component: Add pre and post playback hooks to hda_component ALSA: hda: cs35l41: Use pre and post playback hooks ALSA: hda: cs35l41: Rework System Suspend to ensure correct call separation ALSA: hda: cs35l41: Add device_link between HDA and cs35l41_hda ALSA: hda: cs35l41: Ensure amp is only unmuted during playback include/sound/cs35l41.h | 5 +- sound/pci/hda/cs35l41_hda.c | 288 +++++++++++++++++++++++++-------- sound/pci/hda/hda_component.h | 2 + sound/pci/hda/patch_realtek.c | 10 +- sound/soc/codecs/cs35l41-lib.c | 118 ++++++++++++-- sound/soc/codecs/cs35l41.c | 18 +-- 6 files changed, 343 insertions(+), 98 deletions(-) -- 2.34.1

2 14

[PATCH 0/5] ASoC: st: Use maple tree register cache
by Mark Brown 28 Jul '23

28 Jul '23

The maple tree register cache has now got to the point where is is roughly feature compatible with the rbtree cache, let's convert the ST drivers to use the more modern data structure. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (5): ASoC: sta32x: Update to use maple tree register cache ASoC: sta350: Update to use maple tree register cache ASoC: sta529: Update to use maple tree register cache ASoC: stac9766: Update to use maple tree register cache ASoC: sti-sas: Update to use maple tree register cache sound/soc/codecs/sta32x.c | 2 +- sound/soc/codecs/sta350.c | 2 +- sound/soc/codecs/sta529.c | 2 +- sound/soc/codecs/stac9766.c | 2 +- sound/soc/codecs/sti-sas.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230701-asoc-st-maple-fc1a8f015154 Best regards, -- Mark Brown <broonie(a)kernel.org>

1 6

[PATCH v3 26/50] dt-bindings: atmel-classd: add sam9x7 compatible
by Varshini Rajendran 28 Jul '23

28 Jul '23

Add sam9x7 compatible to DT bindings documentation. Signed-off-by: Varshini Rajendran <varshini.rajendran(a)microchip.com> --- .../devicetree/bindings/sound/atmel,sama5d2-classd.yaml | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml b/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml index 43d04702ac2d..ae3162fcfe02 100644 --- a/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml +++ b/Documentation/devicetree/bindings/sound/atmel,sama5d2-classd.yaml @@ -18,7 +18,12 @@ description: properties: compatible: - const: atmel,sama5d2-classd + oneOf: + - items: + - const: atmel,sama5d2-classd + - items: + - const: microchip,sam9x7-classd + - const: atmel,sama5d2-classd reg: maxItems: 1 -- 2.25.1

2 1

[PATCH 0/2] Intel: sof_rt5682: remove quirk flag
by Brent Lu 28 Jul '23

28 Jul '23

We add a helper funcion to detect amplifier number according to device instance in ACPI table so the SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag and a dmi quirk for 4-amplifier configuration could be safely removed. Also refactory the max_98390_hw_params() function to use an array to handle the TDM parameter. Amplifier number detection and TDM parameter are tested on two Chromebooks. One with 2 MAX98390 and one with 4 MAX98390 amplifier. *** BLURB HERE *** Brent Lu (2): ASoC: Intel: maxim-common: get codec number from ACPI ASoC: Intel: sof_rt5682: remove SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag sound/soc/intel/boards/sof_maxim_common.c | 174 +++++++++++++--------- sound/soc/intel/boards/sof_maxim_common.h | 21 ++- sound/soc/intel/boards/sof_rt5682.c | 37 +---- 3 files changed, 115 insertions(+), 117 deletions(-) -- 2.34.1

7 15

[PATCH v3 0/2] Intel: sof_rt5682: remove quirk flag
by Brent Lu 28 Jul '23

28 Jul '23

We add a helper function to detect amplifier number according to device instance in ACPI table so the SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag and a dmi quirk for 4-amplifier configuration could be safely removed. Also refactor the max_98390_hw_params() function to use an array to handle the TDM parameter. Amplifier number detection and TDM parameter are tested on two Chromebooks. One with 2 MAX98390 and one with 4 MAX98390 amplifier. V2 Changes: - fix some typos in cover letter - use for_each_acpi_dev_match() macro to simplify the codec - use 'unsign int' for codec number variable V3 Changes: - replace pr_err() with dev_err() Brent Lu (2): ASoC: Intel: maxim-common: get codec number from ACPI ASoC: Intel: sof_rt5682: remove SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag sound/soc/intel/boards/sof_maxim_common.c | 171 +++++++++++++--------- sound/soc/intel/boards/sof_maxim_common.h | 21 ++- sound/soc/intel/boards/sof_rt5682.c | 37 +---- 3 files changed, 112 insertions(+), 117 deletions(-) -- 2.34.1

3 4

[PATCH 0/3] Update the register list of MICFIL
by Chancel Liu 28 Jul '23

28 Jul '23

MICFIL IP is upgraded on i.MX93 platform. Add new registers and new bit definition. Chancel Liu (3): ASoC: fsl_micfil: Add new registers and new bit definition ASoC: fsl_micfil: Add fsl_micfil_check_version 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

2 4

[PATCH] ASoC: rt1017: Remove unused function
by Jiapeng Chong 28 Jul '23

28 Jul '23

The function are defined in the rt1017-sdca-sdw.h file, but not called elsewhere, so delete the unused functions. sound/soc/codecs/rt1017-sdca-sdw.h:183:33: warning: ‘rt1017_sdca_mbq_defaults’ defined but not used. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Closes: https://bugzilla.openanolis.cn/show_bug.cgi?id=5998 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/rt1017-sdca-sdw.h | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/rt1017-sdca-sdw.h b/sound/soc/codecs/rt1017-sdca-sdw.h index 5968b9a7d124..4932b5dbe3c0 100644 --- a/sound/soc/codecs/rt1017-sdca-sdw.h +++ b/sound/soc/codecs/rt1017-sdca-sdw.h @@ -180,9 +180,4 @@ static const struct reg_default rt1017_sdca_reg_defaults[] = { RT1017_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, }; -static const struct reg_default rt1017_sdca_mbq_defaults[] = { - { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_FU, - RT1017_SDCA_CTL_FU_VOLUME, 0x01), 0x00 }, -}; - #endif /* __RT1017_SDW_H__ */ -- 2.20.1.7.g153144c

2 1

[RFC v2 00/11] iio: add iio backend device type
by Olivier Moysan 27 Jul '23

27 Jul '23

This v2 is an addon to initial RFC: https://lore.kernel.org/lkml/20230623140944.2613002-1-olivier.moysan@foss.s… Despite the "IIO backend" naming has to be changed (as pointed out by Jonathan previously), it has been kept here, for time being. The appropriated naming still has to be discussed later on. In the previous RFC the "IIO backend" concept was proposed through a set of template APIs. This v2 implements a functionnal exemple based on STM32 DFSDM, to bring scaling support to this peripheral. Olivier Moysan (11): iio: introduce iio backend device of: property: add device link support for io-backends dt-bindings: iio: stm32-dfsdm-adc: add scaling support dt-bindings: iio: adc: add scaling support to sd modulator iio: adc: stm32-dfsdm: manage dfsdm as a channel provider iio: adc: stm32-dfsdm: adopt generic channel bindings iio: adc: stm32-dfsdm: add scaling support to dfsdm iio: adc: sd modulator: add scale and offset support ARM: dts: stm32: adopt new dfsdm bindings on stm32mp151 ARM: dts: stm32: add dfsdm pins muxing on stm32mp15 ARM: dts: stm32: add dfsdm iio support on stm32mp157c-ev .../iio/adc/sigma-delta-modulator.yaml | 9 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 189 ++++++------------ arch/arm/boot/dts/st/stm32mp15-pinctrl.dtsi | 39 ++++ arch/arm/boot/dts/st/stm32mp151.dtsi | 18 +- arch/arm/boot/dts/st/stm32mp157c-ev1.dts | 68 +++++++ drivers/iio/Makefile | 1 + drivers/iio/adc/sd_adc_modulator.c | 106 ++++++++-- drivers/iio/adc/stm32-dfsdm-adc.c | 187 +++++++++++------ drivers/iio/industrialio-backend.c | 107 ++++++++++ drivers/of/property.c | 2 + include/linux/iio/backend.h | 56 ++++++ 11 files changed, 561 insertions(+), 221 deletions(-) create mode 100644 drivers/iio/industrialio-backend.c create mode 100644 include/linux/iio/backend.h -- 2.25.1

1 0

[GIT PULL] ASoC fixes for v6.5-rc3
by Mark Brown 27 Jul '23

27 Jul '23

The following changes since commit 6eaae198076080886b9e7d57f4ae06fa782f90ef: Linux 6.5-rc3 (2023-07-23 15:24:10 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc3 for you to fetch changes up to f85739c0b2b0d98a32f5ca4fcc5501d2b76df4f6: ASoC: atmel: Fix the 8K sample parameter in I2SC master (2023-07-25 12:10:00 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.5 A collection of device specific fixes, none particularly remarkable. There's a set of repetitive fixes for the RealTek drivers fixing an issue with suspend that was replicated in multiple drivers. ---------------------------------------------------------------- Dmytro Maluka (2): ASoC: da7219: Flush pending AAD IRQ when suspending ASoC: da7219: Check for failure reading AAD IRQ events Edgar (1): ASoc: codecs: ES8316: Fix DMIC config Edson Juliano Drosdeck (1): ASoC: nau8821: Add DMI quirk mechanism for active-high jack-detect Guiting Shen (1): ASoC: atmel: Fix the 8K sample parameter in I2SC master Mark Brown (3): ASoC: Merge up fixes from mainline ASoC: da7219: Patches related to a spurious AAD IRQ ASoC: wm8904: Fill the cache for WM8904_ADC_TEST_0 register Matus Gajdos (1): ASoC: fsl_spdif: Silence output on stop Shuming Fan (5): ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0 ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0 ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0 ASoC: rt711: fix for JD event handling in ClockStop Mode0 ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0 sound/soc/atmel/atmel-i2s.c | 5 ++++- sound/soc/codecs/da7219-aad.c | 12 +++++++++--- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/nau8821.c | 41 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- sound/soc/codecs/rt712-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/rt722-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/wm8904.c | 3 +++ sound/soc/fsl/fsl_spdif.c | 2 ++ 11 files changed, 103 insertions(+), 10 deletions(-)

2 1

[PATCH] ALSA: hda/cs35l56: Do some clean up on probe error
by Dan Carpenter 27 Jul '23

27 Jul '23

Smatch complains that this return should be a goto: sound/pci/hda/cs35l56_hda.c:910 cs35l56_hda_common_probe() warn: missing unwind goto? The goto error disables cansleep so that seems reasonable. Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/pci/hda/cs35l56_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 71e95e64f8a4..4c3279f61b94 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -907,7 +907,7 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int id) ret = cs35l56_set_patch(&cs35l56->base); if (ret) - return ret; + goto err; regcache_mark_dirty(cs35l56->base.regmap); regcache_sync(cs35l56->base.regmap); -- 2.39.2

3 2

[PATCH] ALSA: hda/realtek: Support ASUS G713PV laptop
by Pavel Asyutchenko 27 Jul '23

27 Jul '23

This laptop has CS35L41 amp connected via I2C. With this patch speakers begin to work if the missing _DSD properties are added to ACPI tables. Signed-off-by: Pavel Asyutchenko <svenpavel(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7c0f474ea3b3..dff92679ae72 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9680,6 +9680,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JI", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1d1f, "ASUS ROG Strix G17 2023 (G713PV)", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402", ALC245_FIXUP_CS35L41_SPI_2), -- 2.39.2

2 1

[PATCH v2 0/2] Intel: sof_rt5682: remove quirk flag
by Brent Lu 27 Jul '23

27 Jul '23

We add a helper function to detect amplifier number according to device instance in ACPI table so the SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag and a dmi quirk for 4-amplifier configuration could be safely removed. Also refactor the max_98390_hw_params() function to use an array to handle the TDM parameter. Amplifier number detection and TDM parameter are tested on two Chromebooks. One with 2 MAX98390 and one with 4 MAX98390 amplifier. V2 Changes: - fix some typos in cover letter - use for_each_acpi_dev_match() macro to simplify the codec - use 'unsign int' for codec number variable Brent Lu (2): ASoC: Intel: maxim-common: get codec number from ACPI ASoC: Intel: sof_rt5682: remove SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag sound/soc/intel/boards/sof_maxim_common.c | 171 +++++++++++++--------- sound/soc/intel/boards/sof_maxim_common.h | 21 ++- sound/soc/intel/boards/sof_rt5682.c | 37 +---- 3 files changed, 112 insertions(+), 117 deletions(-) -- 2.34.1

4 8

[PATCH 1/1] ASoC: fsl: fsl_qmc_audio: Fix snd_pcm_format_t values handling
by Herve Codina 26 Jul '23

26 Jul '23

Running sparse on fsl_qmc_audio (make C=1) raises the following warnings: fsl_qmc_audio.c:387:26: warning: restricted snd_pcm_format_t degrades to integer fsl_qmc_audio.c:389:59: warning: incorrect type in argument 1 (different base types) fsl_qmc_audio.c:389:59: expected restricted snd_pcm_format_t [usertype] format fsl_qmc_audio.c:389:59: got unsigned int [assigned] i fsl_qmc_audio.c:564:26: warning: restricted snd_pcm_format_t degrades to integer fsl_qmc_audio.c:569:50: warning: incorrect type in argument 1 (different base types) fsl_qmc_audio.c:569:50: expected restricted snd_pcm_format_t [usertype] format fsl_qmc_audio.c:569:50: got int [assigned] i fsl_qmc_audio.c:573:62: warning: incorrect type in argument 1 (different base types) fsl_qmc_audio.c:573:62: expected restricted snd_pcm_format_t [usertype] format fsl_qmc_audio.c:573:62: got int [assigned] i These warnings are due to snd_pcm_format_t values handling done in the driver. Some macros and functions exist to handle safely these values. Use dedicated macros and functions to remove these warnings. Fixes: 075c7125b11c ("ASoC: fsl: Add support for QMC audio") Signed-off-by: Herve Codina <herve.codina(a)bootlin.com> --- sound/soc/fsl/fsl_qmc_audio.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c index 7cbb8e4758cc..56d6b0b039a2 100644 --- a/sound/soc/fsl/fsl_qmc_audio.c +++ b/sound/soc/fsl/fsl_qmc_audio.c @@ -372,8 +372,8 @@ static int qmc_dai_hw_rule_format_by_channels(struct qmc_dai *qmc_dai, struct snd_mask *f_old = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); unsigned int channels = params_channels(params); unsigned int slot_width; + snd_pcm_format_t format; struct snd_mask f_new; - unsigned int i; if (!channels || channels > nb_ts) { dev_err(qmc_dai->dev, "channels %u not supported\n", @@ -384,10 +384,10 @@ static int qmc_dai_hw_rule_format_by_channels(struct qmc_dai *qmc_dai, slot_width = (nb_ts / channels) * 8; snd_mask_none(&f_new); - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(f_old, i)) { - if (snd_pcm_format_physical_width(i) <= slot_width) - snd_mask_set(&f_new, i); + pcm_for_each_format(format) { + if (snd_mask_test_format(f_old, format)) { + if (snd_pcm_format_physical_width(format) <= slot_width) + snd_mask_set_format(&f_new, format); } } @@ -551,26 +551,26 @@ static const struct snd_soc_dai_ops qmc_dai_ops = { static u64 qmc_audio_formats(u8 nb_ts) { - u64 formats; - unsigned int chan_width; unsigned int format_width; - int i; + unsigned int chan_width; + snd_pcm_format_t format; + u64 formats_mask; if (!nb_ts) return 0; - formats = 0; + formats_mask = 0; chan_width = nb_ts * 8; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + pcm_for_each_format(format) { /* * Support format other than little-endian (ie big-endian or * without endianness such as 8bit formats) */ - if (snd_pcm_format_little_endian(i) == 1) + if (snd_pcm_format_little_endian(format) == 1) continue; /* Support physical width multiple of 8bit */ - format_width = snd_pcm_format_physical_width(i); + format_width = snd_pcm_format_physical_width(format); if (format_width == 0 || format_width % 8) continue; @@ -581,9 +581,9 @@ static u64 qmc_audio_formats(u8 nb_ts) if (format_width > chan_width || chan_width % format_width) continue; - formats |= (1ULL << i); + formats_mask |= pcm_format_to_bits(format); } - return formats; + return formats_mask; } static int qmc_audio_dai_parse(struct qmc_audio *qmc_audio, struct device_node *np, -- 2.41.0

2 1

[PATCH v2] ASoC: rt1316: fix key tone missing
by shumingf＠realtek.com 26 Jul '23

26 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> This patch adds a control that there are three options to control the digital volume output. The user could select "immediately" to make volume updates immediately and avoid key tone missing issues. In default, the driver selects that the volume update when a zero-crossing with a soft ramp. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/codecs/rt1316-sdw.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 721821d9e9af..10a53c8d4874 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -424,6 +424,15 @@ static SOC_ENUM_SINGLE_DECL(rt1316_rx_data_ch_enum, SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0), 0, rt1316_rx_data_ch_select); +static const char * const rt1316_dac_output_vol_select[] = { + "immediately", + "zero crossing", + "zero crossing with soft ramp", +}; + +static SOC_ENUM_SINGLE_DECL(rt1316_dac_vol_ctl_enum, + 0xc010, 6, rt1316_dac_output_vol_select); + static const struct snd_kcontrol_new rt1316_snd_controls[] = { /* I2S Data Channel Selection */ @@ -442,6 +451,9 @@ static const struct snd_kcontrol_new rt1316_snd_controls[] = { /* IV mixer Control */ SOC_DOUBLE("Isense Mixer Switch", 0xc605, 2, 0, 1, 1), SOC_DOUBLE("Vsense Mixer Switch", 0xc605, 3, 1, 1, 1), + + /* DAC Output Volume Control */ + SOC_ENUM("DAC Output Vol Control", rt1316_dac_vol_ctl_enum), }; static const struct snd_kcontrol_new rt1316_sto_dac = -- 2.34.1

2 1

[PATCH] ASoC: intel: avs: refactor strncpy usage in topology
by justinstitt＠google.com 26 Jul '23

26 Jul '23

`strncpy` is deprecated for use on NUL-terminated destination strings [1]. A suitable replacement is `strscpy` [2]. There are some hopes that someday the `strncpy` api could be ripped out due to the vast number of suitable replacements (strscpy, strscpy_pad, strtomem, strtomem_pad, strlcpy) [1]. [1]: www.kernel.org/doc/html/latest/process/deprecated.html#strncpy-on-nul-termi… [2]: manpages.debian.org/testing/linux-manual-4.8/strscpy.9.en.html --- Link: https://github.com/KSPP/linux/issues/90 Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- sound/soc/intel/avs/topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index cdb4ec500261..45d0eb2a8e71 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -1388,12 +1388,12 @@ static int avs_route_load(struct snd_soc_component *comp, int index, port = __ffs(mach->mach_params.i2s_link_mask); snprintf(buf, len, route->source, port); - strncpy((char *)route->source, buf, len); + strscpy((char *)route->source, buf, len); snprintf(buf, len, route->sink, port); - strncpy((char *)route->sink, buf, len); + strscpy((char *)route->sink, buf, len); if (route->control) { snprintf(buf, len, route->control, port); - strncpy((char *)route->control, buf, len); + strscpy((char *)route->control, buf, len); } } --- base-commit: 0b4a9fdc9317440a71d4d4c264a5650bf4a90f3c change-id: 20230725-sound-soc-intel-avs-remove-deprecated-strncpy-2bc41a5a5f81 Best regards, -- Justin Stitt <justinstitt(a)google.com>

4 4

[PATCH] ALSA: usb-audio: Update for native DSD support quirks
by Jussi Laako 26 Jul '23

26 Jul '23

Maintenance patch for native DSD support. Remove incorrect T+A device quirks. Move set of device quirks to vendor quirks. Add set of missing device and vendor quirks. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 34 ++++++++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 6 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index f9e200309b9e..598659d761cc 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1876,8 +1876,10 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { - case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */ - case USB_ID(0x21ed, 0xd75a): /* Accuphase DAC-60 option card */ + case USB_ID(0x139f, 0x5504): /* Nagra DAC */ + case USB_ID(0x20b1, 0x3089): /* Mola-Mola DAC */ + case USB_ID(0x2522, 0x0007): /* LH Labs Geek Out 1V5 */ + case USB_ID(0x2522, 0x0009): /* LH Labs Geek Pulse X Inifinity 2V0 */ case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) @@ -1887,14 +1889,18 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ case USB_ID(0x16d0, 0x06b2): /* NuPrime DAC-10 */ - case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ + case USB_ID(0x16d0, 0x06b4): /* NuPrime Audio HD-AVP/AVA */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ + case USB_ID(0x16d0, 0x09d8): /* NuPrime IDA-8 */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ + case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */ + case USB_ID(0x20a0, 0x4143): /* WaveIO USB Audio 2.0 */ case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */ case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */ case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */ case USB_ID(0x2622, 0x0041): /* Audiolab M-DAC+ */ + case USB_ID(0x278b, 0x5100): /* Rotel RC-1590 */ case USB_ID(0x27f7, 0x3002): /* W4S DAC-2v2SE */ case USB_ID(0x29a2, 0x0086): /* Mutec MC3+ USB */ case USB_ID(0x6b42, 0x0042): /* MSB Technology */ @@ -1904,9 +1910,6 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* Amanero Combo384 USB based DACs with native DSD support */ case USB_ID(0x16d0, 0x071a): /* Amanero - Combo384 */ - case USB_ID(0x2ab6, 0x0004): /* T+A DAC8DSD-V2.0, MP1000E-V2.0, MP2000R-V2.0, MP2500R-V2.0, MP3100HV-V2.0 */ - case USB_ID(0x2ab6, 0x0005): /* T+A USB HD Audio 1 */ - case USB_ID(0x2ab6, 0x0006): /* T+A USB HD Audio 2 */ if (fp->altsetting == 2) { switch (le16_to_cpu(chip->dev->descriptor.bcdDevice)) { case 0x199: @@ -2051,6 +2054,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x0644, 0x805f, /* TEAC Model 12 */ QUIRK_FLAG_FORCE_IFACE_RESET), + DEVICE_FLG(0x0644, 0x806b, /* TEAC UD-701 */ + QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY | + QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x06f8, 0xb000, /* Hercules DJ Console (Windows Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ @@ -2089,6 +2095,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), DEVICE_FLG(0x154e, 0x3006, /* Marantz SA-14S1 */ QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), + DEVICE_FLG(0x154e, 0x300b, /* Marantz SA-KI RUBY / SA-12 */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x154e, 0x500e, /* Denon DN-X1600 */ QUIRK_FLAG_IGNORE_CLOCK_SOURCE), DEVICE_FLG(0x1686, 0x00dd, /* Zoom R16/24 */ @@ -2133,6 +2141,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x21b4, 0x0081, /* AudioQuest DragonFly */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x21b4, 0x0230, /* Ayre QB-9 Twenty */ + QUIRK_FLAG_DSD_RAW), + DEVICE_FLG(0x21b4, 0x0232, /* Ayre QX-5 Twenty */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2522, 0x0007, /* LH Labs Geek Out HD Audio 1V5 */ QUIRK_FLAG_SET_IFACE_FIRST), DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ @@ -2175,12 +2187,18 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_VALIDATE_RATES), VENDOR_FLG(0x1235, /* Focusrite Novation */ QUIRK_FLAG_VALIDATE_RATES), + VENDOR_FLG(0x1511, /* AURALiC */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x152a, /* Thesycon devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x18d1, /* iBasso devices */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x1de7, /* Phoenix Audio */ QUIRK_FLAG_GET_SAMPLE_RATE), VENDOR_FLG(0x20b1, /* XMOS based devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x21ed, /* Accuphase Laboratory */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x22d9, /* Oppo */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x23ba, /* Playback Design */ @@ -2196,10 +2214,14 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x2ab6, /* T+A devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x2d87, /* Cayin device */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3336, /* HEM devices */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3353, /* Khadas devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x35f4, /* MSB Technology */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3842, /* EVGA */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0xc502, /* HiBy devices */ -- 2.34.1

2 1

[PATCH v2 00/11] Introduce STM32 Firewall framework
by Gatien Chevallier 26 Jul '23

26 Jul '23

Introduce STM32 Firewall framework for STM32MP1x and STM32MP2x platforms. STM32MP1x(ETZPC) and STM32MP2x(RIFSC) Firewall controllers register to the framework to offer firewall services such as access granting. This series of patches is a new approach on the previous STM32 system bus, history is available here: https://lore.kernel.org/lkml/20230127164040.1047583/ The need for such framework arises from the fact that there are now multiple hardware firewalls implemented across multiple products. Drivers are shared between different products, using the same code. When it comes to firewalls, the purpose mostly stays the same: Protect hardware resources. But the implementation differs, and there are multiple types of firewalls: peripheral, memory, ... Some hardware firewall controllers such as the RIFSC implemented on STM32MP2x platforms may require to take ownership of a resource before being able to use it, hence the requirement for firewall services to take/release the ownership of such resources. On the other hand, hardware firewall configurations are becoming more and more complex. These mecanisms prevent platform crashes or other firewall-related incoveniences by denying access to some resources. The stm32 firewall framework offers an API that is defined in firewall controllers drivers to best fit the specificity of each firewall. For every peripherals protected by either the ETZPC or the RIFSC, the firewall framework checks the firewall controlelr registers to see if the peripheral's access is granted to the Linux kernel. If not, the peripheral is configured as secure, the node is marked populated, so that the driver is not probed for that device. The firewall framework relies on the feature-domain-controller device tree bindings: https://lore.kernel.org/lkml/0c0a82bb-18ae-d057-562b. It is used by peripherals to reference a domain controller, in this case a firewall feature domain. The bus uses the ID referenced by the feature-domains property to know where to look in the firewall to get the security configuration for the peripheral. This allows a device tree description rather than a hardcoded peripheral table in the bus driver. The STM32 ETZPC device is responsible for filtering accesses based on security level, or co-processor isolation for any resource connected to it. The RIFSC is responsible for filtering accesses based on Compartment ID / security level / privilege level for any resource connected to it. STM32MP13/15/25 SoC device tree files are updated in this series to implement this mecanism. Changes in V2: generic: - Add fw_devlink dependency for "feature-domains" property. bindings: - Corrected YAMLS errors highlighted by Rob's robot - Firewall controllers YAMLs no longer define the maxItems for the "feature-domains" property - Renamed st,stm32-rifsc.yaml to st,stm32mp25-rifsc.yaml - Fix examples in YAML files - Change feature-domains maxItems to 2 in firewall consumer files as there should not be more than 2 entries for now - Declare "feature-domain-names" as an optional property for firewall controllers child nodes. - Add missing "feature-domains" property declaration in bosch,m_can.yaml and st,stm32-cryp.yaml files firewall framework: - Support multiple entries for "feature-domains" property - Better handle the device-tree parsing using phandle+args APIs - Remove "resource firewall" type - Add a field for the name of the firewall entry - Fix licenses RIFSC: - Add controller name - Driver is now a module_platform_driver - Fix license ETZPC: - Add controller name - Driver is now a module_platform_driver - Fix license Device trees: - Fix rifsc node name - Move the "ranges" property under the "feature-domains" one Oleksii Moisieiev (1): dt-bindings: Document common device controller bindings Gatien Chevallier (10): dt-bindings: bus: document RIFSC dt-bindings: bus: document ETZPC dt-bindings: treewide: add feature-domains description firewall: introduce stm32_firewall framework of: property: fw_devlink: Add support for "feature-domains" bus: rifsc: introduce RIFSC firewall controller driver arm64: dts: st: add RIFSC as a domain controller for STM32MP25x boards bus: etzpc: introduce ETZPC firewall controller driver ARM: dts: stm32: add ETZPC as a system bus for STM32MP15x boards ARM: dts: stm32: add ETZPC as a system bus for STM32MP13x boards .../bindings/bus/st,stm32-etzpc.yaml | 96 + .../bindings/bus/st,stm32mp25-rifsc.yaml | 105 + .../bindings/crypto/st,stm32-cryp.yaml | 4 + .../bindings/crypto/st,stm32-hash.yaml | 4 + .../devicetree/bindings/dma/st,stm32-dma.yaml | 4 + .../bindings/dma/st,stm32-dmamux.yaml | 4 + .../feature-domain-controller.yaml | 84 + .../devicetree/bindings/i2c/st,stm32-i2c.yaml | 4 + .../bindings/iio/adc/st,stm32-adc.yaml | 4 + .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 4 + .../bindings/iio/dac/st,stm32-dac.yaml | 4 + .../bindings/media/cec/st,stm32-cec.yaml | 4 + .../bindings/media/st,stm32-dcmi.yaml | 4 + .../memory-controllers/st,stm32-fmc2-ebi.yaml | 4 + .../bindings/mfd/st,stm32-lptimer.yaml | 4 + .../bindings/mfd/st,stm32-timers.yaml | 5 + .../devicetree/bindings/mmc/arm,pl18x.yaml | 4 + .../bindings/net/can/bosch,m_can.yaml | 4 + .../devicetree/bindings/net/stm32-dwmac.yaml | 4 + .../bindings/phy/phy-stm32-usbphyc.yaml | 4 + .../bindings/regulator/st,stm32-vrefbuf.yaml | 4 + .../devicetree/bindings/rng/st,stm32-rng.yaml | 4 + .../bindings/serial/st,stm32-uart.yaml | 4 + .../bindings/sound/st,stm32-i2s.yaml | 4 + .../bindings/sound/st,stm32-sai.yaml | 4 + .../bindings/sound/st,stm32-spdifrx.yaml | 4 + .../bindings/spi/st,stm32-qspi.yaml | 4 + .../devicetree/bindings/spi/st,stm32-spi.yaml | 4 + .../devicetree/bindings/usb/dwc2.yaml | 4 + MAINTAINERS | 7 + arch/arm/boot/dts/st/stm32mp131.dtsi | 1027 +++--- arch/arm/boot/dts/st/stm32mp133.dtsi | 51 +- arch/arm/boot/dts/st/stm32mp13xc.dtsi | 19 +- arch/arm/boot/dts/st/stm32mp13xf.dtsi | 19 +- arch/arm/boot/dts/st/stm32mp151.dtsi | 2757 +++++++++-------- arch/arm/boot/dts/st/stm32mp153.dtsi | 52 +- arch/arm/boot/dts/st/stm32mp15xc.dtsi | 19 +- arch/arm64/Kconfig.platforms | 1 + arch/arm64/boot/dts/st/stm32mp251.dtsi | 7 +- drivers/bus/Kconfig | 9 + drivers/bus/Makefile | 1 + drivers/bus/stm32_etzpc.c | 141 + drivers/bus/stm32_firewall.c | 288 ++ drivers/bus/stm32_firewall.h | 83 + drivers/bus/stm32_rifsc.c | 252 ++ drivers/of/property.c | 2 + include/linux/bus/stm32_firewall_device.h | 140 + 47 files changed, 3346 insertions(+), 1919 deletions(-) create mode 100644 Documentation/devicetree/bindings/bus/st,stm32-etzpc.yaml create mode 100644 Documentation/devicetree/bindings/bus/st,stm32mp25-rifsc.yaml create mode 100644 Documentation/devicetree/bindings/feature-controllers/feature-domain-controller.yaml create mode 100644 drivers/bus/stm32_etzpc.c create mode 100644 drivers/bus/stm32_firewall.c create mode 100644 drivers/bus/stm32_firewall.h create mode 100644 drivers/bus/stm32_rifsc.c create mode 100644 include/linux/bus/stm32_firewall_device.h -- 2.35.3

3 22

[PATCH v2] slimbus: Use dev_err_probe instead of dev_err
by Wang Ming 26 Jul '23

26 Jul '23

It is possible that dma_request_chan will return EPROBE_DEFER, which means that dev is not ready yet. In this case, dev_err(dev), there will be no output. This patch fixes the bug. Signed-off-by: Wang Ming <machel(a)vivo.com> --- drivers/slimbus/qcom-ngd-ctrl.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/slimbus/qcom-ngd-ctrl.c b/drivers/slimbus/qcom-ngd-ctrl.c index 77aa6d26476c..d5c2a732f63f 100644 --- a/drivers/slimbus/qcom-ngd-ctrl.c +++ b/drivers/slimbus/qcom-ngd-ctrl.c @@ -679,8 +679,8 @@ static int qcom_slim_ngd_init_rx_msgq(struct qcom_slim_ngd_ctrl *ctrl) ctrl->dma_rx_channel = dma_request_chan(dev, "rx"); if (IS_ERR(ctrl->dma_rx_channel)) { - dev_err(dev, "Failed to request RX dma channel"); - ret = PTR_ERR(ctrl->dma_rx_channel); + ret = dev_err_probe(dev, PTR_ERR(ctrl->dma_rx_channel), + "Failed to request RX dma channel"); ctrl->dma_rx_channel = NULL; return ret; } @@ -717,8 +717,8 @@ static int qcom_slim_ngd_init_tx_msgq(struct qcom_slim_ngd_ctrl *ctrl) ctrl->dma_tx_channel = dma_request_chan(dev, "tx"); if (IS_ERR(ctrl->dma_tx_channel)) { - dev_err(dev, "Failed to request TX dma channel"); - ret = PTR_ERR(ctrl->dma_tx_channel); + ret = dev_err_probe(dev, PTR_ERR(ctrl->dma_tx_channel), + "Failed to request TX dma channel"); ctrl->dma_tx_channel = NULL; return ret; } -- 2.25.1

1 0

topology: nhlt: intel: support more device types and directions
by GitHub pull_request - opened 26 Jul '23

26 Jul '23

alsa-project/alsa-utils pull request #226 was opened from brentlu: In current NHLT table the device type of all SSP endpoints are set to BT Sideband(0) instead of SSP Analog Codec(4) and the direction only supports Render(0) and Capture(1). Here we introduce two new quirks from topology to set the device type correctly and support two more directions: Render with loopback(2) and Feedback for render(3) for speakers with echo reference or IV sense feedback. Request URL : https://github.com/alsa-project/alsa-utils/pull/226 Patch URL : https://github.com/alsa-project/alsa-utils/pull/226.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: compress: add opus codec define
by Robert Lee 26 Jul '23

26 Jul '23

Add to support decode Opus codec in Ogg container. Signed-off-by: Robert Lee <lerobert(a)google.com> --- include/uapi/sound/compress_params.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/uapi/sound/compress_params.h b/include/uapi/sound/compress_params.h index ddc77322d571..bac5797bcb02 100644 --- a/include/uapi/sound/compress_params.h +++ b/include/uapi/sound/compress_params.h @@ -43,7 +43,8 @@ #define SND_AUDIOCODEC_BESPOKE ((__u32) 0x0000000E) #define SND_AUDIOCODEC_ALAC ((__u32) 0x0000000F) #define SND_AUDIOCODEC_APE ((__u32) 0x00000010) -#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_APE +#define SND_AUDIOCODEC_OPUS ((__u32) 0x00000011) +#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_OPUS /* * Profile and modes are listed with bit masks. This allows for a -- 2.41.0.487.g6d72f3e995-goog

3 5

[PATCH 00/26] Add support for QMC HDLC, framer infrastruture and PEF2256 framer
by Herve Codina 26 Jul '23

26 Jul '23

Hi, I have a system where I need to handle an HDLC interface and some audio data. The HDLC data are transferred using a TDM bus on which a PEF2256 (E1/T1 framer) is present. The PEF2256 transfers data from/to the TDM bus to/from the E1 line. This PEF2256 is connected to a PowerQUICC SoC for the control path and the TDM is connected to the SoC (QMC component) for the data path. From the QMC HDLC driver, I need to handle HDLC data using the QMC, carrier detection using the PEF2256 (E1 line carrier) and set/get some PEF2256 configuration. The QMC HDLC driver considers the PEF2256 as a generic framer. It performs operations that involve the PEF2256 through the generic framer API. The audio data are exchanged with the PEF2256 using a CPU DAI connected to the TDM bus through the QMC and the PEF2256 needs to be seen as a codec in order to be linked to the CPU DAI. The codec handles the carrier detection using the PEF2256 and reports the carrier state to the ALSA subsystem using the ASoC jack detection. The codec, even if instantiated by the PEF2256 driver, considers the PEF2256 as a generic framer. The generic framer has: - 2 consumers (QMC HDLC drv and codec) - 1 provider (PEF2256) So, the design is the following: +------------------+ +---------+ | QMC | <- TDM -> | PEF2256 | <-> E1 +---------+ | +-------------+ | | | | CPU DAI | <-data--> | QMC channel | | | | +---------+ | +-------------+ | | | +--------------+ | +-------------+ | | | | QMC HDLC drv | <-data--> | QMC channel | | | | +--------------+ | +-------------+ | | | ^ +------------------+ | | | +--------+ +-------------+ | | +-> | framer | <-> | PEF2256 drv | <- local bus ->| | | | | | +---------+ +-> | | | | | +--------+ | +-------+ | +-------------------> | codec | | | +-------+ | +-------------+ Further more, the TDM timeslots used by the QMC HDLC driver need to be configured at runtime (QMC dynamic timeslots). Several weeks ago, I sent two series related to this topic: - Add the Lantiq PEF2256 audio support [1] - RFC Add support for QMC HDLC and PHY [2] This current series is a rework of these two series taking into account feedbacks previously received. In order to implement all of this, I do the following: 1) Perform some fixes (patches 1, 2) 2) Introduce the QMC HDLC driver (patches 3, 4, 5) 3) Add QMC dynamic timeslot support (patches 6 - 16) 4) Add timeslots change support in QMC HDLC (patch 17) 5) Introduce framer infrastructure (patch 18) 6) Add PEF2256 framer provider (patches 19, 20, 21, 22, 23) 7) Add framer codec as a framer consumer (patch 24) 8) Add framer support as a framer consumer in QMC HDLC (patch 25, 26) The series contains the full story and detailed modifications. If needed, the series can be split and/or commmits can be squashed. Let me know. Best regards, Hervé [1]: https://lore.kernel.org/all/20230417171601.74656-1-herve.codina@bootlin.com/ [2]: https://lore.kernel.org/all/20230323103154.264546-1-herve.codina@bootlin.co… Herve Codina (26): soc: fsl: cpm1: qmc: Fix rx channel reset soc: fsl: cpm1: qmc: Extend the API to provide Rx status dt-bindings: net: Add support for QMC HDLC net: wan: Add support for QMC HDLC MAINTAINERS: Add the Freescale QMC HDLC driver entry soc: fsl: cpm1: qmc: Introduce available timeslots masks soc: fsl: cpm1: qmc: Rename qmc_setup_tsa* to qmc_init_tsa* soc: fsl: cpm1: qmc: Introduce qmc_chan_setup_tsa* soc: fsl: cpm1: qmc: Remove no more needed checks from qmc_check_chans() soc: fsl: cpm1: qmc: Check available timeslots in qmc_check_chans() soc: fsl: cpm1: qmc: Add support for disabling channel TSA entries soc: fsl: cpm1: qmc: Split Tx and Rx TSA entries setup soc: fsl: cpm1: qmc: Introduce is_tsa_64rxtx flag soc: fsl: cpm1: qmc: Handle timeslot entries at channel start() and stop() soc: fsl: cpm1: qmc: Remove timeslots handling from setup_chan() soc: fsl: cpm1: qmc: Introduce functions to change timeslots at runtime wan: qmc_hdlc: Add runtime timeslots changes support net: wan: Add framer framework support dt-bindings: net: Add the Lantiq PEF2256 E1/T1/J1 framer mfd: core: Ensure disabled devices are skiped without aborting net: wan: framer: Add support for the Lantiq PEF2256 framer pinctrl: Add support for the Lantic PEF2256 pinmux MAINTAINERS: Add the Lantiq PEF2256 driver entry ASoC: codecs: Add support for the framer codec dt-bindings: net: fsl,qmc-hdlc: Add framer support net: wan: fsl_qmc_hdlc: Add framer support .../devicetree/bindings/net/fsl,qmc-hdlc.yaml | 46 + .../bindings/net/lantiq,pef2256.yaml | 226 +++++ MAINTAINERS | 17 + drivers/mfd/mfd-core.c | 18 +- drivers/net/wan/Kconfig | 14 + drivers/net/wan/Makefile | 3 + drivers/net/wan/framer/Kconfig | 35 + drivers/net/wan/framer/Makefile | 7 + drivers/net/wan/framer/framer-core.c | 935 ++++++++++++++++++ drivers/net/wan/framer/pef2256/Makefile | 8 + drivers/net/wan/framer/pef2256/pef2256-regs.h | 250 +++++ drivers/net/wan/framer/pef2256/pef2256.c | 880 +++++++++++++++++ drivers/net/wan/fsl_qmc_hdlc.c | 820 +++++++++++++++ drivers/pinctrl/Kconfig | 14 + drivers/pinctrl/Makefile | 1 + drivers/pinctrl/pinctrl-pef2256-regs.h | 65 ++ drivers/pinctrl/pinctrl-pef2256.c | 310 ++++++ drivers/soc/fsl/qe/qmc.c | 467 +++++++-- include/linux/framer/framer-provider.h | 194 ++++ include/linux/framer/framer.h | 215 ++++ include/linux/framer/pef2256.h | 31 + include/soc/fsl/qe/qmc.h | 25 +- sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/framer-codec.c | 423 ++++++++ sound/soc/fsl/fsl_qmc_audio.c | 2 +- 26 files changed, 4929 insertions(+), 94 deletions(-) create mode 100644 Documentation/devicetree/bindings/net/fsl,qmc-hdlc.yaml create mode 100644 Documentation/devicetree/bindings/net/lantiq,pef2256.yaml create mode 100644 drivers/net/wan/framer/Kconfig create mode 100644 drivers/net/wan/framer/Makefile create mode 100644 drivers/net/wan/framer/framer-core.c create mode 100644 drivers/net/wan/framer/pef2256/Makefile create mode 100644 drivers/net/wan/framer/pef2256/pef2256-regs.h create mode 100644 drivers/net/wan/framer/pef2256/pef2256.c create mode 100644 drivers/net/wan/fsl_qmc_hdlc.c create mode 100644 drivers/pinctrl/pinctrl-pef2256-regs.h create mode 100644 drivers/pinctrl/pinctrl-pef2256.c create mode 100644 include/linux/framer/framer-provider.h create mode 100644 include/linux/framer/framer.h create mode 100644 include/linux/framer/pef2256.h create mode 100644 sound/soc/codecs/framer-codec.c -- 2.41.0

3 28

MSI MAG X570S TOMAHAWK MAX WIFI - microphone + front audio panel not working - attempt 2
by GitHub issues - edited 26 Jul '23

26 Jul '23

alsa-project/alsa-ucm-conf issue #337 was edited from motomyslen: I tried with `alsa-ucm-conf-1.2.9` but microphone still not working. Front audio panel detects headphones but no sound and mic not working as well. If I'm not wrong ID of my device is `0db0:a073` and it was added a year ago to `USB-Audio.conf` but still something is wrong. Do I need to install alsa-utils 1.2.9? [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12165950… Related https://github.com/alsa-project/alsa-ucm-conf/issues/186 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/337 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MSI MAG X570S TOMAHAWK MAX WIFI - microphone + front audio panel not working - attempt 2
by GitHub issues - edited 26 Jul '23

26 Jul '23

alsa-project/alsa-ucm-conf issue #337 was edited from motomyslen: I tried with `alsa-ucm-conf-1.2.9` but microphone still not working. Front audio panel detects headphones but no sound and mic not working as well. If I'm not wrong ID of my device is `0db0:a073` and it was added a year ago to `USB-Audio.conf`. Do also need to install alsa-utils 1.2.9? [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12165950… Related https://github.com/alsa-project/alsa-ucm-conf/issues/186 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/337 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MSI MAG X570S TOMAHAWK MAX WIFI - microphone + front audio panel not working - attempt 2
by GitHub issues - edited 26 Jul '23

26 Jul '23

alsa-project/alsa-ucm-conf issue #337 was edited from motomyslen: I tried with `alsa-ucm-conf-1.2.9` but microphone still not working. Front audio panel detects headphones but no sound and mic not working as well. If I'm not wrong ID of my device is `0db0:a073` but it was not added in 1.2.9 or 1.2.8 as I checked in changelog. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12165950… Related https://github.com/alsa-project/alsa-ucm-conf/issues/186 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/337 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

MSI MAG X570S TOMAHAWK MAX WIFI - microphone + front audio panel not working - attempt 2
by GitHub issues - edited 26 Jul '23

26 Jul '23

alsa-project/alsa-ucm-conf issue #337 was edited from motomyslen: I tried with `alsa-ucm-conf-1.2.9` but microphone still not working. Front audio panel detects headphones but no sound and mic not working as well. If I'm not wrong ID of my device is `0db0:a073` but it was not added in 1.2.9 or 1.2.8 as I checked in changelog. [alsa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/12165950… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/337 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH -next] ASoC: cs42l51: change cs42l51_of_match to static
by Yang Yingliang 25 Jul '23

25 Jul '23

cs42l51_of_match is only used in cs42l51-i2c.c now, change it to static. Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> --- sound/soc/codecs/cs42l51-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c index e7db7bcd0296..5ed2ef83dcdb 100644 --- a/sound/soc/codecs/cs42l51-i2c.c +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -19,7 +19,7 @@ static struct i2c_device_id cs42l51_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs42l51_i2c_id); -const struct of_device_id cs42l51_of_match[] = { +static const struct of_device_id cs42l51_of_match[] = { { .compatible = "cirrus,cs42l51", }, { } }; -- 2.25.1

3 2

[PATCH v2 5/5] ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 25 Jul '23

25 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> When the system suspends, peripheral SDCA interrupts are disabled. When system level resume is invoked, the peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/codecs/rt722-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index a9416bd163e8..a38ec5862214 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -463,8 +463,16 @@ static int __maybe_unused rt722_sdca_dev_resume(struct device *dev) if (!rt722->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt722->disable_irq == true) { + mutex_lock(&rt722->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt722->disable_irq = false; + mutex_unlock(&rt722->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT722_PROBE_TIMEOUT)); -- 2.34.1

2 1

[PATCH v2 4/5] ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 25 Jul '23

25 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> When the system suspends, peripheral SDCA interrupts are disabled. When system level resume is invoked, the peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/codecs/rt712-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt712-sdca-sdw.c b/sound/soc/codecs/rt712-sdca-sdw.c index ad06267b0ea0..6bc50396a0f6 100644 --- a/sound/soc/codecs/rt712-sdca-sdw.c +++ b/sound/soc/codecs/rt712-sdca-sdw.c @@ -438,8 +438,16 @@ static int __maybe_unused rt712_sdca_dev_resume(struct device *dev) if (!rt712->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt712->disable_irq == true) { + mutex_lock(&rt712->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt712->disable_irq = false; + mutex_unlock(&rt712->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT712_PROBE_TIMEOUT)); -- 2.34.1

2 1

[PATCH v2 3/5] ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 25 Jul '23

25 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> When the system suspends, peripheral SDCA interrupts are disabled. When system level resume is invoked, the peripheral SDCA interrupts should be enabled to handle JD events. Enable SDCA interrupts in resume sequence when ClockStop Mode0 is applied. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 13a5316ba999..a5f4aa0453f5 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -442,8 +442,16 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_0); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.34.1

2 1

[PATCH v2 2/5] ASoC: rt711: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 25 Jul '23

25 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> When the system suspends, peripheral Imp-defined interrupt is disabled. When system level resume is invoked, the peripheral Imp-defined interrupts should be enabled to handle JD events. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdw.c b/sound/soc/codecs/rt711-sdw.c index 87dafcb4545d..33dced388f9e 100644 --- a/sound/soc/codecs/rt711-sdw.c +++ b/sound/soc/codecs/rt711-sdw.c @@ -538,8 +538,15 @@ static int __maybe_unused rt711_dev_resume(struct device *dev) if (!rt711->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt711->disable_irq == true) { + mutex_lock(&rt711->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt711->disable_irq = false; + mutex_unlock(&rt711->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT711_PROBE_TIMEOUT)); -- 2.34.1

2 1

[PATCH v2 1/5] ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 25 Jul '23

25 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> When the system suspends, peripheral Imp-defined interrupt is disabled. When system level resume is invoked, the peripheral Imp-defined interrupts should be enabled to handle JD events. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index 67404f45389f..4968a8c0064d 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -750,8 +750,15 @@ static int __maybe_unused rt5682_dev_resume(struct device *dev) if (!rt5682->first_hw_init) return 0; - if (!slave->unattach_request) + if (!slave->unattach_request) { + if (rt5682->disable_irq == true) { + mutex_lock(&rt5682->disable_irq_lock); + sdw_write_no_pm(slave, SDW_SCP_INTMASK1, SDW_SCP_INT1_IMPL_DEF); + rt5682->disable_irq = false; + mutex_unlock(&rt5682->disable_irq_lock); + } goto regmap_sync; + } time = wait_for_completion_timeout(&slave->initialization_complete, msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); -- 2.34.1

2 1

[PATCH] ASoc: codecs: ES8316: Fix DMIC config
by Edgar 25 Jul '23

25 Jul '23

According to the datasheet, the DMIC config should be changed to { 0, 2 ,3 } Signed-off-by: Edgar <ljijcj(a)163.com> --- sound/soc/codecs/es8316.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 34cf60769b62..65e497b455d3 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -153,7 +153,7 @@ static const char * const es8316_dmic_txt[] = { "dmic data at high level", "dmic data at low level", }; -static const unsigned int es8316_dmic_values[] = { 0, 1, 2 }; +static const unsigned int es8316_dmic_values[] = { 0, 2, 3 }; static const struct soc_enum es8316_dmic_src_enum = SOC_VALUE_ENUM_SINGLE(ES8316_ADC_DMIC, 0, 3, ARRAY_SIZE(es8316_dmic_txt), -- 2.34.1

2 1

[RFC PATCH] ASoC: soc-dai: don't call PCM audio ops if the stream is not supported
by Peter Suti 25 Jul '23

25 Jul '23

PCM audio ops may be called when the stream is not supported. We should not call the ops in that case to avoid unexpected behavior. hw_params is handled already in soc-pcm.c [0] https://lore.kernel.org/alsa-devel/ae06b00a-f3f7-f9d1-0b58-4d71f3394416@lin… Signed-off-by: Peter Suti <peter.suti(a)streamunlimited.com> --- Tested on 6.1 Compile tested on next/master sound/soc/soc-dai.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/soc-dai.c b/sound/soc/soc-dai.c index 02dd64dea179..73a97ac6ccb8 100644 --- a/sound/soc/soc-dai.c +++ b/sound/soc/soc-dai.c @@ -424,6 +424,9 @@ int snd_soc_dai_startup(struct snd_soc_dai *dai, { int ret = 0; + if (!snd_soc_dai_stream_valid(dai, substream->stream)) + return 0; + if (dai->driver->ops && dai->driver->ops->startup) ret = dai->driver->ops->startup(substream, dai); @@ -439,6 +442,9 @@ void snd_soc_dai_shutdown(struct snd_soc_dai *dai, struct snd_pcm_substream *substream, int rollback) { + if (!snd_soc_dai_stream_valid(dai, substream->stream)) + return; + if (rollback && !soc_dai_mark_match(dai, substream, startup)) return; @@ -603,6 +609,8 @@ int snd_soc_pcm_dai_prepare(struct snd_pcm_substream *substream) int i, ret; for_each_rtd_dais(rtd, i, dai) { + if (!snd_soc_dai_stream_valid(dai, substream->stream)) + continue; if (dai->driver->ops && dai->driver->ops->prepare) { ret = dai->driver->ops->prepare(substream, dai); @@ -619,6 +627,9 @@ static int soc_dai_trigger(struct snd_soc_dai *dai, { int ret = 0; + if (!snd_soc_dai_stream_valid(dai, substream->stream)) + return 0; + if (dai->driver->ops && dai->driver->ops->trigger) ret = dai->driver->ops->trigger(substream, cmd, dai); -- 2.34.1

2 1

[PATCH] ASoC: rt1017: Add RT1017 SDCA amplifier driver
by derek.fang＠realtek.com 25 Jul '23

25 Jul '23

From: Derek Fang <derek.fang(a)realtek.com> This is the initial amplifier driver for rt1017 SDCA version. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1017-sdca-sdw.c | 824 +++++++++++++++++++++++++++++ sound/soc/codecs/rt1017-sdca-sdw.h | 188 +++++++ 4 files changed, 1020 insertions(+) create mode 100644 sound/soc/codecs/rt1017-sdca-sdw.c create mode 100644 sound/soc/codecs/rt1017-sdca-sdw.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 88c3dbe47..8aba442d1 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -184,6 +184,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT1015 imply SND_SOC_RT1015P imply SND_SOC_RT1016 + imply SND_SOC_RT1017_SDCA_SDW imply SND_SOC_RT1019 imply SND_SOC_RT1305 imply SND_SOC_RT1308 @@ -1428,6 +1429,11 @@ config SND_SOC_RT1016 tristate depends on I2C +config SND_SOC_RT1017_SDCA_SDW + tristate "Realtek RT1017 SDCA Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + config SND_SOC_RT1019 tristate depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 32dcc6de5..be367839c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -205,6 +205,7 @@ snd-soc-rt1011-objs := rt1011.o snd-soc-rt1015-objs := rt1015.o snd-soc-rt1015p-objs := rt1015p.o snd-soc-rt1016-objs := rt1016.o +snd-soc-rt1017-sdca-objs := rt1017-sdca-sdw.o snd-soc-rt1019-objs := rt1019.o snd-soc-rt1305-objs := rt1305.o snd-soc-rt1308-objs := rt1308.o @@ -583,6 +584,7 @@ obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o obj-$(CONFIG_SND_SOC_RT1015) += snd-soc-rt1015.o obj-$(CONFIG_SND_SOC_RT1015P) += snd-soc-rt1015p.o obj-$(CONFIG_SND_SOC_RT1016) += snd-soc-rt1016.o +obj-$(CONFIG_SND_SOC_RT1017_SDCA_SDW) += snd-soc-rt1017-sdca.o obj-$(CONFIG_SND_SOC_RT1019) += snd-soc-rt1019.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o diff --git a/sound/soc/codecs/rt1017-sdca-sdw.c b/sound/soc/codecs/rt1017-sdca-sdw.c new file mode 100644 index 000000000..7295f44c7 --- /dev/null +++ b/sound/soc/codecs/rt1017-sdca-sdw.c @@ -0,0 +1,824 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt1017-sdca-sdw.c -- rt1017 SDCA ALSA SoC amplifier audio driver +// +// Copyright(c) 2023 Realtek Semiconductor Corp. +// +// +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/pm_runtime.h> +#include <linux/mod_devicetable.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt1017-sdca-sdw.h" + +static bool rt1017_sdca_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f55: + case 0x3206: + case 0xc000: + case 0xc001: + case 0xc022: + case 0xc030: + case 0xc104: + case 0xc10b: + case 0xc10c: + case 0xc110: + case 0xc112: + case 0xc300: + case 0xc301: + case 0xc318: + case 0xc325 ... 0xc328: + case 0xc331: + case 0xc340: + case 0xc350 ... 0xc351: + case 0xc500: + case 0xc502: + case 0xc504: + case 0xc507: + case 0xc509: + case 0xc510: + case 0xc512: + case 0xc518: + case 0xc51b: + case 0xc51d: + case 0xc520: + case 0xc540 ... 0xc542: + case 0xc550 ... 0xc552: + case 0xc600: + case 0xc602: + case 0xc612: + case 0xc622: + case 0xc632: + case 0xc642: + case 0xc651: + case 0xca00: + case 0xca09 ... 0xca0c: + case 0xca0e ... 0xca0f: + case 0xca10 ... 0xca11: + case 0xca16 ... 0xca17: + case 0xcb00: + case 0xcc00: + case 0xcc02: + case 0xd017: + case 0xd01a ... 0xd01c: + case 0xd101: + case 0xd20c: + case 0xd300: + case 0xd370: + case 0xd500: + case 0xd545 ... 0xd548: + case 0xd5a5 ... 0xd5a8: + case 0xd5aa ... 0xd5ad: + case 0xda04 ... 0xda07: + case 0xda09 ... 0xda0a: + case 0xda0c ... 0xda0f: + case 0xda11 ... 0xda14: + case 0xda16 ... 0xda19: + case 0xdab6 ... 0xdabb: + case 0xdb09 ... 0xdb0a: + case 0xdb14: + + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_UDMPU21, + RT1017_SDCA_CTL_UDMPU_CLUSTER, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_FU, + RT1017_SDCA_CTL_FU_MUTE, 0x01): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_XU22, + RT1017_SDCA_CTL_BYPASS, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_SAPU29, + RT1017_SDCA_CTL_PROT_STAT, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_CS21, + RT1017_SDCA_CTL_FS_INDEX, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_PDE23, + RT1017_SDCA_CTL_REQ_POWER_STATE, 0): + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_PDE22, + RT1017_SDCA_CTL_REQ_POWER_STATE, 0): + return true; + default: + return false; + } +} + +static bool rt1017_sdca_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f55: + case 0xc000: + case 0xc022: + case 0xc351: + case 0xc518: + case SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_SAPU29, + RT1017_SDCA_CTL_PROT_STAT, 0): + return true; + default: + return false; + } +} + +static const struct reg_sequence rt1017_blind_write[] = { + { 0xc001, 0x43 }, + { 0x2f55, 0x02 }, + { 0x3206, 0x80 }, + { 0x005f, 0x7f }, + { 0xd101, 0xa0 }, + { 0xc112, 0xc0 }, + { 0xc104, 0xaa }, + { 0xc110, 0x59 }, + { 0xc112, 0xc0 }, + { 0xc340, 0x80 }, + { 0xd017, 0x2c }, + { 0xd01a, 0xc8 }, + { 0xd01b, 0xcf }, + { 0xd01c, 0x0c }, + { 0xd20c, 0x14 }, + { 0xdb09, 0x0f }, + { 0xdb0a, 0x7f }, + { 0xdb14, 0x03 }, + { 0xcb00, 0x31 }, + { 0xc318, 0x44 }, + { 0xc325, 0xce }, + { 0xc326, 0x13 }, + { 0xc327, 0x5f }, + { 0xc328, 0xf3 }, + { 0xc350, 0xe1 }, + { 0xc351, 0x88 }, + { 0xc030, 0x14 }, + { 0xc331, 0xf2 }, + { 0xc551, 0x0f }, + { 0xc552, 0xff }, + { 0xc651, 0xc0 }, + { 0xc550, 0xd0 }, + { 0xc612, 0x00 }, + { 0xc622, 0x00 }, + { 0xc632, 0x00 }, + { 0xc642, 0x00 }, + { 0xc602, 0xf0 }, + { 0xc600, 0xd0 }, + { 0xcc02, 0x78 }, + { 0xcc00, 0x90 }, + { 0xc300, 0x3f }, + { 0xc301, 0x1d }, + { 0xc10b, 0x2e }, + { 0xc10c, 0x36 }, + + { 0xd5a5, 0x00 }, + { 0xd5a6, 0x6a }, + { 0xd5a7, 0xaa }, + { 0xd5a8, 0xaa }, + { 0xd5aa, 0x00 }, + { 0xd5ab, 0x16 }, + { 0xd5ac, 0xdb }, + { 0xd5ad, 0x6d }, + { 0xd545, 0x09 }, + { 0xd546, 0x30 }, + { 0xd547, 0xf0 }, + { 0xd548, 0xf0 }, + { 0xd500, 0x20 }, + { 0xc504, 0x3f }, + { 0xc540, 0x00 }, + { 0xc541, 0x0a }, + { 0xc542, 0x1a }, + { 0xc512, 0x00 }, + { 0xc520, 0x40 }, + { 0xc51b, 0x7f }, + { 0xc51d, 0x0f }, + { 0xc500, 0x40 }, + { 0xc502, 0xde }, + { 0xc507, 0x05 }, + { 0xc509, 0x05 }, + { 0xc510, 0x40 }, + { 0xc518, 0xc0 }, + { 0xc500, 0xc0 }, + + { 0xda0c, 0x00 }, + { 0xda0d, 0x0b }, + { 0xda0e, 0x55 }, + { 0xda0f, 0x55 }, + { 0xda04, 0x00 }, + { 0xda05, 0x51 }, + { 0xda06, 0xeb }, + { 0xda07, 0x85 }, + { 0xca16, 0x0f }, + { 0xca17, 0x00 }, + { 0xda09, 0x5d }, + { 0xda0a, 0xc0 }, + { 0xda11, 0x26 }, + { 0xda12, 0x66 }, + { 0xda13, 0x66 }, + { 0xda14, 0x66 }, + { 0xda16, 0x79 }, + { 0xda17, 0x99 }, + { 0xda18, 0x99 }, + { 0xda19, 0x99 }, + { 0xca09, 0x00 }, + { 0xca0a, 0x07 }, + { 0xca0b, 0x89 }, + { 0xca0c, 0x61 }, + { 0xca0e, 0x00 }, + { 0xca0f, 0x03 }, + { 0xca10, 0xc4 }, + { 0xca11, 0xb0 }, + { 0xdab6, 0x00 }, + { 0xdab7, 0x01 }, + { 0xdab8, 0x00 }, + { 0xdab9, 0x00 }, + { 0xdaba, 0x00 }, + { 0xdabb, 0x00 }, + { 0xd017, 0x0e }, + { 0xca00, 0xcd }, + { 0xc022, 0x84 }, +}; + +#define RT1017_MAX_REG_NUM 0x4108ffff + +static const struct regmap_config rt1017_sdca_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt1017_sdca_readable_register, + .volatile_reg = rt1017_sdca_volatile_register, + .max_register = RT1017_MAX_REG_NUM, + .reg_defaults = rt1017_sdca_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt1017_sdca_reg_defaults), + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt1017_sdca_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists + * port = 1 for AMP playback + * port = 2 for IV capture + */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 64; + + return 0; +} + +static int rt1017_sdca_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt1017_sdca_priv *rt1017 = dev_get_drvdata(dev); + + if (rt1017->hw_init) + return 0; + + if (rt1017->first_hw_init) { + regcache_cache_only(rt1017->regmap, false); + regcache_cache_bypass(rt1017->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + /* sw reset */ + regmap_write(rt1017->regmap, 0xc000, 0x02); + + /* initial settings - blind write */ + regmap_multi_reg_write(rt1017->regmap, rt1017_blind_write, + ARRAY_SIZE(rt1017_blind_write)); + + if (rt1017->first_hw_init) { + regcache_cache_bypass(rt1017->regmap, false); + regcache_mark_dirty(rt1017->regmap); + } else + rt1017->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt1017->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "hw_init complete\n"); + return 0; +} + +static int rt1017_sdca_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt1017_sdca_priv *rt1017 = dev_get_drvdata(&slave->dev); + + if (status == SDW_SLAVE_UNATTACHED) + rt1017->hw_init = false; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt1017->hw_init || status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt1017_sdca_io_init(&slave->dev, slave); +} + +static const char * const rt1017_rx_data_ch_select[] = { + "Bypass", + "CN1", + "CN2", + "CN3", + "CN4", + "(1+2)/2", + "(1+3)/2", + "(1+4)/2", + "(2+3)/2", + "(2+4)/2", + "(3+4)/2", +}; + +static SOC_ENUM_SINGLE_DECL(rt1017_rx_data_ch_enum, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_UDMPU21, + RT1017_SDCA_CTL_UDMPU_CLUSTER, 0), + 0, rt1017_rx_data_ch_select); + +static const struct snd_kcontrol_new rt1017_sdca_controls[] = { + /* UDMPU Cluster Selection */ + SOC_ENUM("RX Channel Select", rt1017_rx_data_ch_enum), +}; + +static const struct snd_kcontrol_new rt1017_sto_dac = + SOC_DAPM_SINGLE("Switch", + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_FU, RT1017_SDCA_CTL_FU_MUTE, 0x1), + 0, 1, 1); + +static int rt1017_sdca_pde23_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt1017_sdca_priv *rt1017 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt1017->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_PDE23, + RT1017_SDCA_CTL_REQ_POWER_STATE, 0), + ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt1017->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_PDE23, + RT1017_SDCA_CTL_REQ_POWER_STATE, 0), + ps3); + break; + default: + break; + } + return 0; +} + +static int rt1017_sdca_classd_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt1017_sdca_priv *rt1017 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_update_bits(rt1017->regmap, RT1017_PWM_TRIM_1, + RT1017_PWM_FREQ_CTL_SRC_SEL_MASK, RT1017_PWM_FREQ_CTL_SRC_SEL_REG); + regmap_write(rt1017->regmap, RT1017_CLASSD_INT_1, 0x10); + break; + default: + break; + } + + return 0; +} + +static int rt1017_sdca_feedback_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct rt1017_sdca_priv *rt1017 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + regmap_update_bits(rt1017->regmap, 0xd017, 0x1f, 0x08); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(rt1017->regmap, 0xd017, 0x1f, 0x09); + break; + default: + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt1017_sdca_dapm_widgets[] = { + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("DP1RX", "DP1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT_E("DP2TX", "DP2 Capture", 0, SND_SOC_NOPM, 0, 0, + rt1017_sdca_feedback_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* Digital Interface */ + SND_SOC_DAPM_SWITCH("DAC", SND_SOC_NOPM, 0, 0, &rt1017_sto_dac), + + /* Output Lines */ + SND_SOC_DAPM_PGA_E("CLASS D", SND_SOC_NOPM, 0, 0, NULL, 0, + rt1017_sdca_classd_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUTPUT("SPO"), + + SND_SOC_DAPM_SUPPLY("PDE23", SND_SOC_NOPM, 0, 0, + rt1017_sdca_pde23_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_PGA("I Sense", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("V Sense", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SIGGEN("I Gen"), + SND_SOC_DAPM_SIGGEN("V Gen"), +}; + +static const struct snd_soc_dapm_route rt1017_sdca_dapm_routes[] = { + + { "DAC", "Switch", "DP1RX" }, + { "CLASS D", NULL, "DAC" }, + { "CLASS D", NULL, "PDE23" }, + { "SPO", NULL, "CLASS D" }, + + { "I Sense", NULL, "I Gen" }, + { "V Sense", NULL, "V Gen" }, + { "I Sense", NULL, "PDE23" }, + { "V Sense", NULL, "PDE23" }, + { "DP2TX", NULL, "I Sense" }, + { "DP2TX", NULL, "V Sense" }, +}; + +static struct sdw_slave_ops rt1017_sdca_slave_ops = { + .read_prop = rt1017_sdca_read_prop, + .update_status = rt1017_sdca_update_status, +}; + +static int rt1017_sdca_component_probe(struct snd_soc_component *component) +{ + int ret; + + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + + return 0; +} + +static void rt1017_sdca_component_remove(struct snd_soc_component *component) +{ + struct rt1017_sdca_priv *rt1017 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt1017->regmap, true); +} + +static const struct snd_soc_component_driver soc_sdca_component_rt1017 = { + .probe = rt1017_sdca_component_probe, + .remove = rt1017_sdca_component_remove, + .controls = rt1017_sdca_controls, + .num_controls = ARRAY_SIZE(rt1017_sdca_controls), + .dapm_widgets = rt1017_sdca_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1017_sdca_dapm_widgets), + .dapm_routes = rt1017_sdca_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1017_sdca_dapm_routes), + .endianness = 1, +}; + +static int rt1017_sdca_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void rt1017_sdca_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int rt1017_sdca_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1017_sdca_priv *rt1017 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_runtime *sdw_stream; + int retval, port, num_channels, ch_mask; + unsigned int sampling_rate; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!rt1017->sdw_slave) + return -EINVAL; + + /* SoundWire specific configuration */ + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + direction = SDW_DATA_DIR_TX; + port = 2; + } + + num_channels = params_channels(params); + ch_mask = (1 << num_channels) - 1; + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = num_channels; + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + port_config.ch_mask = ch_mask; + port_config.num = port; + + dev_dbg(dai->dev, "frame_rate %d, ch_count %d, bps %d, direction %d, ch_mask %d, port: %d\n", + params_rate(params), num_channels, snd_pcm_format_width(params_format(params)), + direction, ch_mask, port); + + retval = sdw_stream_add_slave(rt1017->sdw_slave, &stream_config, + &port_config, 1, sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 44100: + sampling_rate = RT1017_SDCA_RATE_44100HZ; + break; + case 48000: + sampling_rate = RT1017_SDCA_RATE_48000HZ; + break; + case 96000: + sampling_rate = RT1017_SDCA_RATE_96000HZ; + break; + case 192000: + sampling_rate = RT1017_SDCA_RATE_192000HZ; + break; + default: + dev_err(component->dev, "Rate %d is not supported\n", + params_rate(params)); + return -EINVAL; + } + + /* set sampling frequency */ + regmap_write(rt1017->regmap, + SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_CS21, + RT1017_SDCA_CTL_FS_INDEX, 0), + sampling_rate); + + return 0; +} + +static int rt1017_sdca_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1017_sdca_priv *rt1017 = snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt1017->sdw_slave) + return -EINVAL; + + sdw_stream_remove_slave(rt1017->sdw_slave, sdw_stream); + return 0; +} + +static const struct snd_soc_dai_ops rt1017_sdca_ops = { + .hw_params = rt1017_sdca_pcm_hw_params, + .hw_free = rt1017_sdca_pcm_hw_free, + .set_stream = rt1017_sdca_set_sdw_stream, + .shutdown = rt1017_sdca_shutdown, +}; + +#define RT1017_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) +#define RT1017_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver rt1017_sdca_dai[] = { + { + .name = "rt1017-aif", + .playback = { + .stream_name = "DP1 Playback", + .channels_min = 1, + .channels_max = 1, + .rates = RT1017_STEREO_RATES, + .formats = RT1017_FORMATS, + }, + .capture = { + .stream_name = "DP2 Capture", + .channels_min = 1, + .channels_max = 1, + .rates = RT1017_STEREO_RATES, + .formats = RT1017_FORMATS, + }, + .ops = &rt1017_sdca_ops, + }, +}; + +static int rt1017_sdca_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave) +{ + struct rt1017_sdca_priv *rt1017; + int ret; + + rt1017 = devm_kzalloc(dev, sizeof(*rt1017), GFP_KERNEL); + if (!rt1017) + return -ENOMEM; + + dev_set_drvdata(dev, rt1017); + rt1017->sdw_slave = slave; + rt1017->regmap = regmap; + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt1017->hw_init = false; + rt1017->first_hw_init = false; + + ret = devm_snd_soc_register_component(dev, + &soc_sdca_component_rt1017, + rt1017_sdca_dai, + ARRAY_SIZE(rt1017_sdca_dai)); + + return ret; +} + +static int rt1017_sdca_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + + /* Regmap Initialization */ + regmap = devm_regmap_init_sdw(slave, &rt1017_sdca_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return rt1017_sdca_init(&slave->dev, regmap, slave); +} + +static int rt1017_sdca_sdw_remove(struct sdw_slave *slave) +{ + struct rt1017_sdca_priv *rt1017 = dev_get_drvdata(&slave->dev); + + if (rt1017->first_hw_init) + pm_runtime_disable(&slave->dev); + + return 0; +} + +static const struct sdw_device_id rt1017_sdca_id[] = { + SDW_SLAVE_ENTRY_EXT(0x025d, 0x1017, 0x3, 0x1, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt1017_sdca_id); + +static int __maybe_unused rt1017_sdca_dev_suspend(struct device *dev) +{ + struct rt1017_sdca_priv *rt1017 = dev_get_drvdata(dev); + + if (!rt1017->hw_init) + return 0; + + regcache_cache_only(rt1017->regmap, true); + + return 0; +} + +#define RT1017_PROBE_TIMEOUT 5000 + +static int __maybe_unused rt1017_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt1017_sdca_priv *rt1017 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt1017->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT1017_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt1017->regmap, false); + regcache_sync(rt1017->regmap); + + return 0; +} + +static const struct dev_pm_ops rt1017_sdca_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt1017_sdca_dev_suspend, rt1017_sdca_dev_resume) + SET_RUNTIME_PM_OPS(rt1017_sdca_dev_suspend, rt1017_sdca_dev_resume, NULL) +}; + +static struct sdw_driver rt1017_sdca_sdw_driver = { + .driver = { + .name = "rt1017-sdca", + .owner = THIS_MODULE, + .pm = &rt1017_sdca_pm, + }, + .probe = rt1017_sdca_sdw_probe, + .remove = rt1017_sdca_sdw_remove, + .ops = &rt1017_sdca_slave_ops, + .id_table = rt1017_sdca_id, +}; +module_sdw_driver(rt1017_sdca_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT1017 driver SDCA SDW"); +MODULE_AUTHOR("Derek Fang <derek.fang(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt1017-sdca-sdw.h b/sound/soc/codecs/rt1017-sdca-sdw.h new file mode 100644 index 000000000..5968b9a7d --- /dev/null +++ b/sound/soc/codecs/rt1017-sdca-sdw.h @@ -0,0 +1,188 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt1017-sdca-sdw.h -- RT1017 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2023 Realtek Semiconductor Corp. + */ + +#ifndef __RT1017_SDW_H__ +#define __RT1017_SDW_H__ + +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <linux/soundwire/sdw_registers.h> +#include <sound/soc.h> + +/* RT1017 SDCA Control - function number */ +#define FUNC_NUM_SMART_AMP 0x04 + +/* RT1017 SDCA entity */ +#define RT1017_SDCA_ENT_PDE23 0x31 +#define RT1017_SDCA_ENT_PDE22 0x33 +#define RT1017_SDCA_ENT_CS21 0x21 +#define RT1017_SDCA_ENT_SAPU29 0x29 +#define RT1017_SDCA_ENT_XU22 0x22 +#define RT1017_SDCA_ENT_FU 0x03 +#define RT1017_SDCA_ENT_UDMPU21 0x02 + +/* RT1017 SDCA control */ +#define RT1017_SDCA_CTL_FS_INDEX 0x10 +#define RT1017_SDCA_CTL_REQ_POWER_STATE 0x01 +#define RT1017_SDCA_CTL_PROT_STAT 0x11 +#define RT1017_SDCA_CTL_BYPASS 0x01 +#define RT1017_SDCA_CTL_FU_MUTE 0x01 +#define RT1017_SDCA_CTL_FU_VOLUME 0x02 +#define RT1017_SDCA_CTL_UDMPU_CLUSTER 0x10 + + +#define RT1017_CLASSD_INT_1 0xd300 +#define RT1017_PWM_TRIM_1 0xd370 + + +#define RT1017_PWM_FREQ_CTL_SRC_SEL_MASK (0x3 << 2) +#define RT1017_PWM_FREQ_CTL_SRC_SEL_EFUSE (0x2 << 2) +#define RT1017_PWM_FREQ_CTL_SRC_SEL_REG (0x0 << 2) + +enum { + RT1017_SDCA_RATE_44100HZ = 0x8, + RT1017_SDCA_RATE_48000HZ = 0x9, + RT1017_SDCA_RATE_96000HZ = 0xb, + RT1017_SDCA_RATE_192000HZ = 0xd, +}; + +struct rt1017_sdca_priv { + struct snd_soc_component *component; + struct regmap *regmap; + struct sdw_slave *sdw_slave; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; +}; + +static const struct reg_default rt1017_sdca_reg_defaults[] = { + { 0x3206, 0x00 }, + { 0xc001, 0x43 }, + { 0xc030, 0x54 }, + { 0xc104, 0x8a }, + { 0xc10b, 0x2f }, + { 0xc10c, 0x2f }, + { 0xc110, 0x49 }, + { 0xc112, 0x10 }, + { 0xc300, 0xff }, + { 0xc301, 0xdd }, + { 0xc318, 0x40 }, + { 0xc325, 0x00 }, + { 0xc326, 0x00 }, + { 0xc327, 0x00 }, + { 0xc328, 0x02 }, + { 0xc331, 0xb2 }, + { 0xc340, 0x02 }, + { 0xc350, 0x21 }, + { 0xc500, 0x00 }, + { 0xc502, 0x00 }, + { 0xc504, 0x3f }, + { 0xc507, 0x1f }, + { 0xc509, 0x1f }, + { 0xc510, 0x40 }, + { 0xc512, 0x00 }, + { 0xc518, 0x02 }, + { 0xc51b, 0x7f }, + { 0xc51d, 0x0f }, + { 0xc520, 0x00 }, + { 0xc540, 0x80 }, + { 0xc541, 0x00 }, + { 0xc542, 0x0a }, + { 0xc550, 0x80 }, + { 0xc551, 0x0f }, + { 0xc552, 0xff }, + { 0xc600, 0x10 }, + { 0xc602, 0x83 }, + { 0xc612, 0x40 }, + { 0xc622, 0x40 }, + { 0xc632, 0x40 }, + { 0xc642, 0x40 }, + { 0xc651, 0x00 }, + { 0xca00, 0xc1 }, + { 0xca09, 0x00 }, + { 0xca0a, 0x51 }, + { 0xca0b, 0xeb }, + { 0xca0c, 0x85 }, + { 0xca0e, 0x00 }, + { 0xca0f, 0x10 }, + { 0xca10, 0x62 }, + { 0xca11, 0x4d }, + { 0xca16, 0x0f }, + { 0xca17, 0x00 }, + { 0xcb00, 0x10 }, + { 0xcc00, 0x10 }, + { 0xcc02, 0x0b }, + { 0xd017, 0x09 }, + { 0xd01a, 0x00 }, + { 0xd01b, 0x00 }, + { 0xd01c, 0x00 }, + { 0xd101, 0xa0 }, + { 0xd20c, 0x14 }, + { 0xd300, 0x0f }, + { 0xd370, 0x18 }, + { 0xd500, 0x00 }, + { 0xd545, 0x0b }, + { 0xd546, 0xf9 }, + { 0xd547, 0xb2 }, + { 0xd548, 0xa9 }, + { 0xd5a5, 0x00 }, + { 0xd5a6, 0x00 }, + { 0xd5a7, 0x00 }, + { 0xd5a8, 0x00 }, + { 0xd5aa, 0x00 }, + { 0xd5ab, 0x00 }, + { 0xd5ac, 0x00 }, + { 0xd5ad, 0x00 }, + { 0xda04, 0x03 }, + { 0xda05, 0x33 }, + { 0xda06, 0x33 }, + { 0xda07, 0x33 }, + { 0xda09, 0x5d }, + { 0xda0a, 0xc0 }, + { 0xda0c, 0x00 }, + { 0xda0d, 0x01 }, + { 0xda0e, 0x5d }, + { 0xda0f, 0x86 }, + { 0xda11, 0x20 }, + { 0xda12, 0x00 }, + { 0xda13, 0x00 }, + { 0xda14, 0x00 }, + { 0xda16, 0x7f }, + { 0xda17, 0xff }, + { 0xda18, 0xff }, + { 0xda19, 0xff }, + { 0xdab6, 0x00 }, + { 0xdab7, 0x01 }, + { 0xdab8, 0x00 }, + { 0xdab9, 0x01 }, + { 0xdaba, 0x00 }, + { 0xdabb, 0x01 }, + { 0xdb09, 0x0f }, + { 0xdb0a, 0xff }, + { 0xdb14, 0x00 }, + + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_UDMPU21, + RT1017_SDCA_CTL_UDMPU_CLUSTER, 0), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_FU, + RT1017_SDCA_CTL_FU_MUTE, 0x01), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_XU22, + RT1017_SDCA_CTL_BYPASS, 0), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_CS21, + RT1017_SDCA_CTL_FS_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_PDE23, + RT1017_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_PDE22, + RT1017_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, +}; + +static const struct reg_default rt1017_sdca_mbq_defaults[] = { + { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_FU, + RT1017_SDCA_CTL_FU_VOLUME, 0x01), 0x00 }, +}; + +#endif /* __RT1017_SDW_H__ */ -- 2.34.1

2 1

[PATCH v2 1/5] ASoC: codecs: ES8326: Change Hp_detect register names
by Zhu Ning 25 Jul '23

25 Jul '23

The old register naming method is confusing. The reg 0x57 decides the default headset hardware connection type, and the reg 0xfb is the headset detection status register, which changes during headset insertion. Change the name to ES8326_HPDET_TYPE and ES8326_HPDET_STA. Signed-off-by: Zhu Ning <zhuning0077(a)gmail.com> --- sound/soc/codecs/es8326.c | 10 +++++----- sound/soc/codecs/es8326.h | 13 ++++++++----- 2 files changed, 13 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index 4399ba36aee2..e953c0157ba0 100644 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -159,7 +159,7 @@ static const struct snd_soc_dapm_route es8326_dapm_routes[] = { }; static const struct regmap_range es8326_volatile_ranges[] = { - regmap_reg_range(ES8326_HP_DETECT, ES8326_HP_DETECT), + regmap_reg_range(ES8326_HPDET_STA, ES8326_HPDET_STA), }; static const struct regmap_access_table es8326_volatile_table = { @@ -519,7 +519,7 @@ static void es8326_jack_button_handler(struct work_struct *work) return; mutex_lock(&es8326->lock); - iface = snd_soc_component_read(comp, ES8326_HP_DETECT); + iface = snd_soc_component_read(comp, ES8326_HPDET_STA); switch (iface) { case 0x93: /* pause button detected */ @@ -578,7 +578,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) unsigned int iface; mutex_lock(&es8326->lock); - iface = snd_soc_component_read(comp, ES8326_HP_DETECT); + iface = snd_soc_component_read(comp, ES8326_HPDET_STA); dev_dbg(comp->dev, "gpio flag %#04x", iface); if ((iface & ES8326_HPINSERT_FLAG) == 0) { /* Jack unplugged or spurious IRQ */ @@ -651,7 +651,7 @@ static int es8326_resume(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC1_SRC, es8326->mic1_src); regmap_write(es8326->regmap, ES8326_ADC2_SRC, es8326->mic2_src); regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0x88); - regmap_write(es8326->regmap, ES8326_HP_DET, + regmap_write(es8326->regmap, ES8326_HPDET_TYPE, ES8326_HP_DET_SRC_PIN9 | es8326->jack_pol); regmap_write(es8326->regmap, ES8326_INT_SOURCE, es8326->interrupt_src); regmap_write(es8326->regmap, ES8326_INTOUT_IO, es8326->interrupt_clk); @@ -743,7 +743,7 @@ static void es8326_enable_jack_detect(struct snd_soc_component *component, mutex_lock(&es8326->lock); if (es8326->jd_inverted) - snd_soc_component_update_bits(component, ES8326_HP_DET, + snd_soc_component_update_bits(component, ES8326_HPDET_TYPE, ES8326_HP_DET_JACK_POL, ~es8326->jack_pol); es8326->jack = jack; diff --git a/sound/soc/codecs/es8326.h b/sound/soc/codecs/es8326.h index 3f8f7da58062..cd04d11a88d9 100644 --- a/sound/soc/codecs/es8326.h +++ b/sound/soc/codecs/es8326.h @@ -73,15 +73,19 @@ #define ES8326_DRC_RECOVERY 0x53 #define ES8326_DRC_WINSIZE 0x54 #define ES8326_HPJACK_TIMER 0x56 -#define ES8326_HP_DET 0x57 +#define ES8326_HPDET_TYPE 0x57 #define ES8326_INT_SOURCE 0x58 #define ES8326_INTOUT_IO 0x59 #define ES8326_SDINOUT1_IO 0x5A #define ES8326_SDINOUT23_IO 0x5B #define ES8326_JACK_PULSE 0x5C +#define ES8326_HP_MISC 0xF7 +#define ES8326_CTIA_OMTP_STA 0xF8 #define ES8326_PULLUP_CTL 0xF9 -#define ES8326_HP_DETECT 0xFB +#define ES8326_CSM_I2C_STA 0xFA +#define ES8326_HPDET_STA 0xFB +#define ES8326_CSM_MUTE_STA 0xFC #define ES8326_CHIP_ID1 0xFD #define ES8326_CHIP_ID2 0xFE #define ES8326_CHIP_VERSION 0xFF @@ -146,7 +150,7 @@ #define ES8326_ADC3_SHIFT 0 #define ES8326_ADC4_SHIFT 3 -/* ES8326_HP_DET */ +/* ES8326_HPDET_TYPE */ #define ES8326_HP_DET_SRC_PIN27 (1 << 5) #define ES8326_HP_DET_SRC_PIN9 (1 << 4) #define ES8326_HP_DET_JACK_POL (1 << 3) @@ -174,7 +178,7 @@ #define ES8326_SDINOUT2_SHIFT 4 #define ES8326_SDINOUT3_SHIFT 0 -/* ES8326_HP_DETECT */ +/* ES8326_HPDET_STA */ #define ES8326_HPINSERT_FLAG (1 << 1) #define ES8326_HPBUTTON_FLAG (1 << 0) @@ -182,4 +186,3 @@ #define ES8326_VERSION_B (1 << 0) #endif - -- 2.34.1

2 5

[PATCH V2 0/4] ASoC: codecs: Add Awinic AW88261 audio amplifier driver
by wangweidong.a＠awinic.com 25 Jul '23

25 Jul '23

From: Weidong Wang <wangweidong.a(a)awinic.com> The awinic AW88261 is an I2S/TDM input, high efficiency digital Smart K audio amplifier v1 -> v2: Use dev_err_prober instead of dev_err Delete unwanted dev_dbg Delect print messages on allocation errors. The commit information has been changed Delete EXPORT_SYMBOL_GPL Modify {} usage errors The aw88395_lib file is compatible with the bin parsing part of aw88261 Weidong Wang (4): ASoC: codecs: Add code for bin parsing compatible with aw88261 ASoC: codecs: Add aw88261 audio amplifier driver ASoC: codecs: aw88261 device related operation functions ASoC: codecs: aw88261 chip register file, Kconfig and Makefile .../bindings/sound/awinic,aw88395.yaml | 4 +- sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 3 + sound/soc/codecs/aw88261/aw88261.c | 517 +++++++++++ sound/soc/codecs/aw88261/aw88261.h | 52 ++ sound/soc/codecs/aw88261/aw88261_device.c | 876 ++++++++++++++++++ sound/soc/codecs/aw88261/aw88261_device.h | 79 ++ sound/soc/codecs/aw88261/aw88261_reg.h | 377 ++++++++ sound/soc/codecs/aw88395/aw88395_lib.c | 194 +++- sound/soc/codecs/aw88395/aw88395_reg.h | 1 + 10 files changed, 2098 insertions(+), 18 deletions(-) create mode 100644 sound/soc/codecs/aw88261/aw88261.c create mode 100644 sound/soc/codecs/aw88261/aw88261.h create mode 100644 sound/soc/codecs/aw88261/aw88261_device.c create mode 100644 sound/soc/codecs/aw88261/aw88261_device.h create mode 100644 sound/soc/codecs/aw88261/aw88261_reg.h base-commit: 0b5547c51827e053cc754db47d3ec3e6c2c451d2 -- 2.41.0

3 6

[PATCH 00/11] Even more msm bindings fixes
by Konrad Dybcio 25 Jul '23

25 Jul '23

This series contains couple more random dt schema warning fixes, centered around linux-msm. All of the patches (except) the last two are pretty much independent. Patch 1 is not only a bindings fix, but also a functional one. Signed-off-by: Konrad Dybcio <konrad.dybcio(a)linaro.org> --- Konrad Dybcio (11): arm64: dts: qcom: sc7180: Fix DSI0_PHY reg-names arm64: dts: qcom: msm8939: Drop "qcom,idle-state-spc" compatible arm64: dts: qcom: msm8939: Add missing 'cache-unified' to L2 arm64: dts: qcom: apq8039-t2: Drop inexistent property dt-bindings: sound: Convert pm8916-wcd-analog-codec to YAML dt-bindings: mfd: qcom,spmi-pmic: Reference pm8916 wcd analog codec schema dt-bindings: iommu: qcom_iommu: Allow 'tbu' clock dt-bindings: usb: ci-hdrc-usb2: Fix handling pinctrl properties dt-bindings: arm: msm: kpss-acc: Make the optional reg truly optional dt-bindings: usb: dwc3: Allow just 1 interrupt for MSM8996 arm64: dts: qcom: msm8996: Add missing interrupt to the USB2 controller .../devicetree/bindings/arm/msm/qcom,kpss-acc.yaml | 1 + .../devicetree/bindings/iommu/qcom,iommu.yaml | 4 + .../devicetree/bindings/mfd/qcom,spmi-pmic.yaml | 2 +- .../bindings/sound/qcom,msm8916-wcd-analog.txt | 101 ------------- .../sound/qcom,pm8916-wcd-analog-codec.yaml | 160 +++++++++++++++++++++ .../devicetree/bindings/usb/ci-hdrc-usb2.yaml | 27 +--- .../devicetree/bindings/usb/qcom,dwc3.yaml | 2 +- arch/arm64/boot/dts/qcom/apq8039-t2.dts | 1 - arch/arm64/boot/dts/qcom/msm8939.dtsi | 4 +- arch/arm64/boot/dts/qcom/msm8996.dtsi | 3 + arch/arm64/boot/dts/qcom/sc7180.dtsi | 4 +- 11 files changed, 182 insertions(+), 127 deletions(-) --- base-commit: 3eedd211ad93c322fb360b83a3d76a2c6cd622dc change-id: 20230627-topic-more_bindings-325a6f4b93a7 Best regards, -- Konrad Dybcio <konrad.dybcio(a)linaro.org>

9 29

[PATCH 00/10] Introduce STM32 Firewall framework
by Gatien Chevallier 25 Jul '23

25 Jul '23

Introduce STM32 Firewall framework for STM32MP1x and STM32MP2x platforms. STM32MP1x(ETZPC) and STM32MP2x(RIFSC) Firewall controllers register to the framework to offer firewall services such as access granting. This series of patches is a new approach on the previous STM32 system bus, history is available here: https://lore.kernel.org/lkml/20230127164040.1047583/ The need for such framework arises from the fact that there are now multiple hardware firewalls implemented across multiple products. Drivers are shared between different products, using the same code. When it comes to firewalls, the purpose mostly stays the same: Protect hardware resources. But the implementation differs, and there are multiple types of firewalls: peripheral, memory, ... Some hardware firewall controllers such as the RIFSC implemented on STM32MP2x platforms may require to take ownership of a resource before being able to use it, hence the requirement for firewall services to take/release the ownership of such resources. On the other hand, hardware firewall configurations are becoming more and more complex. These mecanisms prevent platform crashes or other firewall-related incoveniences by denying access to some resources. The stm32 firewall framework offers an API that is defined in firewall controllers drivers to best fit the specificity of each firewall. For every peripherals protected by either the ETZPC or the RIFSC, the firewall framework checks the firewall controlelr registers to see if the peripheral's access is granted to the Linux kernel. If not, the peripheral is configured as secure, the node is marked populated, so that the driver is not probed for that device. The firewall framework relies on the feature-domain-controller device tree bindings: https://lore.kernel.org/lkml/0c0a82bb-18ae-d057-562b. It is used by peripherals to reference a domain controller, in this case a firewall feature domain. The bus uses the ID referenced by the feature-domains property to know where to look in the firewall to get the security configuration for the peripheral. This allows a device tree description rather than a hardcoded peripheral table in the bus driver. The STM32 ETZPC device is responsible for filtering accesses based on security level, or co-processor isolation for any resource connected to it. The RIFSC is responsible for filtering accesses based on Compartment ID / security level / privilege level for any resource connected to it. STM32MP13/15/25 SoC device tree files are updated in this series to implement this mecanism. Oleksii Moisieiev (1): dt-bindings: Document common device controller bindings Gatien Chevallier (9): dt-bindings: bus: add device tree bindings for RIFSC dt-bindings: bus: add device tree bindings for ETZPC dt-bindings: treewide: add feature-domains description in binding files firewall: introduce stm32_firewall framework bus: rifsc: introduce RIFSC firewall controller driver arm64: dts: st: add RIFSC as a domain controller for STM32MP25x boards bus: etzpc: introduce ETZPC firewall controller driver ARM: dts: stm32: add ETZPC as a system bus for STM32MP15x boards ARM: dts: stm32: add ETZPC as a system bus for STM32MP13x boards .../bindings/bus/st,stm32-etzpc.yaml | 90 + .../bindings/bus/st,stm32-rifsc.yaml | 101 + .../bindings/crypto/st,stm32-hash.yaml | 4 + .../devicetree/bindings/dma/st,stm32-dma.yaml | 4 + .../bindings/dma/st,stm32-dmamux.yaml | 4 + .../feature-domain-controller.yaml | 84 + .../devicetree/bindings/i2c/st,stm32-i2c.yaml | 4 + .../bindings/iio/adc/st,stm32-adc.yaml | 4 + .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 4 + .../bindings/iio/dac/st,stm32-dac.yaml | 4 + .../bindings/media/cec/st,stm32-cec.yaml | 4 + .../bindings/media/st,stm32-dcmi.yaml | 4 + .../memory-controllers/st,stm32-fmc2-ebi.yaml | 4 + .../bindings/mfd/st,stm32-lptimer.yaml | 4 + .../bindings/mfd/st,stm32-timers.yaml | 5 + .../devicetree/bindings/mmc/arm,pl18x.yaml | 4 + .../devicetree/bindings/net/stm32-dwmac.yaml | 4 + .../bindings/phy/phy-stm32-usbphyc.yaml | 4 + .../bindings/regulator/st,stm32-vrefbuf.yaml | 4 + .../devicetree/bindings/rng/st,stm32-rng.yaml | 4 + .../bindings/serial/st,stm32-uart.yaml | 4 + .../bindings/sound/st,stm32-i2s.yaml | 4 + .../bindings/sound/st,stm32-sai.yaml | 4 + .../bindings/sound/st,stm32-spdifrx.yaml | 4 + .../bindings/spi/st,stm32-qspi.yaml | 4 + .../devicetree/bindings/spi/st,stm32-spi.yaml | 4 + .../devicetree/bindings/usb/dwc2.yaml | 4 + MAINTAINERS | 7 + arch/arm/boot/dts/st/stm32mp131.dtsi | 1027 +++--- arch/arm/boot/dts/st/stm32mp133.dtsi | 51 +- arch/arm/boot/dts/st/stm32mp13xc.dtsi | 19 +- arch/arm/boot/dts/st/stm32mp13xf.dtsi | 19 +- arch/arm/boot/dts/st/stm32mp151.dtsi | 2757 +++++++++-------- arch/arm/boot/dts/st/stm32mp153.dtsi | 52 +- arch/arm/boot/dts/st/stm32mp15xc.dtsi | 19 +- arch/arm64/Kconfig.platforms | 1 + arch/arm64/boot/dts/st/stm32mp251.dtsi | 5 +- drivers/bus/Kconfig | 10 + drivers/bus/Makefile | 1 + drivers/bus/stm32_etzpc.c | 137 + drivers/bus/stm32_firewall.c | 252 ++ drivers/bus/stm32_firewall.h | 83 + drivers/bus/stm32_rifsc.c | 248 ++ include/linux/bus/stm32_firewall_device.h | 134 + 44 files changed, 3276 insertions(+), 1918 deletions(-) create mode 100644 Documentation/devicetree/bindings/bus/st,stm32-etzpc.yaml create mode 100644 Documentation/devicetree/bindings/bus/st,stm32-rifsc.yaml create mode 100644 Documentation/devicetree/bindings/feature-controllers/feature-domain-controller.yaml create mode 100644 drivers/bus/stm32_etzpc.c create mode 100644 drivers/bus/stm32_firewall.c create mode 100644 drivers/bus/stm32_firewall.h create mode 100644 drivers/bus/stm32_rifsc.c create mode 100644 include/linux/bus/stm32_firewall_device.h -- 2.25.1

6 42

[GIT PULL] ASoC updates for v6.6-early
by Mark Brown 25 Jul '23

25 Jul '23

The following changes since commit 6eaae198076080886b9e7d57f4ae06fa782f90ef: Linux 6.5-rc3 (2023-07-23 15:24:10 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.6-early for you to fetch changes up to 85d12eda2382cd5b93eed720b5a08f39d42cfef4: ALSA: hda: Adding support for CS35L56 on HDA (2023-07-24 18:42:26 +0100) ---------------------------------------------------------------- ASoC: Updates for v6.6 Here's an initial batch of updates for ASoC for this release cycle. We've got a bunch of new drivers in here, a bit of core work from Morimoto-san and quite a lot of janitorial work. There's several updates that pull in changes from other subsystems in order to build on them: - An adaptor to allow use of IIO DACs and ADCs in ASoC which pulls in some IIO changes. - Create a library function for intlog10() and use it in the NAU8825 driver. - Include the ASoC tests, including the topology tests, in the default KUnit full test coverage. This also involves enabling UML builds of ALSA since that's the default KUnit test environment which pulls in the addition of some stubs to the driver. - More factoring out from Morimoto-san. - Convert a lot of drivers to use the more modern maple tree register cache. - Support for AMD machines with MAX98388 and NAU8821, Cirrus Logic CS35L36, Intel AVS machines with ES8336 and RT5663 and NXP i.MX93. ---------------------------------------------------------------- Amadeusz Sławiński (2): ASoC: Intel: avs: Add rt5663 machine board ASoC: Intel: avs: Load rt5663 board on KBL-based platforms Andreas Kemnade (1): ASoC: ti: omap-mcbsp: Ignore errors for getting fck_src Andy Shevchenko (8): ASoC: rt5677: Refactor GPIO support code ASoC: rt5677: Use agnostic irq_domain_create_linear() ASoC: rt5677: Use device_get_match_data() ASoC: rt5677: Sort headers alphabetically lib/math: Move dvb_math.c into lib/math/int_log.c lib/math/int_log: Use ARRAY_SIZE(logtable) where makes sense lib/math/int_log: Replace LGPL-2.1-or-later boilerplate with SPDX identifier ASoC: nau8825: Replace copied'n'pasted intlog10() Arnd Bergmann (1): ASoC: amd: acp: fix SND_SOC_AMD_ACP_PCI depdenencies Bjorn Andersson (1): ASoC: codecs: lpass: Log clk_get() failures Brent Lu (1): ASoC: Intel: sof_rt5682: add jsl_rt5650 board config Cezary Rojewski (6): ASoC: codecs: es8316: Add support for 24 MHz MCLK ASoC: codecs: es8316: Add support for S24_3LE format ASoC: Intel: avs: Add es8336 machine board ASoC: Intel: avs: Load es8336 board on KBL-based platforms ASoC: Intel: avs: rt5682: Add missing components ASoC: Intel: avs: rt5682: Tidy up hw_params() Chancel Liu (3): ASoC: imx-pcm-rpmsg: Set PCM hardware parameters separately ASoC: dt-bindings: fsl_rpmsg: Add compatible string for i.MX93 ASoC: fsl_rpmsg: Add support for i.MX93 platform Cristian Ciocaltea (3): ASoC: amd: vangogh: Make use of DRV_NAME ASoC: amd: vangogh: Use dmi_first_match() for DMI quirk handling ASoC: amd: vangogh: Add support for NAU8821/MAX98388 variant Dan Carpenter (1): ASoC: amd: acp: delete unnecessary NULL check Francesco Dolcini (3): ASoC: dt-bindings: nau8822: Add #sound-dai-cells ASoC: dt-bindings: nau8822: Add MCLK clock ASoC: dt-bindings: wm8904: Convert to dtschema Frank Li (1): ASoC: codec: wm8960: add additional probe check for codec identification Herve Codina (13): ASoC: dt-bindings: Add audio-iio-aux ASoC: dt-bindings: simple-card: Add additional-devs subnode iio: inkern: Check error explicitly in iio_channel_read_max() iio: consumer.h: Fix raw values documentation notes iio: inkern: Remove the 'unused' variable usage in iio_channel_read_max() iio: inkern: Fix headers inclusion order minmax: Introduce {min,max}_array() iio: inkern: Use max_array() to get the maximum value from an array iio: inkern: Replace a FIXME comment by a TODO one iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Convert macros to return a compound literal ASoC: codecs: Add support for the generic IIO auxiliary devices ASoC: simple-card: Handle additional devices Jack Yu (1): ASoC: rt722-sdca: Remove redundant sdca mask Konrad Dybcio (1): ASoC: Convert pm8916-wcd-analog-codec to YAML Kuninori Morimoto (10): ASoC: soc-core.c: initialize dlc on snd_soc_get_dai_id() ASoC: soc-core.c: cleanup soc_dai_link_sanity_check() ASoC: soc-core: protect dlc->of_node under mutex ASoC: soc-dai.c: add DAI get/match functions ASoC: soc-core.c: enable multi Component ASoC: soc-core.c: add snd_soc_get_dai_via_args() ASoC: soc-core.c: add snd_soc_dlc_use_cpu_as_platform() ASoC: soc-core.c: add snd_soc_copy_dai_args() ASoC: simple-card-utils.c: enable multi Component support ASoC: simple-card.c: enable multi Component support Mark Brown (88): Merge existing fixes from asoc/for-6.5 into new branch Add support for IIO devices in ASoC ASoC: remove copy of intlog10() ASoC: Intel: avs: New boards and fixes to existing AMD Vangogh support for NAU8821/MAX98388 ASoC: mt8188: add memory-region support Add acpi mahine id's for vangogh platform and Refactor acp legacy driver and add ASoC: rt5677: Refactor GPIO and use ASoC: Another set of platform remove conversions ASoC: ad1836: Update to use maple tree register cache ASoC: ad1980: Update to use maple tree register cache ASoC: adau1372: Update to use maple tree register cache ASoC: adau1373: Update to use maple tree register cache ASoC: adau1701: Update to use maple tree register cache ASoC: adau1761: Update to use maple tree register cache ASoC: adau1781: Update to use maple tree register cache ASoC: adau1977: Update to use maple tree register cache ASoC: adau7118: Update to use maple tree register cache ASoC: adav80x: Update to use maple tree register cache ASoC: ad: Update Analog Devices drivers to maple tree ASoC: wcd9335: Update to use maple tree register cache ASoC: wcd938x: Update to use maple tree register cache ASoC: wsa881x: Update to use maple tree register cache ASoC: wsa883x: Update to use maple tree register cache ASoC: qcom: Use the maple tree register cache Add support for rpmsg sound card on i.MX93 platform Add Probe functionality support for amd platforms. ASoC: cs35l36: Update to use maple tree register cache ASoC: cs35l41: Update to use maple tree register cache ASoC: cs35l45: Update to use maple tree register cache ASoC: cs35l56: Update to use maple tree register cache ASoC: cs42l51: Update to use maple tree register cache ASoC: cs4265: Update to use maple tree register cache ASoC: cs4270: Update to use maple tree register cache ASoC: cs42l52: Update to use maple tree register cache ASoC: cs42l56: Update to use maple tree register cache ASoC: cs42xx8: Update to use maple tree register cache ASoC: cs4349: Update to use maple tree register cache ASoC: wm2200: Update to use maple tree register cache ASoC: wm5100: Update to use maple tree register cache ASoC: wm9081: Update to use maple tree register cache ASoC: wm9090: Update to use maple tree register cache ASoC: wm8510: Update to use maple tree register cache ASoC: wm8523: Update to use maple tree register cache ASoC: wm8580: Update to use maple tree register cache ASoC: wm8711: Update to use maple tree register cache ASoC: wm8728: Update to use maple tree register cache ASoC: wm8731: Update to use maple tree register cache ASoC: wm8737: Update to use maple tree register cache ASoC: wm8741: Update to use maple tree register cache ASoC: wm8750: Update to use maple tree register cache ASoC: wm8753: Update to use maple tree register cache ASoC: wm8770: Update to use maple tree register cache ASoC: wm8776: Update to use maple tree register cache ASoC: wm8804: Update to use maple tree register cache ASoC: wm8900: Update to use maple tree register cache ASoC: wm8903: Update to use maple tree register cache ASoC: wm8904: Update to use maple tree register cache ASoC: wm8960: Update to use maple tree register cache ASoC: wm8961: Update to use maple tree register cache ASoC: wm8962: Update to use maple tree register cache ASoC: wm8991: Update to use maple tree register cache ASoC: wm8993: Update to use maple tree register cache ASoC: wm8995: Update to use maple tree register cache ASoC: wm8996: Update to use maple tree register cache ASoC: wm8940: Update to use maple tree register cache ASoC: wm8955: Update to use maple tree register cache ASoC: wm8971: Update to use maple tree register cache ASoC: wm8978: Update to use maple tree register cache ASoC: wm8983: Update to use maple tree register cache ASoC: wm8985: Update to use maple tree register cache ASoC: wm8988: Update to use maple tree register cache ASoC: wm9705: Update to use maple tree register cache ASoC: wm9712: Update to use maple tree register cache ASoC: wm9713: Update to use maple tree register cache driver core: Provide stubs for !IOMEM builds platform: Provide stubs for !HAS_IOMEM builds ALSA: Enable build with UML kunit: Enable ASoC in all_tests.config ASoC: topology: Add explicit build option ASoC: add multi Component support ASoC: cirrus: Use the maple tree register cache ASoC: Improve coverage in default KUnit runs ASoC: dt-bindings: nau8822: minor updates ASoC: codecs: msm8916-wcd-analog: Cleanup DT bindings ASoC: Merge up fixes from mainline ALSA: hda: Adding support for CS35L56 on HDA Mastan Katragadda (1): ASoC: SOF: amd: refactor PSP smn_read Matus Gajdos (1): ASoC: fsl_spdif: Add support for 22.05 kHz sample rate Maxim Kochetkov (1): ASoC: dwc: Add TDM mode support Min-Hua Chen (1): ASoC: q6dsp: q6apm: make g_apm static Randy Dunlap (1): ASoC: stac9766: fix build errors with REGMAP_AC97 Richard Fitzgerald (5): ASoC: cs35l56: Patch soft registers to defaults ASoC: cs35l56: Move runtime suspend/resume to shared library ASoC: cs35l56: Move cs_dsp init into shared library ASoC: cs35l56: Move part of cs35l56_init() to shared library ALSA: hda: Fix missing header dependencies Rob Herring (1): ASoC: dt-bindings: audio-graph-card2: Drop incomplete example Seven Lee (1): ASoC: dt-bindings: nau8821: Convert to dtschema Shengjiu Wang (1): ASoC: wm8960: Add DAC filter characteristics selection Shuming Fan (2): ASoC: rt5645: implement set_jack callback ASoC: rt5645: add the system level suspend-resume callback Simon Trimmer (7): ASoC: cs35l56: Move shared data into a common data structure ASoC: cs35l56: Make cs35l56_system_reset() code more generic ASoC: cs35l56: Convert utility functions to use common data structure ASoC: cs35l56: Move utility functions to shared file ASoC: cs35l56: Make common function for control port wait ASoC: cs35l56: Make a common function to shutdown the DSP ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier Stephan Gerhold (5): ASoC: dt-bindings: pm8916-analog-codec: Fix misleading example ASoC: dt-bindings: pm8916-analog-codec: Drop pointless reg-names ASoC: dt-bindings: pm8916-analog-codec: Drop invalid mclk ASoC: codecs: msm8916-wcd-analog: Drop invalid mclk ASoC: codecs: msm8916-wcd-analog: Properly handle probe errors Syed Saba Kareem (10): ASoC: amd: acp: refactor the acp init and de-init sequence ASoC: amd: acp: add acp i2s master clock generation for rembrandt platform ASoC: amd: acp: remove the redundant acp enable/disable interrupts functions ASoC: amd: acp: store platform device reference created in pci probe call ASoC: amd: acp: add pm ops support for acp pci driver ASoC: amd: acp: store xfer_resolution of the stream ASoC: amd: acp: export config_acp_dma() and config_pte_for_stream() symbols ASoC: amd: acp: store the pdm stream channel mask ASoC: amd: acp: move pdm macros to common header file ASoC: amd: acp: add pm ops support for rembrandt platform Trevor Wu (3): ASoC: mediatek: mt8188: add memory-region support ASoC: dt-bindings: mediatek,mt8188-afe: add memory-region ASoC: mediatek: mt8188: DPCM used FE and BE merged parameters Uwe Kleine-König (2): ASoC: amd: ps-sdw-dma: Convert to platform remove callback returning void ASoC: starfive: jh7110_tdm: Convert to platform remove callback returning void V sujith kumar Reddy (3): ASoC: SOF: amd: Add Probe functionality support for amd platforms. ASoC: SOF: Add acp-probe id to sof probe client driver for registration. ASoC: SOF: amd: Add Probe register offset for renoir and rembrandt platform. Venkata Prasad Potturu (4): ASoC: amd: acp: Add machine driver support for nau8821 codec ASoC: amd: acp: Add machine driver support for max98388 codec ASoC: amd: Add acpi machine id's for vangogh platform ASoC: amd: Add new dmi entries to config entry Yang Li (2): ASoC: tas2781: No need to set device_driver owner ASoC: amd: acp: clean up some inconsistent indentings Yangtao Li (5): ASoC: bcm: bcm63xx-i2s-whistler: Convert to devm_platform_ioremap_resource() ASoC: ti: Convert to devm_platform_ioremap_resource_byname() ASoC: mediatek: mt8186: Convert to devm_platform_ioremap_resource() ASoC: pxa: Use devm_platform_get_and_ioremap_resource() ASoC: tegra: tegra20_ac97: Use devm_platform_get_and_ioremap_resource() Zhu Ning (1): ASoC: codecs: ES8326: Add es8326_mute function Documentation/core-api/kernel-api.rst | 7 +- .../devicetree/bindings/sound/audio-iio-aux.yaml | 64 ++ .../devicetree/bindings/sound/fsl,rpmsg.yaml | 1 + .../bindings/sound/mediatek,mt8188-afe.yaml | 7 + .../devicetree/bindings/sound/nau8821.txt | 55 -- .../devicetree/bindings/sound/nuvoton,nau8821.yaml | 125 +++ .../devicetree/bindings/sound/nuvoton,nau8822.yaml | 12 + .../bindings/sound/qcom,msm8916-wcd-analog.txt | 101 --- .../sound/qcom,pm8916-wcd-analog-codec.yaml | 153 ++++ .../devicetree/bindings/sound/simple-card.yaml | 53 ++ .../devicetree/bindings/sound/wlf,wm8904.yaml | 74 ++ Documentation/devicetree/bindings/sound/wm8904.txt | 33 - Documentation/driver-api/media/dtv-common.rst | 9 - drivers/iio/inkern.c | 86 +- drivers/media/dvb-core/Makefile | 2 +- drivers/media/dvb-frontends/af9013_priv.h | 2 +- drivers/media/dvb-frontends/af9033_priv.h | 2 +- drivers/media/dvb-frontends/cxd2820r_priv.h | 2 +- drivers/media/dvb-frontends/cxd2841er.c | 2 +- .../cxd2880/cxd2880_tnrdmd_dvbt2_mon.c | 2 +- .../cxd2880/cxd2880_tnrdmd_dvbt_mon.c | 2 +- drivers/media/dvb-frontends/cxd2880/cxd2880_top.c | 2 +- drivers/media/dvb-frontends/dib7000p.c | 2 +- drivers/media/dvb-frontends/dib8000.c | 2 +- drivers/media/dvb-frontends/dib9000.c | 2 +- drivers/media/dvb-frontends/drxk_hard.c | 2 +- drivers/media/dvb-frontends/lgdt3305.c | 2 +- drivers/media/dvb-frontends/lgdt3306a.c | 2 +- drivers/media/dvb-frontends/lgdt330x.c | 2 +- drivers/media/dvb-frontends/m88ds3103_priv.h | 2 +- drivers/media/dvb-frontends/mn88443x.c | 2 +- drivers/media/dvb-frontends/mn88472_priv.h | 2 +- drivers/media/dvb-frontends/mn88473_priv.h | 2 +- drivers/media/dvb-frontends/or51132.c | 2 +- drivers/media/dvb-frontends/or51211.c | 2 +- drivers/media/dvb-frontends/rtl2830_priv.h | 2 +- drivers/media/dvb-frontends/rtl2832_priv.h | 2 +- drivers/media/dvb-frontends/si2165.c | 2 +- drivers/media/dvb-frontends/stv0367.c | 2 +- drivers/media/dvb-frontends/tc90522.c | 2 +- drivers/media/dvb-frontends/tda10048.c | 2 +- include/linux/device.h | 26 + include/linux/iio/consumer.h | 37 +- include/{media/dvb_math.h => linux/int_log.h} | 18 +- include/linux/minmax.h | 64 ++ include/linux/platform_device.h | 28 + include/sound/cs35l56.h | 29 +- include/sound/simple_card_utils.h | 5 +- include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 138 ++- include/sound/soc.h | 6 + lib/math/Makefile | 2 +- .../dvb-core/dvb_math.c => lib/math/int_log.c | 26 +- sound/Kconfig | 4 - sound/pci/hda/Kconfig | 31 + sound/pci/hda/Makefile | 6 + sound/pci/hda/cs35l56_hda.c | 995 +++++++++++++++++++++ sound/pci/hda/cs35l56_hda.h | 48 + sound/pci/hda/cs35l56_hda_i2c.c | 69 ++ sound/pci/hda/cs35l56_hda_spi.c | 68 ++ sound/pci/hda/hda_auto_parser.h | 2 + sound/pci/hda/hda_generic.h | 3 + sound/soc/Kconfig | 11 + sound/soc/amd/Kconfig | 5 +- sound/soc/amd/acp-config.c | 33 + sound/soc/amd/acp/Kconfig | 8 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-i2s.c | 2 + sound/soc/amd/acp/acp-legacy-common.c | 347 +++++++ sound/soc/amd/acp/acp-mach-common.c | 267 ++++++ sound/soc/amd/acp/acp-mach.h | 2 + sound/soc/amd/acp/acp-pci.c | 57 +- sound/soc/amd/acp/acp-pdm.c | 13 +- sound/soc/amd/acp/acp-platform.c | 6 +- sound/soc/amd/acp/acp-rembrandt.c | 184 ++-- sound/soc/amd/acp/acp-renoir.c | 115 +-- sound/soc/amd/acp/acp-sof-mach.c | 16 + sound/soc/amd/acp/amd.h | 51 ++ sound/soc/amd/mach-config.h | 1 + sound/soc/amd/ps/ps-sdw-dma.c | 5 +- sound/soc/amd/vangogh/acp5x-mach.c | 149 ++- sound/soc/bcm/bcm63xx-i2s-whistler.c | 16 +- sound/soc/codecs/Kconfig | 13 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1836.c | 2 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/adau1372.c | 2 +- sound/soc/codecs/adau1373.c | 2 +- sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/adau1761.c | 2 +- sound/soc/codecs/adau1781.c | 2 +- sound/soc/codecs/adau1977.c | 2 +- sound/soc/codecs/adau7118-i2c.c | 2 +- sound/soc/codecs/adav80x.c | 2 +- sound/soc/codecs/audio-iio-aux.c | 344 +++++++ sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs35l41-lib.c | 4 +- sound/soc/codecs/cs35l45-tables.c | 4 +- sound/soc/codecs/cs35l56-i2c.c | 14 +- sound/soc/codecs/cs35l56-sdw.c | 72 +- sound/soc/codecs/cs35l56-shared.c | 482 +++++++++- sound/soc/codecs/cs35l56-spi.c | 10 +- sound/soc/codecs/cs35l56.c | 651 +++----------- sound/soc/codecs/cs35l56.h | 15 +- sound/soc/codecs/cs4265.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs42l51.c | 2 +- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l56.c | 2 +- sound/soc/codecs/cs42xx8.c | 2 +- sound/soc/codecs/cs4349.c | 2 +- sound/soc/codecs/es8316.c | 5 +- sound/soc/codecs/es8326.c | 75 +- sound/soc/codecs/es8326.h | 9 +- sound/soc/codecs/lpass-rx-macro.c | 10 +- sound/soc/codecs/lpass-tx-macro.c | 10 +- sound/soc/codecs/lpass-va-macro.c | 6 +- sound/soc/codecs/lpass-wsa-macro.c | 10 +- sound/soc/codecs/msm8916-wcd-analog.c | 56 +- sound/soc/codecs/nau8825.c | 93 +- sound/soc/codecs/rt5645.c | 50 ++ sound/soc/codecs/rt5677.c | 117 +-- sound/soc/codecs/rt5677.h | 92 +- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca.c | 3 +- sound/soc/codecs/tas2781-i2c.c | 1 - sound/soc/codecs/wcd9335.c | 2 +- sound/soc/codecs/wcd938x-sdw.c | 2 +- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5100.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8737.c | 2 +- sound/soc/codecs/wm8741.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8770.c | 2 +- sound/soc/codecs/wm8776.c | 2 +- sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8904.c | 2 +- sound/soc/codecs/wm8940.c | 2 +- sound/soc/codecs/wm8955.c | 2 +- sound/soc/codecs/wm8960.c | 12 +- sound/soc/codecs/wm8961.c | 2 +- sound/soc/codecs/wm8962.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8978.c | 2 +- sound/soc/codecs/wm8983.c | 2 +- sound/soc/codecs/wm8985.c | 2 +- sound/soc/codecs/wm8988.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8995.c | 2 +- sound/soc/codecs/wm8996.c | 2 +- sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm9090.c | 2 +- sound/soc/codecs/wm9705.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 2 +- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 2 +- sound/soc/dwc/dwc-i2s.c | 65 +- sound/soc/dwc/dwc-pcm.c | 8 +- sound/soc/dwc/local.h | 24 + sound/soc/fsl/fsl_rpmsg.c | 8 + sound/soc/fsl/fsl_spdif.c | 8 +- sound/soc/fsl/fsl_spdif.h | 6 +- sound/soc/fsl/imx-pcm-rpmsg.c | 12 +- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/generic/simple-card-utils.c | 23 +- sound/soc/generic/simple-card.c | 66 +- sound/soc/intel/avs/board_selection.c | 16 + sound/soc/intel/avs/boards/Kconfig | 20 + sound/soc/intel/avs/boards/Makefile | 4 + sound/soc/intel/avs/boards/es8336.c | 315 +++++++ sound/soc/intel/avs/boards/rt5663.c | 254 ++++++ sound/soc/intel/avs/boards/rt5682.c | 67 +- sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/sof_rt5682.c | 80 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 12 + sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 4 +- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 8 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 15 + sound/soc/pxa/pxa2xx-i2s.c | 13 +- sound/soc/qcom/qdsp6/q6apm.c | 2 +- sound/soc/soc-core.c | 273 ++++-- sound/soc/sof/amd/Kconfig | 8 + sound/soc/sof/amd/Makefile | 1 + sound/soc/sof/amd/acp-common.c | 4 + sound/soc/sof/amd/acp-ipc.c | 26 + sound/soc/sof/amd/acp-probes.c | 147 +++ sound/soc/sof/amd/acp.c | 34 +- sound/soc/sof/amd/acp.h | 13 +- sound/soc/sof/amd/pci-rmb.c | 2 + sound/soc/sof/amd/pci-rn.c | 2 + sound/soc/sof/sof-client-probes.c | 1 + sound/soc/starfive/jh7110_tdm.c | 5 +- sound/soc/tegra/tegra20_ac97.c | 3 +- sound/soc/ti/omap-dmic.c | 4 +- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-mcpdm.c | 3 +- tools/testing/kunit/configs/all_tests.config | 5 + 209 files changed, 5743 insertions(+), 1841 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-iio-aux.yaml delete mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml delete mode 100644 Documentation/devicetree/bindings/sound/qcom,msm8916-wcd-analog.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,pm8916-wcd-analog-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8904.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt rename include/{media/dvb_math.h => linux/int_log.h} (63%) rename drivers/media/dvb-core/dvb_math.c => lib/math/int_log.c (84%) create mode 100644 sound/pci/hda/cs35l56_hda.c create mode 100644 sound/pci/hda/cs35l56_hda.h create mode 100644 sound/pci/hda/cs35l56_hda_i2c.c create mode 100644 sound/pci/hda/cs35l56_hda_spi.c create mode 100644 sound/soc/amd/acp/acp-legacy-common.c create mode 100644 sound/soc/codecs/audio-iio-aux.c create mode 100644 sound/soc/intel/avs/boards/es8336.c create mode 100644 sound/soc/intel/avs/boards/rt5663.c create mode 100644 sound/soc/sof/amd/acp-probes.c

2 1

[PATCH] ALSA: hda/relatek: Enable Mute LED on HP 250 G8
by Luka Guzenko 25 Jul '23

25 Jul '23

This HP Notebook used ALC236 codec with COEF 0x07 idx 1 controlling the mute LED. Enable already existing quirk for this device. Signed-off-by: Luka Guzenko <l.guzenko(a)web.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 52ea7d757d6a..9a65dfb00cb2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9541,6 +9541,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8811, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), SND_PCI_QUIRK(0x103c, 0x8812, "HP Spectre x360 15-eb1xxx", ALC285_FIXUP_HP_SPECTRE_X360_EB1), + SND_PCI_QUIRK(0x103c, 0x881d, "HP 250 G8 Notebook PC", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8846, "HP EliteBook 850 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8847, "HP EliteBook x360 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x884b, "HP EliteBook 840 Aero G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.41.0

2 1

[PATCH] ASoC: SOF: ipc4-topology: fix application of sizeof to pointer
by sunran001＠208suo.com 25 Jul '23

25 Jul '23

The coccinelle check report: ./drivers/scsi/csiostor/csio_mb.c:1554:46-52: ERROR: application of sizeof to pointer Signed-off-by: Ran Sun <sunran001(a)208suo.com> --- sound/soc/sof/ipc4-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index a4e1a70b607d..2dbe87dbd239 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -218,7 +218,7 @@ static int sof_ipc4_get_audio_fmt(struct snd_soc_component *scomp, ret = sof_update_ipc_object(scomp, available_fmt, SOF_AUDIO_FMT_NUM_TOKENS, swidget->tuples, - swidget->num_tuples, sizeof(available_fmt), 1); + swidget->num_tuples, sizeof(*available_fmt), 1); if (ret) { dev_err(scomp->dev, "Failed to parse audio format token count\n"); return ret;

4 3

[PATCH v5 00/11] ALSA: hda: Adding support for CS35L56 on HDA systems
by Richard Fitzgerald 25 Jul '23

25 Jul '23

This set of patches adds support for using the CS35L56 boosted smart amplifier on HDA systems. In these systems the CS35L56 audio is routed through a HDA-to-I2S bridge codec. This doesn't include the changes to the Realtek driver to actually hook up the CS35L56 driver, because we don't yet have the QUIRK IDs to associate it with. But we want to publish the driver now so that it is available for bringing up hardware with the CS35L56. The first 9 patches are moving code out of the ASoC driver and into the shared library so that it can be shared with the HDA driver. Patch #10 fixes missing #includes in the HDA headers so that the CS35L56 driver doesn't have to #include headers that it doesn't use. CHANGES SINCE V5 The series has been rebased onto commit df4167d658d4 ("ASoC: cs35l56: Patch soft registers to defaults"), which removes cs35l56_reread_firmware_registers() and adds cs35l56_set_patch(). Changes are trivial so I have carried forward Mark's and Takashi's acks. Patch 01: - Also changes cs35l56_set_patch() argument to new struct cs35l56_base. Patch 11: - Remove call to cs35l56_reread_firmware_registers() - Add call to cs35l56_set_patch() in cs35l56_hda_common_probe() - Do not add mute_hook to struct hda_component. This was for use with CS8409 as an HDA bridge, but production systems will use Realtek, which has its own mute control. - Use strscpy() instead of strcpy() in cs35l56_hda_mixer_info() - Fix missing word in Kconfig option title Richard Fitzgerald (4): ASoC: cs35l56: Move runtime suspend/resume to shared library ASoC: cs35l56: Move cs_dsp init into shared library ASoC: cs35l56: Move part of cs35l56_init() to shared library ALSA: hda: Fix missing header dependencies Simon Trimmer (7): ASoC: cs35l56: Move shared data into a common data structure ASoC: cs35l56: Make cs35l56_system_reset() code more generic ASoC: cs35l56: Convert utility functions to use common data structure ASoC: cs35l56: Move utility functions to shared file ASoC: cs35l56: Make common function for control port wait ASoC: cs35l56: Make a common function to shutdown the DSP ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier include/sound/cs35l56.h | 29 +- sound/pci/hda/Kconfig | 31 + sound/pci/hda/Makefile | 6 + sound/pci/hda/cs35l56_hda.c | 995 ++++++++++++++++++++++++++++++ sound/pci/hda/cs35l56_hda.h | 48 ++ sound/pci/hda/cs35l56_hda_i2c.c | 69 +++ sound/pci/hda/cs35l56_hda_spi.c | 68 ++ sound/pci/hda/hda_auto_parser.h | 2 + sound/pci/hda/hda_generic.h | 3 + sound/soc/codecs/cs35l56-i2c.c | 14 +- sound/soc/codecs/cs35l56-sdw.c | 72 +-- sound/soc/codecs/cs35l56-shared.c | 458 +++++++++++++- sound/soc/codecs/cs35l56-spi.c | 10 +- sound/soc/codecs/cs35l56.c | 648 ++++--------------- sound/soc/codecs/cs35l56.h | 15 +- 15 files changed, 1877 insertions(+), 591 deletions(-) create mode 100644 sound/pci/hda/cs35l56_hda.c create mode 100644 sound/pci/hda/cs35l56_hda.h create mode 100644 sound/pci/hda/cs35l56_hda_i2c.c create mode 100644 sound/pci/hda/cs35l56_hda_spi.c -- 2.30.2

3 17

[PATCH] ALSA: usb-audio: Add quirk for Microsoft Modern Wireless Headset
by Takashi Iwai 25 Jul '23

25 Jul '23

Microsoft Modern Wireless Headset (appearing on the host as "Microsoft USB Link") has a playback and a capture mixer volume/switch, but they are fairly broken. The descriptor reports wrong dB ranges for playback, and the capture volume/switch don't influence on the actual recording at all. Moreover, there seem instabilities in the connection, and at best, we should disable the runtime PM. So this ended up with a quirk entry for: - Correct the playback dB range; I picked up some reasonable values but it's a guess work - Disable the capture mixer; it's completely useless and confuses PA/PW - Suppress get-sample-rate, apply the delay for message handling, and suppress the auto-suspend The behavior of the wheel control on the headset is somehow flaky, too, but it's an issue of HID. Link: https://bugzilla.suse.com/show_bug.cgi?id=1207129 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_maps.c | 14 ++++++++++++++ sound/usb/quirks.c | 3 +++ 2 files changed, 17 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index f4bd1e8ae4b6..23260aa1919d 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -374,6 +374,15 @@ static const struct usbmix_name_map corsair_virtuoso_map[] = { { 0 } }; +/* Microsoft USB Link headset */ +/* a guess work: raw playback volume values are from 2 to 129 */ +static const struct usbmix_dB_map ms_usb_link_dB = { -3225, 0, true }; +static const struct usbmix_name_map ms_usb_link_map[] = { + { 9, NULL, .dB = &ms_usb_link_dB }, + { 10, NULL }, /* Headset Capture volume; seems non-working, disabled */ + { 0 } /* terminator */ +}; + /* ASUS ROG Zenith II with Realtek ALC1220-VB */ static const struct usbmix_name_map asus_zenith_ii_map[] = { { 19, NULL, 12 }, /* FU, Input Gain Pad - broken response, disabled */ @@ -668,6 +677,11 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x1395, 0x0025), .map = sennheiser_pc8_map, }, + { + /* Microsoft USB Link headset */ + .id = USB_ID(0x045e, 0x083c), + .map = ms_usb_link_map, + }, { 0 } /* terminator */ }; diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 30bcb80b1cb8..f9e200309b9e 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -2013,6 +2013,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x041e, 0x4080, /* Creative Live Cam VF0610 */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x045e, 0x083c, /* MS USB Link headset */ + QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_CTL_MSG_DELAY | + QUIRK_FLAG_DISABLE_AUTOSUSPEND), DEVICE_FLG(0x046d, 0x084c, /* Logitech ConferenceCam Connect */ QUIRK_FLAG_GET_SAMPLE_RATE | QUIRK_FLAG_CTL_MSG_DELAY_1M), DEVICE_FLG(0x046d, 0x0991, /* Logitech QuickCam Pro */ -- 2.35.3

1 0

alsa-utils-1.2.4's aplay play 11.025k wav file by digital Mi Headphone (S24_3LE) with abnormal electrical noise sound
by GitHub issues - opened 25 Jul '23

25 Jul '23

alsa-project/alsa-lib issue #338 was opened from raintiliang: Hi, we use aplay play 11.025k wav file by digital Mi Headphone (S24_3LE) with abnormal electrical noise sound, But aplay 8K, 16K, 32K,48K wav as normal except 11.025k, 22.05k, 44.1k. May i have any idea ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/338 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1 0/5] Add PWM-DAC audio support for StarFive JH7110 RISC-V SoC
by Hal Feng 25 Jul '23

25 Jul '23

This patchset adds PWM-DAC audio support for the StarFive JH7110 SoC. The PWM-DAC module does not require a hardware codec, so add a dummy codec driver for it. The fourth patch depends on the patch [1]. The fifth patch depends on the patchset [2], patch [3]. [1] https://lore.kernel.org/all/20230526145402.450-3-walker.chen@starfivetech.c… [2] https://lore.kernel.org/all/20230518101234.143748-1-xingyu.wu@starfivetech.… [3] https://lore.kernel.org/all/20230322094820.24738-5-walker.chen@starfivetech… Hal Feng (5): ASoC: dt-bindings: Add StarFive JH7110 dummy PWM-DAC transmitter ASoC: codecs: Add StarFive JH7110 dummy PWM-DAC transmitter driver ASoC: dt-bindings: Add StarFive JH7110 PWM-DAC controller ASoC: starfive: Add JH7110 PWM-DAC driver riscv: dts: starfive: Add JH7110 PWM-DAC support .../sound/starfive,jh7110-pwmdac-dit.yaml | 38 + .../sound/starfive,jh7110-pwmdac.yaml | 76 ++ MAINTAINERS | 8 + .../jh7110-starfive-visionfive-2.dtsi | 50 ++ arch/riscv/boot/dts/starfive/jh7110.dtsi | 13 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/jh7110_pwmdac_transmitter.c | 74 ++ sound/soc/starfive/Kconfig | 9 + sound/soc/starfive/Makefile | 1 + sound/soc/starfive/jh7110_pwmdac.c | 787 ++++++++++++++++++ 11 files changed, 1062 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac-dit.yaml create mode 100644 Documentation/devicetree/bindings/sound/starfive,jh7110-pwmdac.yaml create mode 100644 sound/soc/codecs/jh7110_pwmdac_transmitter.c create mode 100644 sound/soc/starfive/jh7110_pwmdac.c base-commit: 45a3e24f65e90a047bef86f927ebdc4c710edaa1 prerequisite-patch-id: 8c735dffc6d5388a35a76b16e914a2f9722ad979 prerequisite-patch-id: ffa1f5831e75722c9f41603f009f762c9fd525e2 prerequisite-patch-id: 36e69700dfc0375b950b0e23086ed3b722cb84a4 prerequisite-patch-id: 0b49b996d7a404ea548e1734c12933ec749e92b9 prerequisite-patch-id: 81f7c65712c4901a7a178ddcd98ffc55f3b473ff prerequisite-patch-id: f342fbf594014b072378528bea94c01fb2186e1a prerequisite-patch-id: 39e1be2a3d1593577ab997f55f59367cba665aa7 prerequisite-patch-id: 0159f09bb0a1ff711a00ae17ef5b12662c9c7d3d prerequisite-patch-id: 2ddada18ab6ea5cd1da14212aaf59632f5203d40 prerequisite-patch-id: d5abfba63fc07ff97b5023911513c260bb7a53e1 prerequisite-patch-id: b37ac15032973e1fcd918f157c82a0606775c9e9 prerequisite-patch-id: 6abf359fa445f4104432ddee27044dfbfb128417 prerequisite-patch-id: 2f7aca99e714a4c590a91baa015080ac0902814d prerequisite-patch-id: 32cabbc4e7a97ec14d5c28a477fa483784f86709 prerequisite-patch-id: d449b1957dd77c2537c38585daa75974c94c529a -- 2.38.1

3 20

[PATCH] ASoC: wm8904: Fill the cache for WM8904_ADC_TEST_0 register
by Mark Brown 24 Jul '23

24 Jul '23

The WM8904_ADC_TEST_0 register is modified as part of updating the OSR controls but does not have a cache default, leading to errors when we try to modify these controls in cache only mode with no prior read: wm8904 3-001a: ASoC: error at snd_soc_component_update_bits on wm8904.3-001a for register: [0x000000c6] -16 Add a read of the register to probe() to fill the cache and avoid both the error messages and the misconfiguration of the chip which will result. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm8904.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 068e610b1b4c..f2baee7c332e 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2308,6 +2308,9 @@ static int wm8904_i2c_probe(struct i2c_client *i2c) regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0, WM8904_POBCTRL, 0); + /* Fill the cache for the ADC test register */ + regmap_read(wm8904->regmap, WM8904_ADC_TEST_0, &val); + /* Can leave the device powered off until we need it */ regcache_cache_only(wm8904->regmap, true); regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); --- base-commit: fdf0eaf11452d72945af31804e2a1048ee1b574c change-id: 20230722-asoc-fix-wm8904-adc-test-read-5a0051264a0b Best regards, -- Mark Brown <broonie(a)kernel.org>

2 2

[PATCH 0/2] ASoC: da7219: Patches related to a spurious AAD IRQ issue
by Dmytro Maluka 24 Jul '23

24 Jul '23

This series includes 2 patches related to (but not fixing) the following I2C failure which occurs sometimes during system suspend or resume and indicates a problem with a spurious DA7219 interrupt: [ 355.876211] i2c_designware i2c_designware.3: Transfer while suspended [ 355.876245] WARNING: CPU: 2 PID: 3576 at drivers/i2c/busses/i2c-designware-master.c:570 i2c_dw_xfer+0x411/0x440 ... [ 355.876462] Call Trace: [ 355.876468] <TASK> [ 355.876475] ? update_load_avg+0x1b3/0x615 [ 355.876484] __i2c_transfer+0x101/0x1d8 [ 355.876494] i2c_transfer+0x74/0x10d [ 355.876504] regmap_i2c_read+0x6a/0x9c [ 355.876513] _regmap_raw_read+0x179/0x223 [ 355.876521] regmap_raw_read+0x1e1/0x28e [ 355.876527] regmap_bulk_read+0x17d/0x1ba [ 355.876532] ? __wake_up+0xed/0x1bb [ 355.876542] da7219_aad_irq_thread+0x54/0x2c9 [snd_soc_da7219 5fb8ebb2179cf2fea29af090f3145d68ed8e2184] [ 355.876556] irq_thread+0x13c/0x231 [ 355.876563] ? irq_forced_thread_fn+0x5f/0x5f [ 355.876570] ? irq_thread_fn+0x4d/0x4d [ 355.876576] kthread+0x13a/0x152 [ 355.876581] ? synchronize_irq+0xc3/0xc3 [ 355.876587] ? kthread_blkcg+0x31/0x31 [ 355.876592] ret_from_fork+0x1f/0x30 [ 355.876601] </TASK> This log shows that DA7219 AAD interrupt handler da7219_aad_irq_thread() is unexpectedly running when DA7219 is suspended and should not generate interrupts. As a result, the IRQ handler is trying to read AAD IRQ event status over I2C and is hitting the I2C driver "Transfer while suspended" failure. Patch #1 adds synchronize_irq() when suspending DA7219, to prevent the IRQ handler from running after suspending if there is a pending IRQ generated before suspending. With this patch the above failure is still reproducible, so this patch does not fix any real observed issue so far, but at least is useful for confirming that the above issue is not caused by a pending IRQ but rather looks like a DA7219 hardware issue with an unexpectedly generated IRQ. Patch #2 does not fix the above issue either, but it prevents its potentially harmful side effects. With the existing code, if the issue occurs and the IRQ handler fails to read the AAD IRQ events status over I2C, it does not check that and tries to use the garbage uninitialized value of the events status, potentially reporting bogus events. This patch fixes that by adding missing error checking. In fact I'm sending these patches not only to submit them for review but also to ask Renesas folks for any hints on a possible cause of the described DA7219 issue (AAD interrupts spuriously firing after jack detection is already disabled) or how to debug it further. Dmytro Maluka (2): ASoC: da7219: Flush pending AAD IRQ when suspending ASoC: da7219: Check for failure reading AAD IRQ events sound/soc/codecs/da7219-aad.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) base-commit: fdf0eaf11452d72945af31804e2a1048ee1b574c -- 2.41.0.255.g8b1d071c50-goog

2 3

[PATCH] ASoC: nau8821: Add DMI quirk mechanism for active-high jack-detect
by Edson Juliano Drosdeck 24 Jul '23

24 Jul '23

Add a quirk mechanism to allow specifying that active-high jack-detection should be used on platforms where this info is not available in devicetree. And add an entry for the Positivo CW14Q01P-V2 to the DMI table, so that jack-detection will work properly on this laptop. Signed-off-by: Edson Juliano Drosdeck <edson.drosdeck(a)gmail.com> --- sound/soc/codecs/nau8821.c | 41 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 41 insertions(+) diff --git a/sound/soc/codecs/nau8821.c b/sound/soc/codecs/nau8821.c index 96d75882b33a..ca6beb2d2649 100644 --- a/sound/soc/codecs/nau8821.c +++ b/sound/soc/codecs/nau8821.c @@ -10,6 +10,7 @@ #include <linux/acpi.h> #include <linux/clk.h> #include <linux/delay.h> +#include <linux/dmi.h> #include <linux/init.h> #include <linux/i2c.h> #include <linux/module.h> @@ -25,6 +26,13 @@ #include <sound/tlv.h> #include "nau8821.h" +#define NAU8821_JD_ACTIVE_HIGH BIT(0) + +static int nau8821_quirk; +static int quirk_override = -1; +module_param_named(quirk, quirk_override, uint, 0444); +MODULE_PARM_DESC(quirk, "Board-specific quirk override"); + #define NAU_FREF_MAX 13500000 #define NAU_FVCO_MAX 100000000 #define NAU_FVCO_MIN 90000000 @@ -1792,6 +1800,33 @@ static int nau8821_setup_irq(struct nau8821 *nau8821) return 0; } +/* Please keep this list alphabetically sorted */ +static const struct dmi_system_id nau8821_quirk_table[] = { + { + /* Positivo CW14Q01P-V2 */ + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Positivo Tecnologia SA"), + DMI_MATCH(DMI_BOARD_NAME, "CW14Q01P-V2"), + }, + .driver_data = (void *)(NAU8821_JD_ACTIVE_HIGH), + }, + {} +}; + +static void nau8821_check_quirks(void) +{ + const struct dmi_system_id *dmi_id; + + if (quirk_override != -1) { + nau8821_quirk = quirk_override; + return; + } + + dmi_id = dmi_first_match(nau8821_quirk_table); + if (dmi_id) + nau8821_quirk = (unsigned long)dmi_id->driver_data; +} + static int nau8821_i2c_probe(struct i2c_client *i2c) { struct device *dev = &i2c->dev; @@ -1812,6 +1847,12 @@ static int nau8821_i2c_probe(struct i2c_client *i2c) nau8821->dev = dev; nau8821->irq = i2c->irq; + + nau8821_check_quirks(); + + if (nau8821_quirk & NAU8821_JD_ACTIVE_HIGH) + nau8821->jkdet_polarity = 0; + nau8821_print_device_properties(nau8821); nau8821_reset_chip(nau8821->regmap); -- 2.39.2

2 1

[PATCH] ASoC: codecs: lpass: Log clk_get() failures
by Bjorn Andersson 24 Jul '23

24 Jul '23

The LPASS macro drivers all acquire a number of clocks, but give no indication when clk_get() fails, making it hard to identify and debug system configuration issues. Make these drivers provide useful debug information when this happens. Signed-off-by: Bjorn Andersson <quic_bjorande(a)quicinc.com> --- sound/soc/codecs/lpass-rx-macro.c | 10 +++++----- sound/soc/codecs/lpass-tx-macro.c | 10 +++++----- sound/soc/codecs/lpass-va-macro.c | 6 +++--- sound/soc/codecs/lpass-wsa-macro.c | 10 +++++----- 4 files changed, 18 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 685ca95ef4a9..29197d34ec09 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -3537,25 +3537,25 @@ static int rx_macro_probe(struct platform_device *pdev) rx->macro = devm_clk_get_optional(dev, "macro"); if (IS_ERR(rx->macro)) - return PTR_ERR(rx->macro); + return dev_err_probe(dev, PTR_ERR(rx->macro), "unable to get macro clock\n"); rx->dcodec = devm_clk_get_optional(dev, "dcodec"); if (IS_ERR(rx->dcodec)) - return PTR_ERR(rx->dcodec); + return dev_err_probe(dev, PTR_ERR(rx->dcodec), "unable to get dcodec clock\n"); rx->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(rx->mclk)) - return PTR_ERR(rx->mclk); + return dev_err_probe(dev, PTR_ERR(rx->mclk), "unable to get mclk clock\n"); if (flags & LPASS_MACRO_FLAG_HAS_NPL_CLOCK) { rx->npl = devm_clk_get(dev, "npl"); if (IS_ERR(rx->npl)) - return PTR_ERR(rx->npl); + return dev_err_probe(dev, PTR_ERR(rx->npl), "unable to get npl clock\n"); } rx->fsgen = devm_clk_get(dev, "fsgen"); if (IS_ERR(rx->fsgen)) - return PTR_ERR(rx->fsgen); + return dev_err_probe(dev, PTR_ERR(rx->fsgen), "unable to get fsgen clock\n"); rx->pds = lpass_macro_pds_init(dev); if (IS_ERR(rx->pds)) diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index de978c3d70b7..3e33418898e8 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -1967,25 +1967,25 @@ static int tx_macro_probe(struct platform_device *pdev) tx->macro = devm_clk_get_optional(dev, "macro"); if (IS_ERR(tx->macro)) - return PTR_ERR(tx->macro); + return dev_err_probe(dev, PTR_ERR(tx->macro), "unable to get macro clock\n"); tx->dcodec = devm_clk_get_optional(dev, "dcodec"); if (IS_ERR(tx->dcodec)) - return PTR_ERR(tx->dcodec); + return dev_err_probe(dev, PTR_ERR(tx->dcodec), "unable to get dcodec clock\n"); tx->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(tx->mclk)) - return PTR_ERR(tx->mclk); + return dev_err_probe(dev, PTR_ERR(tx->mclk), "unable to get mclk clock\n"); if (flags & LPASS_MACRO_FLAG_HAS_NPL_CLOCK) { tx->npl = devm_clk_get(dev, "npl"); if (IS_ERR(tx->npl)) - return PTR_ERR(tx->npl); + return dev_err_probe(dev, PTR_ERR(tx->npl), "unable to get npl clock\n"); } tx->fsgen = devm_clk_get(dev, "fsgen"); if (IS_ERR(tx->fsgen)) - return PTR_ERR(tx->fsgen); + return dev_err_probe(dev, PTR_ERR(tx->fsgen), "unable to get fsgen clock\n"); tx->pds = lpass_macro_pds_init(dev); if (IS_ERR(tx->pds)) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 74724448da50..b71ef03c4aef 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1457,15 +1457,15 @@ static int va_macro_probe(struct platform_device *pdev) va->macro = devm_clk_get_optional(dev, "macro"); if (IS_ERR(va->macro)) - return PTR_ERR(va->macro); + return dev_err_probe(dev, PTR_ERR(va->macro), "unable to get macro clock\n"); va->dcodec = devm_clk_get_optional(dev, "dcodec"); if (IS_ERR(va->dcodec)) - return PTR_ERR(va->dcodec); + return dev_err_probe(dev, PTR_ERR(va->dcodec), "unable to get dcodec clock\n"); va->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(va->mclk)) - return PTR_ERR(va->mclk); + return dev_err_probe(dev, PTR_ERR(va->mclk), "unable to get mclk clock\n"); va->pds = lpass_macro_pds_init(dev); if (IS_ERR(va->pds)) diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 8ba7dc89daaa..ec6859ec0d38 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -2396,25 +2396,25 @@ static int wsa_macro_probe(struct platform_device *pdev) wsa->macro = devm_clk_get_optional(dev, "macro"); if (IS_ERR(wsa->macro)) - return PTR_ERR(wsa->macro); + return dev_err_probe(dev, PTR_ERR(wsa->macro), "unable to get macro clock\n"); wsa->dcodec = devm_clk_get_optional(dev, "dcodec"); if (IS_ERR(wsa->dcodec)) - return PTR_ERR(wsa->dcodec); + return dev_err_probe(dev, PTR_ERR(wsa->dcodec), "unable to get dcodec clock\n"); wsa->mclk = devm_clk_get(dev, "mclk"); if (IS_ERR(wsa->mclk)) - return PTR_ERR(wsa->mclk); + return dev_err_probe(dev, PTR_ERR(wsa->mclk), "unable to get mclk clock\n"); if (flags & LPASS_MACRO_FLAG_HAS_NPL_CLOCK) { wsa->npl = devm_clk_get(dev, "npl"); if (IS_ERR(wsa->npl)) - return PTR_ERR(wsa->npl); + return dev_err_probe(dev, PTR_ERR(wsa->npl), "unable to get npl clock\n"); } wsa->fsgen = devm_clk_get(dev, "fsgen"); if (IS_ERR(wsa->fsgen)) - return PTR_ERR(wsa->fsgen); + return dev_err_probe(dev, PTR_ERR(wsa->fsgen), "unable to get fsgen clock\n"); base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) -- 2.25.1

3 2

[PATCH] Make snd-hda-intel work for newest HP laptop models with Realtek codec ALC245 and CSC3551 chipset
by Matthias Ostermaier 24 Jul '23

24 Jul '23

Hardware: HP laptop models - HP Dragonfly Folio 13.5 inch - HP Envy x360 15-ew0xxx - HP Envy 16-h0xxx all with codec Realtek ALC245 (ALC287 variant) and Cirrus Logic Smart Amplifier chipset CSC3551) codec vendor_id: 0x10ec0245, subsystem_id: 0x103c8a29, revision_id:0x100001 Kernel version: 6.2+ Output of alsa-info.sh attached (with the patch applied) Problem description: The CSC3551 chipset does not get initialized correctly in aforementioned laptop models. While the rest of snd-hda-intel recognizes the hardware correctly, the smart amplifier does not get initialized so that the built-in loudspeakers do not work. The patches originate from discussions and tests under https://github.com/xoocoon/hp-15-ew0xxx-snd-fix. Patches: ---------------------------------------------------------------------------- --- sound/pci/hda/cs35l41_hda.c.orig 2023-07-22 15:43:12.016044919 +0200 +++ sound/pci/hda/cs35l41_hda.c 2023-07-22 16:10:23.937995251 +0200 @@ -1244,6 +1244,10 @@ hw_cfg->bst_type = CS35L41_EXT_BOOST; hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; hw_cfg->gpio1.valid = true; + } else if (strncmp(hid, "CSC3551", 7) == 0) { + hw_cfg->bst_type = CS35L41_EXT_BOOST; + hw_cfg->gpio1.func = CS35l41_VSPK_SWITCH; + hw_cfg->gpio1.valid = true; } else { /* * Note: CLSA010(0/1) are special cases which use a slightly different design. --- sound/pci/hda/patch_realtek.c.orig 2023-07-22 15:43:58.711204631 +0200 +++ sound/pci/hda/patch_realtek.c 2023-07-22 15:51:23.224124227 +0200 @@ -9452,6 +9452,9 @@ SND_PCI_QUIRK(0x103c, 0x89c6, "Zbook Fury 17 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ca, "HP", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), SND_PCI_QUIRK(0x103c, 0x89d3, "HP EliteBook 645 G9 (MB 89D2)", ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF), + SND_PCI_QUIRK(0x103c, 0x8a06, "HP Dragonfly Folio 13.5 inch G3 2-in-1 Notebook PC", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x103c, 0x8a29, "HP Envy x360 15-ew0xxx", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8a2c, "HP Envy 16-h0xxx", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x8a78, "HP Dev One", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x8aa0, "HP ProBook 440 G9 (MB 8A9E)", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8aa3, "HP ProBook 450 G9 (MB 8AA1)", ALC236_FIXUP_HP_GPIO_LED), -------------------------------------------------------------------------- Unresolved issues: The LED indicating muted sound on the laptop keyboard does not work. The same holds for the LED indicating a deactivated microphone. These issues are not addressed by the patches.

2 1

Re: [PATCH v2 0/7] ASoC: mediatek: Add support for MT7986 SoC
by Krzysztof Kozlowski 24 Jul '23

24 Jul '23

On 24/07/2023 08:17, Maso Huang (黃加竹) wrote: > Gentle ping :) > > Hi Rob and Angelo, > > Can you help to check this latest series and share your thoughts if > any? You got comments to fix! Implement what you were asked to do. Best regards, Krzysztof

1 0

[PATCH AUTOSEL 6.4 28/58] ALSA: hda: Add Loongson LS7A HD-Audio support
by Sasha Levin 24 Jul '23

24 Jul '23

From: Yanteng Si <siyanteng(a)loongson.cn> [ Upstream commit 28bd137a3c8e105587ba8c55b68ef43b519b270f ] Add the new PCI ID 0x0014 0x7a07 and the new PCI ID 0x0014 0x7a37 Loongson HDA controller. Signed-off-by: Yanteng Si <siyanteng(a)loongson.cn> Acked-by: Huacai Chen <chenhuacai(a)loongson.cn> Link: https://lore.kernel.org/r/993587483b9509796b29a416f257fcfb4b15c6ea.16861288… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/linux/pci_ids.h | 3 +++ sound/hda/hdac_device.c | 1 + sound/pci/hda/hda_intel.c | 7 +++++++ sound/pci/hda/patch_hdmi.c | 1 + 4 files changed, 12 insertions(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index 95f33dadb2be2..c0c4ca8e28510 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -158,6 +158,9 @@ #define PCI_VENDOR_ID_LOONGSON 0x0014 +#define PCI_DEVICE_ID_LOONGSON_HDA 0x7a07 +#define PCI_DEVICE_ID_LOONGSON_HDMI 0x7a37 + #define PCI_VENDOR_ID_TTTECH 0x0357 #define PCI_DEVICE_ID_TTTECH_MC322 0x000a diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 6c043fbd606f1..bbf7bcdb449a8 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -645,6 +645,7 @@ struct hda_vendor_id { }; static const struct hda_vendor_id hda_vendor_ids[] = { + { 0x0014, "Loongson" }, { 0x1002, "ATI" }, { 0x1013, "Cirrus Logic" }, { 0x1057, "Motorola" }, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3226691ac923c..9c353dc7740c4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -237,6 +237,7 @@ enum { AZX_DRIVER_CTHDA, AZX_DRIVER_CMEDIA, AZX_DRIVER_ZHAOXIN, + AZX_DRIVER_LOONGSON, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -360,6 +361,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_ZHAOXIN] = "HDA Zhaoxin", + [AZX_DRIVER_LOONGSON] = "HDA Loongson", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -2809,6 +2811,11 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI }, /* Zhaoxin */ { PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN }, + /* Loongson HDAudio*/ + {PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, PCI_DEVICE_ID_LOONGSON_HDA), + .driver_data = AZX_DRIVER_LOONGSON }, + {PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, PCI_DEVICE_ID_LOONGSON_HDMI), + .driver_data = AZX_DRIVER_LOONGSON }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5c0b1a09fd57c..260d3e64f6589 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4505,6 +4505,7 @@ static int patch_gf_hdmi(struct hda_codec *codec) * patch entries */ static const struct hda_device_id snd_hda_id_hdmi[] = { +HDA_CODEC_ENTRY(0x00147a47, "Loongson HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x1002793c, "RS600 HDMI", patch_atihdmi), HDA_CODEC_ENTRY(0x10027919, "RS600 HDMI", patch_atihdmi), HDA_CODEC_ENTRY(0x1002791a, "RS690/780 HDMI", patch_atihdmi), -- 2.39.2

2 1

[PATCH AUTOSEL 5.4 12/12] ALSA: hda: fix a possible null-pointer dereference due to data race in snd_hdac_regmap_sync()
by Sasha Levin 24 Jul '23

24 Jul '23

From: Tuo Li <islituo(a)gmail.com> [ Upstream commit 1f4a08fed450db87fbb5ff5105354158bdbe1a22 ] The variable codec->regmap is often protected by the lock codec->regmap_lock when is accessed. However, it is accessed without holding the lock when is accessed in snd_hdac_regmap_sync(): if (codec->regmap) In my opinion, this may be a harmful race, because if codec->regmap is set to NULL right after the condition is checked, a null-pointer dereference can occur in the called function regcache_sync(): map->lock(map->lock_arg); --> Line 360 in drivers/base/regmap/regcache.c To fix this possible null-pointer dereference caused by data race, the mutex_lock coverage is extended to protect the if statement as well as the function call to regcache_sync(). [ Note: the lack of the regmap_lock itself is harmless for the current codec driver implementations, as snd_hdac_regmap_sync() is only for PM runtime resume that is prohibited during the codec probe. But the change makes the whole code more consistent, so it's merged as is -- tiwai ] Reported-by: BassCheck <bass(a)buaa.edu.cn> Signed-off-by: Tuo Li <islituo(a)gmail.com> Link: https://lore.kernel.org/r/20230703031016.1184711-1-islituo@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_regmap.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index 49780399c2849..a035a7d74ce09 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -596,10 +596,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw_once); */ void snd_hdac_regmap_sync(struct hdac_device *codec) { - if (codec->regmap) { - mutex_lock(&codec->regmap_lock); + mutex_lock(&codec->regmap_lock); + if (codec->regmap) regcache_sync(codec->regmap); - mutex_unlock(&codec->regmap_lock); - } + mutex_unlock(&codec->regmap_lock); } EXPORT_SYMBOL_GPL(snd_hdac_regmap_sync); -- 2.39.2

1 0

[PATCH AUTOSEL 5.4 11/12] ALSA: hda/realtek: Add quirks for Unis H3C Desktop B760 & Q760
by Sasha Levin 24 Jul '23

24 Jul '23

From: dengxiang <dengxiang(a)nfschina.com> [ Upstream commit 73f1c75d5e6bd8ce2a887ef493a66ad1b16ed704 ] These models use NSIWAY amplifiers for internal speaker, but cannot put sound outside from these amplifiers. So eapd verbs are needed to initialize the amplifiers. They can be added during boot to get working sound out of internal speaker. Signed-off-by: dengxiang <dengxiang(a)nfschina.com> Link: https://lore.kernel.org/r/20230703021751.2945750-1-dengxiang@nfschina.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 6d8d9fc1da0b0..c0bcbab7b6560 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9877,6 +9877,7 @@ enum { ALC897_FIXUP_HP_HSMIC_VERB, ALC897_FIXUP_LENOVO_HEADSET_MODE, ALC897_FIXUP_HEADSET_MIC_PIN2, + ALC897_FIXUP_UNIS_H3C_X500S, }; static const struct hda_fixup alc662_fixups[] = { @@ -10316,6 +10317,13 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MODE }, + [ALC897_FIXUP_UNIS_H3C_X500S] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x14, AC_VERB_SET_EAPD_BTLENABLE, 0 }, + {} + }, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -10477,6 +10485,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"}, {.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC669_FIXUP_ACER_ASPIRE_ETHOS, .name = "aspire-ethos"}, + {.id = ALC897_FIXUP_UNIS_H3C_X500S, .name = "unis-h3c-x500s"}, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 5.10 15/16] ALSA: hda: fix a possible null-pointer dereference due to data race in snd_hdac_regmap_sync()
by Sasha Levin 24 Jul '23

24 Jul '23

From: Tuo Li <islituo(a)gmail.com> [ Upstream commit 1f4a08fed450db87fbb5ff5105354158bdbe1a22 ] The variable codec->regmap is often protected by the lock codec->regmap_lock when is accessed. However, it is accessed without holding the lock when is accessed in snd_hdac_regmap_sync(): if (codec->regmap) In my opinion, this may be a harmful race, because if codec->regmap is set to NULL right after the condition is checked, a null-pointer dereference can occur in the called function regcache_sync(): map->lock(map->lock_arg); --> Line 360 in drivers/base/regmap/regcache.c To fix this possible null-pointer dereference caused by data race, the mutex_lock coverage is extended to protect the if statement as well as the function call to regcache_sync(). [ Note: the lack of the regmap_lock itself is harmless for the current codec driver implementations, as snd_hdac_regmap_sync() is only for PM runtime resume that is prohibited during the codec probe. But the change makes the whole code more consistent, so it's merged as is -- tiwai ] Reported-by: BassCheck <bass(a)buaa.edu.cn> Signed-off-by: Tuo Li <islituo(a)gmail.com> Link: https://lore.kernel.org/r/20230703031016.1184711-1-islituo@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_regmap.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index d75f31eb9d78f..bf35acca5ea0e 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -597,10 +597,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw_once); */ void snd_hdac_regmap_sync(struct hdac_device *codec) { - if (codec->regmap) { - mutex_lock(&codec->regmap_lock); + mutex_lock(&codec->regmap_lock); + if (codec->regmap) regcache_sync(codec->regmap); - mutex_unlock(&codec->regmap_lock); - } + mutex_unlock(&codec->regmap_lock); } EXPORT_SYMBOL_GPL(snd_hdac_regmap_sync); -- 2.39.2

1 0

[PATCH AUTOSEL 5.10 14/16] ALSA: hda/realtek: Add quirks for Unis H3C Desktop B760 & Q760
by Sasha Levin 24 Jul '23

24 Jul '23

From: dengxiang <dengxiang(a)nfschina.com> [ Upstream commit 73f1c75d5e6bd8ce2a887ef493a66ad1b16ed704 ] These models use NSIWAY amplifiers for internal speaker, but cannot put sound outside from these amplifiers. So eapd verbs are needed to initialize the amplifiers. They can be added during boot to get working sound out of internal speaker. Signed-off-by: dengxiang <dengxiang(a)nfschina.com> Link: https://lore.kernel.org/r/20230703021751.2945750-1-dengxiang@nfschina.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7d6a36b647a73..e2666919842d2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10694,6 +10694,7 @@ enum { ALC897_FIXUP_HP_HSMIC_VERB, ALC897_FIXUP_LENOVO_HEADSET_MODE, ALC897_FIXUP_HEADSET_MIC_PIN2, + ALC897_FIXUP_UNIS_H3C_X500S, }; static const struct hda_fixup alc662_fixups[] = { @@ -11133,6 +11134,13 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MODE }, + [ALC897_FIXUP_UNIS_H3C_X500S] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x14, AC_VERB_SET_EAPD_BTLENABLE, 0 }, + {} + }, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -11294,6 +11302,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"}, {.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC669_FIXUP_ACER_ASPIRE_ETHOS, .name = "aspire-ethos"}, + {.id = ALC897_FIXUP_UNIS_H3C_X500S, .name = "unis-h3c-x500s"}, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 23/24] ALSA: hda: fix a possible null-pointer dereference due to data race in snd_hdac_regmap_sync()
by Sasha Levin 24 Jul '23

24 Jul '23

From: Tuo Li <islituo(a)gmail.com> [ Upstream commit 1f4a08fed450db87fbb5ff5105354158bdbe1a22 ] The variable codec->regmap is often protected by the lock codec->regmap_lock when is accessed. However, it is accessed without holding the lock when is accessed in snd_hdac_regmap_sync(): if (codec->regmap) In my opinion, this may be a harmful race, because if codec->regmap is set to NULL right after the condition is checked, a null-pointer dereference can occur in the called function regcache_sync(): map->lock(map->lock_arg); --> Line 360 in drivers/base/regmap/regcache.c To fix this possible null-pointer dereference caused by data race, the mutex_lock coverage is extended to protect the if statement as well as the function call to regcache_sync(). [ Note: the lack of the regmap_lock itself is harmless for the current codec driver implementations, as snd_hdac_regmap_sync() is only for PM runtime resume that is prohibited during the codec probe. But the change makes the whole code more consistent, so it's merged as is -- tiwai ] Reported-by: BassCheck <bass(a)buaa.edu.cn> Signed-off-by: Tuo Li <islituo(a)gmail.com> Link: https://lore.kernel.org/r/20230703031016.1184711-1-islituo@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_regmap.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index fe3587547cfec..39610a15bcc98 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -597,10 +597,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw_once); */ void snd_hdac_regmap_sync(struct hdac_device *codec) { - if (codec->regmap) { - mutex_lock(&codec->regmap_lock); + mutex_lock(&codec->regmap_lock); + if (codec->regmap) regcache_sync(codec->regmap); - mutex_unlock(&codec->regmap_lock); - } + mutex_unlock(&codec->regmap_lock); } EXPORT_SYMBOL_GPL(snd_hdac_regmap_sync); -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 22/24] ALSA: hda/realtek: Add quirks for Unis H3C Desktop B760 & Q760
by Sasha Levin 24 Jul '23

24 Jul '23

From: dengxiang <dengxiang(a)nfschina.com> [ Upstream commit 73f1c75d5e6bd8ce2a887ef493a66ad1b16ed704 ] These models use NSIWAY amplifiers for internal speaker, but cannot put sound outside from these amplifiers. So eapd verbs are needed to initialize the amplifiers. They can be added during boot to get working sound out of internal speaker. Signed-off-by: dengxiang <dengxiang(a)nfschina.com> Link: https://lore.kernel.org/r/20230703021751.2945750-1-dengxiang@nfschina.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 22f0d1b359f0f..3937ab8464541 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10782,6 +10782,7 @@ enum { ALC897_FIXUP_HP_HSMIC_VERB, ALC897_FIXUP_LENOVO_HEADSET_MODE, ALC897_FIXUP_HEADSET_MIC_PIN2, + ALC897_FIXUP_UNIS_H3C_X500S, }; static const struct hda_fixup alc662_fixups[] = { @@ -11221,6 +11222,13 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MODE }, + [ALC897_FIXUP_UNIS_H3C_X500S] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x14, AC_VERB_SET_EAPD_BTLENABLE, 0 }, + {} + }, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -11382,6 +11390,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"}, {.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC669_FIXUP_ACER_ASPIRE_ETHOS, .name = "aspire-ethos"}, + {.id = ALC897_FIXUP_UNIS_H3C_X500S, .name = "unis-h3c-x500s"}, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 34/34] ALSA: hda/realtek: Add quirk for ASUS ROG GZ301V
by Sasha Levin 24 Jul '23

24 Jul '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 5251605f4d297a0eb5d3b7f39f9dcee9e4d0115a ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG GZ301V series which uses an SPI connected Cirrus amp. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230706223323.30871-2-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cad413b4e26e0..7a5ce1db75cf3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9548,6 +9548,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), + SND_PCI_QUIRK(0x1043, 0x1573, "ASUS GZ301V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 32/34] ALSA: hda/realtek: Add quirk for ASUS ROG GA402X
by Sasha Levin 24 Jul '23

24 Jul '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 9abc77fb144fe916fd2f592dc4b8c7bade02e58a ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG GA402X series which uses an I2C connected Cirrus amp. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230704044619.19343-3-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5cb843d56cd16..cad413b4e26e0 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9544,6 +9544,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650P", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1463, "Asus GA402X", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 31/34] ALSA: hda/realtek: Add quirk for ASUS ROG GX650P
by Sasha Levin 24 Jul '23

24 Jul '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 8cc87c055d28320e5fa5457922f43bc07dec58bd ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG GV601V series which uses an I2C connected Cirrus amp. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230704044619.19343-2-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce560129e2faa..5cb843d56cd16 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7065,6 +7065,8 @@ enum { ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1, ALC285_FIXUP_ASUS_HEADSET_MIC, + ALC285_FIXUP_ASUS_I2C_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_I2C_HEADSET_MIC, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -8051,6 +8053,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1 }, + [ALC285_FIXUP_ASUS_I2C_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC287_FIXUP_CS35L41_I2C_2 + }, + [ALC285_FIXUP_ASUS_I2C_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1b, 0x03a11c30 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_I2C_SPEAKER2_TO_DAC1 + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9525,6 +9543,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), + SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650P", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 30/34] ALSA: hda: fix a possible null-pointer dereference due to data race in snd_hdac_regmap_sync()
by Sasha Levin 24 Jul '23

24 Jul '23

From: Tuo Li <islituo(a)gmail.com> [ Upstream commit 1f4a08fed450db87fbb5ff5105354158bdbe1a22 ] The variable codec->regmap is often protected by the lock codec->regmap_lock when is accessed. However, it is accessed without holding the lock when is accessed in snd_hdac_regmap_sync(): if (codec->regmap) In my opinion, this may be a harmful race, because if codec->regmap is set to NULL right after the condition is checked, a null-pointer dereference can occur in the called function regcache_sync(): map->lock(map->lock_arg); --> Line 360 in drivers/base/regmap/regcache.c To fix this possible null-pointer dereference caused by data race, the mutex_lock coverage is extended to protect the if statement as well as the function call to regcache_sync(). [ Note: the lack of the regmap_lock itself is harmless for the current codec driver implementations, as snd_hdac_regmap_sync() is only for PM runtime resume that is prohibited during the codec probe. But the change makes the whole code more consistent, so it's merged as is -- tiwai ] Reported-by: BassCheck <bass(a)buaa.edu.cn> Signed-off-by: Tuo Li <islituo(a)gmail.com> Link: https://lore.kernel.org/r/20230703031016.1184711-1-islituo@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_regmap.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index fe3587547cfec..39610a15bcc98 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -597,10 +597,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw_once); */ void snd_hdac_regmap_sync(struct hdac_device *codec) { - if (codec->regmap) { - mutex_lock(&codec->regmap_lock); + mutex_lock(&codec->regmap_lock); + if (codec->regmap) regcache_sync(codec->regmap); - mutex_unlock(&codec->regmap_lock); - } + mutex_unlock(&codec->regmap_lock); } EXPORT_SYMBOL_GPL(snd_hdac_regmap_sync); -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 29/34] ALSA: hda/realtek: Add quirks for Unis H3C Desktop B760 & Q760
by Sasha Levin 24 Jul '23

24 Jul '23

From: dengxiang <dengxiang(a)nfschina.com> [ Upstream commit 73f1c75d5e6bd8ce2a887ef493a66ad1b16ed704 ] These models use NSIWAY amplifiers for internal speaker, but cannot put sound outside from these amplifiers. So eapd verbs are needed to initialize the amplifiers. They can be added during boot to get working sound out of internal speaker. Signed-off-by: dengxiang <dengxiang(a)nfschina.com> Link: https://lore.kernel.org/r/20230703021751.2945750-1-dengxiang@nfschina.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index eb049014f87ac..ce560129e2faa 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11235,6 +11235,7 @@ enum { ALC897_FIXUP_HP_HSMIC_VERB, ALC897_FIXUP_LENOVO_HEADSET_MODE, ALC897_FIXUP_HEADSET_MIC_PIN2, + ALC897_FIXUP_UNIS_H3C_X500S, }; static const struct hda_fixup alc662_fixups[] = { @@ -11674,6 +11675,13 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MODE }, + [ALC897_FIXUP_UNIS_H3C_X500S] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x14, AC_VERB_SET_EAPD_BTLENABLE, 0 }, + {} + }, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -11835,6 +11843,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"}, {.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC669_FIXUP_ACER_ASPIRE_ETHOS, .name = "aspire-ethos"}, + {.id = ALC897_FIXUP_UNIS_H3C_X500S, .name = "unis-h3c-x500s"}, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 40/40] ALSA: hda/realtek: Add quirk for ASUS ROG GZ301V
by Sasha Levin 24 Jul '23

24 Jul '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 5251605f4d297a0eb5d3b7f39f9dcee9e4d0115a ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG GZ301V series which uses an SPI connected Cirrus amp. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230706223323.30871-2-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7534772539316..f7b5765db50d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9559,6 +9559,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), + SND_PCI_QUIRK(0x1043, 0x1573, "ASUS GZ301V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1662, "ASUS GV301QH", ALC294_FIXUP_ASUS_DUAL_SPK), SND_PCI_QUIRK(0x1043, 0x1683, "ASUS UM3402YAR", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x16b2, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 38/40] ALSA: hda/realtek: Add quirk for ASUS ROG G614Jx
by Sasha Levin 24 Jul '23

24 Jul '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 33d7c9c3bf70ed91191a2bedbbc03783b824b5de ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG G614J series which uses an SPI connected Cirrus amp. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230704044619.19343-5-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e49390b714f61..7534772539316 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9583,6 +9583,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), + SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JI", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 37/40] ALSA: hda/realtek: Amend G634 quirk to enable rear speakers
by Sasha Levin 24 Jul '23

24 Jul '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit b759a5f097cd42c666f1ebca8da50ff507435fbe ] Amends the last quirk for the G634 with 0x1caf subsys to enable the rear speakers via pincfg. Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230704044619.19343-4-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 20f29d59b242b..e49390b714f61 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7065,6 +7065,7 @@ enum { ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1, ALC285_FIXUP_ASUS_HEADSET_MIC, + ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS, ALC285_FIXUP_ASUS_I2C_SPEAKER2_TO_DAC1, ALC285_FIXUP_ASUS_I2C_HEADSET_MIC, ALC280_FIXUP_HP_HEADSET_MIC, @@ -8053,6 +8054,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1 }, + [ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x14, 0x90170120 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_HEADSET_MIC + }, [ALC285_FIXUP_ASUS_I2C_SPEAKER2_TO_DAC1] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_speaker2_to_dac1, @@ -9573,7 +9583,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c23, "Asus X55U", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), - SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 36/40] ALSA: hda/realtek: Add quirk for ASUS ROG GA402X
by Sasha Levin 24 Jul '23

24 Jul '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 9abc77fb144fe916fd2f592dc4b8c7bade02e58a ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG GA402X series which uses an I2C connected Cirrus amp. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230704044619.19343-3-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cda8415dab9fb..20f29d59b242b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9544,6 +9544,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650P", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), + SND_PCI_QUIRK(0x1043, 0x1463, "Asus GA402X", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601V", ALC285_FIXUP_ASUS_HEADSET_MIC), -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 35/40] ALSA: hda/realtek: Add quirk for ASUS ROG GX650P
by Sasha Levin 24 Jul '23

24 Jul '23

From: "Luke D. Jones" <luke(a)ljones.dev> [ Upstream commit 8cc87c055d28320e5fa5457922f43bc07dec58bd ] Adds the required quirk to enable the Cirrus amp and correct pins on the ASUS ROG GV601V series which uses an I2C connected Cirrus amp. While this works if the related _DSD properties are made available, these aren't included in the ACPI of these laptops (yet). Signed-off-by: Luke D. Jones <luke(a)ljones.dev> Link: https://lore.kernel.org/r/20230704044619.19343-2-luke@ljones.dev Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7a8f8aef0f7e5..cda8415dab9fb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7065,6 +7065,8 @@ enum { ALC285_FIXUP_SPEAKER2_TO_DAC1, ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1, ALC285_FIXUP_ASUS_HEADSET_MIC, + ALC285_FIXUP_ASUS_I2C_SPEAKER2_TO_DAC1, + ALC285_FIXUP_ASUS_I2C_HEADSET_MIC, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -8051,6 +8053,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_ASUS_SPEAKER2_TO_DAC1 }, + [ALC285_FIXUP_ASUS_I2C_SPEAKER2_TO_DAC1] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC287_FIXUP_CS35L41_I2C_2 + }, + [ALC285_FIXUP_ASUS_I2C_HEADSET_MIC] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1b, 0x03a11c30 }, + { } + }, + .chained = true, + .chain_id = ALC285_FIXUP_ASUS_I2C_SPEAKER2_TO_DAC1 + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -9525,6 +9543,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1313, "Asus K42JZ", ALC269VB_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), + SND_PCI_QUIRK(0x1043, 0x1433, "ASUS GX650P", ALC285_FIXUP_ASUS_I2C_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1473, "ASUS GU604V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1483, "ASUS GU603V", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1493, "ASUS GV601V", ALC285_FIXUP_ASUS_HEADSET_MIC), -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 34/40] ALSA: hda: fix a possible null-pointer dereference due to data race in snd_hdac_regmap_sync()
by Sasha Levin 24 Jul '23

24 Jul '23

From: Tuo Li <islituo(a)gmail.com> [ Upstream commit 1f4a08fed450db87fbb5ff5105354158bdbe1a22 ] The variable codec->regmap is often protected by the lock codec->regmap_lock when is accessed. However, it is accessed without holding the lock when is accessed in snd_hdac_regmap_sync(): if (codec->regmap) In my opinion, this may be a harmful race, because if codec->regmap is set to NULL right after the condition is checked, a null-pointer dereference can occur in the called function regcache_sync(): map->lock(map->lock_arg); --> Line 360 in drivers/base/regmap/regcache.c To fix this possible null-pointer dereference caused by data race, the mutex_lock coverage is extended to protect the if statement as well as the function call to regcache_sync(). [ Note: the lack of the regmap_lock itself is harmless for the current codec driver implementations, as snd_hdac_regmap_sync() is only for PM runtime resume that is prohibited during the codec probe. But the change makes the whole code more consistent, so it's merged as is -- tiwai ] Reported-by: BassCheck <bass(a)buaa.edu.cn> Signed-off-by: Tuo Li <islituo(a)gmail.com> Link: https://lore.kernel.org/r/20230703031016.1184711-1-islituo@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/hdac_regmap.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c index fe3587547cfec..39610a15bcc98 100644 --- a/sound/hda/hdac_regmap.c +++ b/sound/hda/hdac_regmap.c @@ -597,10 +597,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_regmap_update_raw_once); */ void snd_hdac_regmap_sync(struct hdac_device *codec) { - if (codec->regmap) { - mutex_lock(&codec->regmap_lock); + mutex_lock(&codec->regmap_lock); + if (codec->regmap) regcache_sync(codec->regmap); - mutex_unlock(&codec->regmap_lock); - } + mutex_unlock(&codec->regmap_lock); } EXPORT_SYMBOL_GPL(snd_hdac_regmap_sync); -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 33/40] ALSA: hda/realtek: Add quirks for Unis H3C Desktop B760 & Q760
by Sasha Levin 24 Jul '23

24 Jul '23

From: dengxiang <dengxiang(a)nfschina.com> [ Upstream commit 73f1c75d5e6bd8ce2a887ef493a66ad1b16ed704 ] These models use NSIWAY amplifiers for internal speaker, but cannot put sound outside from these amplifiers. So eapd verbs are needed to initialize the amplifiers. They can be added during boot to get working sound out of internal speaker. Signed-off-by: dengxiang <dengxiang(a)nfschina.com> Link: https://lore.kernel.org/r/20230703021751.2945750-1-dengxiang@nfschina.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dabfdecece264..7a8f8aef0f7e5 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11237,6 +11237,7 @@ enum { ALC897_FIXUP_HP_HSMIC_VERB, ALC897_FIXUP_LENOVO_HEADSET_MODE, ALC897_FIXUP_HEADSET_MIC_PIN2, + ALC897_FIXUP_UNIS_H3C_X500S, }; static const struct hda_fixup alc662_fixups[] = { @@ -11676,6 +11677,13 @@ static const struct hda_fixup alc662_fixups[] = { .chained = true, .chain_id = ALC897_FIXUP_LENOVO_HEADSET_MODE }, + [ALC897_FIXUP_UNIS_H3C_X500S] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x14, AC_VERB_SET_EAPD_BTLENABLE, 0 }, + {} + }, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -11837,6 +11845,7 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_USI_HEADSET_MODE, .name = "usi-headset"}, {.id = ALC662_FIXUP_LENOVO_MULTI_CODECS, .name = "dual-codecs"}, {.id = ALC669_FIXUP_ACER_ASPIRE_ETHOS, .name = "aspire-ethos"}, + {.id = ALC897_FIXUP_UNIS_H3C_X500S, .name = "unis-h3c-x500s"}, {} }; -- 2.39.2

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[PATCH AUTOSEL 5.4 02/12] ALSA: emu10k1: roll up loops in DSP setup code for Audigy
by Sasha Levin 24 Jul '23

24 Jul '23

From: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> [ Upstream commit 8cabf83c7aa54530e699be56249fb44f9505c4f3 ] There is no apparent reason for the massive code duplication. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> Link: https://lore.kernel.org/r/20230510173917.3073107-3-oswald.buddenhagen@gmx.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/emu10k1/emufx.c | 112 +++----------------------------------- 1 file changed, 9 insertions(+), 103 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index e053f0d58bdd0..2f3cfcfcdb9a3 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1536,14 +1536,8 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr += 2; /* Master volume (will be renamed later) */ - A_OP(icode, &ptr, iMAC0, A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS)); + for (z = 0; z < 8; z++) + A_OP(icode, &ptr, iMAC0, A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS)); snd_emu10k1_init_mono_control(&controls[nctl++], "Wave Master Playback Volume", gpr, 0); gpr += 2; @@ -1627,102 +1621,14 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) dev_dbg(emu->card->dev, "emufx.c: gpr=0x%x, tmp=0x%x\n", gpr, tmp); */ - /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ - /* A_P16VIN(0) is delayed by one sample, - * so all other A_P16VIN channels will need to also be delayed - */ - /* Left ADC in. 1 of 2 */ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); - /* Right ADC in 1 of 2 */ - gpr_map[gpr++] = 0x00000000; - /* Delaying by one sample: instead of copying the input - * value A_P16VIN to output A_FXBUS2 as in the first channel, - * we use an auxiliary register, delaying the value by one - * sample - */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); - /* For 96kHz mode */ - /* Left ADC in. 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); - /* Right ADC in 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); - /* Pavel Hofman - we still have voices, A_FXBUS2s, and - * A_P16VINs available - - * let's add 8 more capture channels - total of 16 - */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x10)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x12)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x14)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x16)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x18)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1a)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1c)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1e)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), - A_C_00000000, A_C_00000000); + /* A_P16VIN(0) is delayed by one sample, so all other A_P16VIN channels + * will need to also be delayed; we use an auxiliary register for that. */ + for (z = 1; z < 0x10; z++) { + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr), A_FXBUS2(z * 2) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr), A_P16VIN(z), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + } } #if 0 -- 2.39.2

1 0

[PATCH AUTOSEL 5.10 11/22] ALSA: hda: Add Loongson LS7A HD-Audio support
by Sasha Levin 24 Jul '23

24 Jul '23

From: Yanteng Si <siyanteng(a)loongson.cn> [ Upstream commit 28bd137a3c8e105587ba8c55b68ef43b519b270f ] Add the new PCI ID 0x0014 0x7a07 and the new PCI ID 0x0014 0x7a37 Loongson HDA controller. Signed-off-by: Yanteng Si <siyanteng(a)loongson.cn> Acked-by: Huacai Chen <chenhuacai(a)loongson.cn> Link: https://lore.kernel.org/r/993587483b9509796b29a416f257fcfb4b15c6ea.16861288… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/linux/pci_ids.h | 3 +++ sound/hda/hdac_device.c | 1 + sound/pci/hda/hda_intel.c | 7 +++++++ sound/pci/hda/patch_hdmi.c | 1 + 4 files changed, 12 insertions(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index 4b34a5c125999..84cb7b32aa26f 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -153,6 +153,9 @@ #define PCI_VENDOR_ID_LOONGSON 0x0014 +#define PCI_DEVICE_ID_LOONGSON_HDA 0x7a07 +#define PCI_DEVICE_ID_LOONGSON_HDMI 0x7a37 + #define PCI_VENDOR_ID_TTTECH 0x0357 #define PCI_DEVICE_ID_TTTECH_MC322 0x000a diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index bfd8585776767..5315a3eb114ee 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -645,6 +645,7 @@ struct hda_vendor_id { }; static const struct hda_vendor_id hda_vendor_ids[] = { + { 0x0014, "Loongson" }, { 0x1002, "ATI" }, { 0x1013, "Cirrus Logic" }, { 0x1057, "Motorola" }, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1f641712233ef..13226f4f2d119 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -274,6 +274,7 @@ enum { AZX_DRIVER_CTHDA, AZX_DRIVER_CMEDIA, AZX_DRIVER_ZHAOXIN, + AZX_DRIVER_LOONGSON, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -397,6 +398,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_ZHAOXIN] = "HDA Zhaoxin", + [AZX_DRIVER_LOONGSON] = "HDA Loongson", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -2838,6 +2840,11 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI }, /* Zhaoxin */ { PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN }, + /* Loongson HDAudio*/ + {PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, PCI_DEVICE_ID_LOONGSON_HDA), + .driver_data = AZX_DRIVER_LOONGSON }, + {PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, PCI_DEVICE_ID_LOONGSON_HDMI), + .driver_data = AZX_DRIVER_LOONGSON }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index c19afe4861949..5ff7e93024154 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4307,6 +4307,7 @@ static int patch_gf_hdmi(struct hda_codec *codec) * patch entries */ static const struct hda_device_id snd_hda_id_hdmi[] = { +HDA_CODEC_ENTRY(0x00147a47, "Loongson HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x1002793c, "RS600 HDMI", patch_atihdmi), HDA_CODEC_ENTRY(0x10027919, "RS600 HDMI", patch_atihdmi), HDA_CODEC_ENTRY(0x1002791a, "RS690/780 HDMI", patch_atihdmi), -- 2.39.2

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[PATCH AUTOSEL 5.10 07/22] ASoC: Intel: sof_sdw: Add support for Rex soundwire
by Sasha Levin 24 Jul '23

24 Jul '23

From: Uday M Bhat <uday.m.bhat(a)intel.com> [ Upstream commit 164e5dc17525181c05563f0a06796f1a363801d5 ] Add rex entry in the soundwire quirk table Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Uday M Bhat <uday.m.bhat(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230602202225.249209-28-pierre-louis.bossart@lin… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 30cc8bbceb79b..cbbb50ddc7954 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -207,6 +207,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2_100K), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Rex"), + }, + .driver_data = (void *)(SOF_SDW_PCH_DMIC), + }, /* LunarLake devices */ { .callback = sof_sdw_quirk_cb, -- 2.39.2

1 0

[PATCH AUTOSEL 5.10 04/22] ASoC: Intel: sof_sdw: add quirk for LNL RVP
by Sasha Levin 24 Jul '23

24 Jul '23

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit dfe25fea968dc4884e12d471c8263f0f611b380a ] We should use RT711_JD2_100K for on board rt711 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Link: https://lore.kernel.org/r/20230512173305.65399-9-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 91a7cb33042d8..30cc8bbceb79b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -207,6 +207,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2_100K), }, + /* LunarLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Lunar Lake Client Platform"), + }, + .driver_data = (void *)(RT711_JD2_100K), + }, {} }; -- 2.39.2

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[PATCH AUTOSEL 5.10 03/22] ASoC: Intel: sof_sdw: add quirk for MTL RVP
by Sasha Levin 24 Jul '23

24 Jul '23

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 289e1df00e49a229a1c924c059242e759a552f01 ] We should use RT711_JD2_100K for on board rt711. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Link: https://lore.kernel.org/r/20230512173305.65399-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index f5d8f7951cfc3..91a7cb33042d8 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -199,6 +199,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_RT715_DAI_ID_FIX | SOF_SDW_PCH_DMIC), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Meteor Lake Client Platform"), + }, + .driver_data = (void *)(RT711_JD2_100K), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 5.10 02/22] ALSA: emu10k1: roll up loops in DSP setup code for Audigy
by Sasha Levin 24 Jul '23

24 Jul '23

From: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> [ Upstream commit 8cabf83c7aa54530e699be56249fb44f9505c4f3 ] There is no apparent reason for the massive code duplication. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> Link: https://lore.kernel.org/r/20230510173917.3073107-3-oswald.buddenhagen@gmx.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/emu10k1/emufx.c | 112 +++----------------------------------- 1 file changed, 9 insertions(+), 103 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 4e76ed0e91d5d..e17b93b25d2ff 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1560,14 +1560,8 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr += 2; /* Master volume (will be renamed later) */ - A_OP(icode, &ptr, iMAC0, A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS)); + for (z = 0; z < 8; z++) + A_OP(icode, &ptr, iMAC0, A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS)); snd_emu10k1_init_mono_control(&controls[nctl++], "Wave Master Playback Volume", gpr, 0); gpr += 2; @@ -1651,102 +1645,14 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) dev_dbg(emu->card->dev, "emufx.c: gpr=0x%x, tmp=0x%x\n", gpr, tmp); */ - /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ - /* A_P16VIN(0) is delayed by one sample, - * so all other A_P16VIN channels will need to also be delayed - */ - /* Left ADC in. 1 of 2 */ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); - /* Right ADC in 1 of 2 */ - gpr_map[gpr++] = 0x00000000; - /* Delaying by one sample: instead of copying the input - * value A_P16VIN to output A_FXBUS2 as in the first channel, - * we use an auxiliary register, delaying the value by one - * sample - */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); - /* For 96kHz mode */ - /* Left ADC in. 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); - /* Right ADC in 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); - /* Pavel Hofman - we still have voices, A_FXBUS2s, and - * A_P16VINs available - - * let's add 8 more capture channels - total of 16 - */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x10)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x12)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x14)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x16)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x18)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1a)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1c)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1e)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), - A_C_00000000, A_C_00000000); + /* A_P16VIN(0) is delayed by one sample, so all other A_P16VIN channels + * will need to also be delayed; we use an auxiliary register for that. */ + for (z = 1; z < 0x10; z++) { + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr), A_FXBUS2(z * 2) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr), A_P16VIN(z), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + } } #if 0 -- 2.39.2

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[PATCH AUTOSEL 5.15 19/23] ALSA: hda/realtek: Add quirks for ROG ALLY CS35l41 audio
by Sasha Levin 24 Jul '23

24 Jul '23

From: Matthew Anderson <ruinairas1992(a)gmail.com> [ Upstream commit 724418b84e6248cd27599607b7e5fac365b8e3f5 ] This requires a patched ACPI table or a firmware from ASUS to work because the system does not come with the _DSD field for the CSC3551. Link: https://bugzilla.kernel.org/show_bug.cgi?id=217550 Signed-off-by: Matthew Anderson <ruinairas1992(a)gmail.com> Tested-by: Philip Mueller <philm(a)manjaro.org> Link: https://lore.kernel.org/r/20230621161714.9442-1-ruinairas1992@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 22f0d1b359f0f..42a1446a43eed 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6921,6 +6921,10 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, + ALC294_FIXUP_ASUS_ALLY, + ALC294_FIXUP_ASUS_ALLY_PINS, + ALC294_FIXUP_ASUS_ALLY_VERBS, + ALC294_FIXUP_ASUS_ALLY_SPEAKER, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, ALC294_FIXUP_ASUS_GX502_HP, @@ -8183,6 +8187,47 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, + [ALC294_FIXUP_ASUS_ALLY] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_i2c_two, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS + }, + [ALC294_FIXUP_ASUS_ALLY_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1a, 0x03a11c30 }, + { 0x21, 0x03211420 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_VERBS + }, + [ALC294_FIXUP_ASUS_ALLY_VERBS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x46 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0004 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x47 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xa47a }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x49 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0049}, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x4a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x201b }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x6b }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x4278}, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_SPEAKER + }, + [ALC294_FIXUP_ASUS_ALLY_SPEAKER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + }, [ALC285_FIXUP_THINKPAD_X1_GEN7] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_thinkpad_x1_gen7, @@ -9107,6 +9152,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), + SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally RC71L_RC71L", ALC294_FIXUP_ASUS_ALLY), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 16/23] ASoC: SOF: core: Free the firmware trace before calling snd_sof_shutdown()
by Sasha Levin 24 Jul '23

24 Jul '23

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit d389dcb3a48cec4f03c16434c0bf98a4c635372a ] The shutdown is called on reboot/shutdown of the machine. At this point the firmware tracing cannot be used anymore but in case of IPC3 it is using and keeping a DMA channel active (dtrace). For Tiger Lake platforms we have a quirk in place to fix rare reboot issues when a DMA was active before rebooting the system. If the tracing is enabled this quirk will be always used and a print appears on the kernel log which might be misleading or not even correct. Release the fw tracing before executing the shutdown to make sure that this known DMA user is cleared away. Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230616100039.378150-4-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 59d0d7b2b55c8..5f8e89da7b578 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -406,8 +406,10 @@ int snd_sof_device_shutdown(struct device *dev) if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); - if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) + if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) { + sof_fw_trace_free(sdev); return snd_sof_shutdown(sdev); + } return 0; } -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 10/23] ASoC: SOF: Intel: fix SoundWire/HDaudio mutual exclusion
by Sasha Levin 24 Jul '23

24 Jul '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit f751b99255cacd9ffe8c4bbf99767ad670cee1f7 ] The functionality described in Commit 61bef9e68dca ("ASoC: SOF: Intel: hda: enforce exclusion between HDaudio and SoundWire") does not seem to be properly implemented with two issues that need to be corrected. a) The test used is incorrect when DisplayAudio codecs are not supported. b) Conversely when only Display Audio codecs can be found, we do want to start the SoundWire links, if any. That will help add the relevant topologies and machine descriptors, and identify cases where the SoundWire information in ACPI needs to be modified with a quirk. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20230606222529.57156-2-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 35cbef171f4a3..038d09f6203aa 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1040,12 +1040,22 @@ static int hda_generic_machine_select(struct snd_sof_dev *sdev) pdata->machine = hda_mach; pdata->tplg_filename = tplg_filename; - if (codec_num == 2) { + if (codec_num == 2 || + (codec_num == 1 && !HDA_IDISP_CODEC(bus->codec_mask))) { /* * Prevent SoundWire links from starting when an external * HDaudio codec is used */ hda_mach->mach_params.link_mask = 0; + } else { + /* + * Allow SoundWire links to start when no external HDaudio codec + * was detected. This will not create a SoundWire card but + * will help detect if any SoundWire codec reports as ATTACHED. + */ + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; + + hda_mach->mach_params.link_mask = hdev->info.link_mask; } } } -- 2.39.2

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[PATCH AUTOSEL 5.15 09/23] ALSA: hda: Add Loongson LS7A HD-Audio support
by Sasha Levin 24 Jul '23

24 Jul '23

From: Yanteng Si <siyanteng(a)loongson.cn> [ Upstream commit 28bd137a3c8e105587ba8c55b68ef43b519b270f ] Add the new PCI ID 0x0014 0x7a07 and the new PCI ID 0x0014 0x7a37 Loongson HDA controller. Signed-off-by: Yanteng Si <siyanteng(a)loongson.cn> Acked-by: Huacai Chen <chenhuacai(a)loongson.cn> Link: https://lore.kernel.org/r/993587483b9509796b29a416f257fcfb4b15c6ea.16861288… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/linux/pci_ids.h | 3 +++ sound/hda/hdac_device.c | 1 + sound/pci/hda/hda_intel.c | 7 +++++++ sound/pci/hda/patch_hdmi.c | 1 + 4 files changed, 12 insertions(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index 8a1e264735668..b77bfd88d6c51 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -154,6 +154,9 @@ #define PCI_VENDOR_ID_LOONGSON 0x0014 +#define PCI_DEVICE_ID_LOONGSON_HDA 0x7a07 +#define PCI_DEVICE_ID_LOONGSON_HDMI 0x7a37 + #define PCI_VENDOR_ID_TTTECH 0x0357 #define PCI_DEVICE_ID_TTTECH_MC322 0x000a diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index bfd8585776767..5315a3eb114ee 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -645,6 +645,7 @@ struct hda_vendor_id { }; static const struct hda_vendor_id hda_vendor_ids[] = { + { 0x0014, "Loongson" }, { 0x1002, "ATI" }, { 0x1013, "Cirrus Logic" }, { 0x1057, "Motorola" }, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 1379ac07df350..29b1b2f278baa 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -240,6 +240,7 @@ enum { AZX_DRIVER_CTHDA, AZX_DRIVER_CMEDIA, AZX_DRIVER_ZHAOXIN, + AZX_DRIVER_LOONGSON, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -363,6 +364,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_ZHAOXIN] = "HDA Zhaoxin", + [AZX_DRIVER_LOONGSON] = "HDA Loongson", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -2776,6 +2778,11 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI }, /* Zhaoxin */ { PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN }, + /* Loongson HDAudio*/ + {PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, PCI_DEVICE_ID_LOONGSON_HDA), + .driver_data = AZX_DRIVER_LOONGSON }, + {PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, PCI_DEVICE_ID_LOONGSON_HDMI), + .driver_data = AZX_DRIVER_LOONGSON }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3cd3b5c49e45e..e777c56ab959a 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4318,6 +4318,7 @@ static int patch_gf_hdmi(struct hda_codec *codec) * patch entries */ static const struct hda_device_id snd_hda_id_hdmi[] = { +HDA_CODEC_ENTRY(0x00147a47, "Loongson HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x1002793c, "RS600 HDMI", patch_atihdmi), HDA_CODEC_ENTRY(0x10027919, "RS600 HDMI", patch_atihdmi), HDA_CODEC_ENTRY(0x1002791a, "RS690/780 HDMI", patch_atihdmi), -- 2.39.2

1 0

[PATCH AUTOSEL 5.15 04/23] ASoC: amd: vangogh: Add check for acp config flags in vangogh platform
by Sasha Levin 24 Jul '23

24 Jul '23

From: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> [ Upstream commit e89f45edb747ed88e97a5771dd6d3dd1eb517873 ] We have SOF and generic ACP support enabled for Vangogh platform on some machines. Since we have same PCI id used for probing, add check for machine configuration flag to avoid conflict with newer pci drivers. Such machine flag has been initialized via dmi match on few Vangogh based machines. If no flag is specified probe and register older platform device. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Link: https://lore.kernel.org/r/20230530110802.674939-1-venkataprasad.potturu@amd… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/vangogh/acp5x.h | 2 ++ sound/soc/amd/vangogh/pci-acp5x.c | 7 ++++++- 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/vangogh/acp5x.h b/sound/soc/amd/vangogh/acp5x.h index a808635f97400..ac9827393c453 100644 --- a/sound/soc/amd/vangogh/acp5x.h +++ b/sound/soc/amd/vangogh/acp5x.h @@ -145,6 +145,8 @@ static inline void acp_writel(u32 val, void __iomem *base_addr) writel(val, base_addr - ACP5x_PHY_BASE_ADDRESS); } +int snd_amd_acp_find_config(struct pci_dev *pci); + static inline u64 acp_get_byte_count(struct i2s_stream_instance *rtd, int direction) { diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index a57b762d9f2e1..1b4bf357f2015 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -121,10 +121,15 @@ static int snd_acp5x_probe(struct pci_dev *pci, { struct acp5x_dev_data *adata; struct platform_device_info pdevinfo[ACP5x_DEVS]; - unsigned int irqflags; + unsigned int irqflags, flag; int ret, i; u32 addr, val; + /* Return if acp config flag is defined */ + flag = snd_amd_acp_find_config(pci); + if (flag) + return -ENODEV; + irqflags = IRQF_SHARED; if (pci->revision != 0x50) return -ENODEV; -- 2.39.2

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[PATCH AUTOSEL 5.15 02/23] ALSA: emu10k1: roll up loops in DSP setup code for Audigy
by Sasha Levin 24 Jul '23

24 Jul '23

From: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> [ Upstream commit 8cabf83c7aa54530e699be56249fb44f9505c4f3 ] There is no apparent reason for the massive code duplication. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> Link: https://lore.kernel.org/r/20230510173917.3073107-3-oswald.buddenhagen@gmx.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/emu10k1/emufx.c | 112 +++----------------------------------- 1 file changed, 9 insertions(+), 103 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 6cf7c8b1de47d..4ca65f425f39c 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1563,14 +1563,8 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr += 2; /* Master volume (will be renamed later) */ - A_OP(icode, &ptr, iMAC0, A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS)); + for (z = 0; z < 8; z++) + A_OP(icode, &ptr, iMAC0, A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS)); snd_emu10k1_init_mono_control(&controls[nctl++], "Wave Master Playback Volume", gpr, 0); gpr += 2; @@ -1654,102 +1648,14 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) dev_dbg(emu->card->dev, "emufx.c: gpr=0x%x, tmp=0x%x\n", gpr, tmp); */ - /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ - /* A_P16VIN(0) is delayed by one sample, - * so all other A_P16VIN channels will need to also be delayed - */ - /* Left ADC in. 1 of 2 */ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); - /* Right ADC in 1 of 2 */ - gpr_map[gpr++] = 0x00000000; - /* Delaying by one sample: instead of copying the input - * value A_P16VIN to output A_FXBUS2 as in the first channel, - * we use an auxiliary register, delaying the value by one - * sample - */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); - /* For 96kHz mode */ - /* Left ADC in. 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); - /* Right ADC in 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); - /* Pavel Hofman - we still have voices, A_FXBUS2s, and - * A_P16VINs available - - * let's add 8 more capture channels - total of 16 - */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x10)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x12)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x14)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x16)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x18)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1a)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1c)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1e)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), - A_C_00000000, A_C_00000000); + /* A_P16VIN(0) is delayed by one sample, so all other A_P16VIN channels + * will need to also be delayed; we use an auxiliary register for that. */ + for (z = 1; z < 0x10; z++) { + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr), A_FXBUS2(z * 2) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr), A_P16VIN(z), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + } } #if 0 -- 2.39.2

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[PATCH AUTOSEL 6.1 37/41] ALSA: hda/realtek: Add quirks for ROG ALLY CS35l41 audio
by Sasha Levin 24 Jul '23

24 Jul '23

From: Matthew Anderson <ruinairas1992(a)gmail.com> [ Upstream commit 724418b84e6248cd27599607b7e5fac365b8e3f5 ] This requires a patched ACPI table or a firmware from ASUS to work because the system does not come with the _DSD field for the CSC3551. Link: https://bugzilla.kernel.org/show_bug.cgi?id=217550 Signed-off-by: Matthew Anderson <ruinairas1992(a)gmail.com> Tested-by: Philip Mueller <philm(a)manjaro.org> Link: https://lore.kernel.org/r/20230621161714.9442-1-ruinairas1992@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 920e44ba998a5..5849670965f77 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7120,6 +7120,10 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, + ALC294_FIXUP_ASUS_ALLY, + ALC294_FIXUP_ASUS_ALLY_PINS, + ALC294_FIXUP_ASUS_ALLY_VERBS, + ALC294_FIXUP_ASUS_ALLY_SPEAKER, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, ALC294_FIXUP_ASUS_GX502_HP, @@ -8432,6 +8436,47 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, + [ALC294_FIXUP_ASUS_ALLY] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_i2c_two, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS + }, + [ALC294_FIXUP_ASUS_ALLY_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1a, 0x03a11c30 }, + { 0x21, 0x03211420 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_VERBS + }, + [ALC294_FIXUP_ASUS_ALLY_VERBS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x46 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0004 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x47 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xa47a }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x49 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0049}, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x4a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x201b }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x6b }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x4278}, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_SPEAKER + }, + [ALC294_FIXUP_ASUS_ALLY_SPEAKER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + }, [ALC285_FIXUP_THINKPAD_X1_GEN7] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_thinkpad_x1_gen7, @@ -9534,6 +9579,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), + SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally RC71L_RC71L", ALC294_FIXUP_ASUS_ALLY), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.39.2

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[PATCH AUTOSEL 6.1 33/41] ASoC: SOF: core: Free the firmware trace before calling snd_sof_shutdown()
by Sasha Levin 24 Jul '23

24 Jul '23

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit d389dcb3a48cec4f03c16434c0bf98a4c635372a ] The shutdown is called on reboot/shutdown of the machine. At this point the firmware tracing cannot be used anymore but in case of IPC3 it is using and keeping a DMA channel active (dtrace). For Tiger Lake platforms we have a quirk in place to fix rare reboot issues when a DMA was active before rebooting the system. If the tracing is enabled this quirk will be always used and a print appears on the kernel log which might be misleading or not even correct. Release the fw tracing before executing the shutdown to make sure that this known DMA user is cleared away. Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230616100039.378150-4-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 625977a29d8a8..75a1e2c6539f2 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -479,8 +479,10 @@ int snd_sof_device_shutdown(struct device *dev) if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); - if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) + if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) { + sof_fw_trace_free(sdev); return snd_sof_shutdown(sdev); + } return 0; } -- 2.39.2

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[PATCH AUTOSEL 6.1 21/41] ASoC: SOF: Intel: fix SoundWire/HDaudio mutual exclusion
by Sasha Levin 24 Jul '23

24 Jul '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit f751b99255cacd9ffe8c4bbf99767ad670cee1f7 ] The functionality described in Commit 61bef9e68dca ("ASoC: SOF: Intel: hda: enforce exclusion between HDaudio and SoundWire") does not seem to be properly implemented with two issues that need to be corrected. a) The test used is incorrect when DisplayAudio codecs are not supported. b) Conversely when only Display Audio codecs can be found, we do want to start the SoundWire links, if any. That will help add the relevant topologies and machine descriptors, and identify cases where the SoundWire information in ACPI needs to be modified with a quirk. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20230606222529.57156-2-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 1188ec51816bd..63764afdcf617 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1309,12 +1309,22 @@ static void hda_generic_machine_select(struct snd_sof_dev *sdev, hda_mach->mach_params.dmic_num = dmic_num; pdata->tplg_filename = tplg_filename; - if (codec_num == 2) { + if (codec_num == 2 || + (codec_num == 1 && !HDA_IDISP_CODEC(bus->codec_mask))) { /* * Prevent SoundWire links from starting when an external * HDaudio codec is used */ hda_mach->mach_params.link_mask = 0; + } else { + /* + * Allow SoundWire links to start when no external HDaudio codec + * was detected. This will not create a SoundWire card but + * will help detect if any SoundWire codec reports as ATTACHED. + */ + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; + + hda_mach->mach_params.link_mask = hdev->info.link_mask; } *mach = hda_mach; -- 2.39.2

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[PATCH AUTOSEL 6.1 20/41] ALSA: hda: Add Loongson LS7A HD-Audio support
by Sasha Levin 24 Jul '23

24 Jul '23

From: Yanteng Si <siyanteng(a)loongson.cn> [ Upstream commit 28bd137a3c8e105587ba8c55b68ef43b519b270f ] Add the new PCI ID 0x0014 0x7a07 and the new PCI ID 0x0014 0x7a37 Loongson HDA controller. Signed-off-by: Yanteng Si <siyanteng(a)loongson.cn> Acked-by: Huacai Chen <chenhuacai(a)loongson.cn> Link: https://lore.kernel.org/r/993587483b9509796b29a416f257fcfb4b15c6ea.16861288… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/linux/pci_ids.h | 3 +++ sound/hda/hdac_device.c | 1 + sound/pci/hda/hda_intel.c | 7 +++++++ sound/pci/hda/patch_hdmi.c | 1 + 4 files changed, 12 insertions(+) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index 95f33dadb2be2..c0c4ca8e28510 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -158,6 +158,9 @@ #define PCI_VENDOR_ID_LOONGSON 0x0014 +#define PCI_DEVICE_ID_LOONGSON_HDA 0x7a07 +#define PCI_DEVICE_ID_LOONGSON_HDMI 0x7a37 + #define PCI_VENDOR_ID_TTTECH 0x0357 #define PCI_DEVICE_ID_TTTECH_MC322 0x000a diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index bfd8585776767..5315a3eb114ee 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -645,6 +645,7 @@ struct hda_vendor_id { }; static const struct hda_vendor_id hda_vendor_ids[] = { + { 0x0014, "Loongson" }, { 0x1002, "ATI" }, { 0x1013, "Cirrus Logic" }, { 0x1057, "Motorola" }, diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 3226691ac923c..9c353dc7740c4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -237,6 +237,7 @@ enum { AZX_DRIVER_CTHDA, AZX_DRIVER_CMEDIA, AZX_DRIVER_ZHAOXIN, + AZX_DRIVER_LOONGSON, AZX_DRIVER_GENERIC, AZX_NUM_DRIVERS, /* keep this as last entry */ }; @@ -360,6 +361,7 @@ static const char * const driver_short_names[] = { [AZX_DRIVER_CTHDA] = "HDA Creative", [AZX_DRIVER_CMEDIA] = "HDA C-Media", [AZX_DRIVER_ZHAOXIN] = "HDA Zhaoxin", + [AZX_DRIVER_LOONGSON] = "HDA Loongson", [AZX_DRIVER_GENERIC] = "HD-Audio Generic", }; @@ -2809,6 +2811,11 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_GENERIC | AZX_DCAPS_PRESET_ATI_HDMI }, /* Zhaoxin */ { PCI_DEVICE(0x1d17, 0x3288), .driver_data = AZX_DRIVER_ZHAOXIN }, + /* Loongson HDAudio*/ + {PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, PCI_DEVICE_ID_LOONGSON_HDA), + .driver_data = AZX_DRIVER_LOONGSON }, + {PCI_DEVICE(PCI_VENDOR_ID_LOONGSON, PCI_DEVICE_ID_LOONGSON_HDMI), + .driver_data = AZX_DRIVER_LOONGSON }, { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 7b5e09070ab9b..b9b138de57951 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4509,6 +4509,7 @@ static int patch_gf_hdmi(struct hda_codec *codec) * patch entries */ static const struct hda_device_id snd_hda_id_hdmi[] = { +HDA_CODEC_ENTRY(0x00147a47, "Loongson HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x1002793c, "RS600 HDMI", patch_atihdmi), HDA_CODEC_ENTRY(0x10027919, "RS600 HDMI", patch_atihdmi), HDA_CODEC_ENTRY(0x1002791a, "RS690/780 HDMI", patch_atihdmi), -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 16/41] ASoC: Intel: sof_sdw: Add support for Rex soundwire
by Sasha Levin 24 Jul '23

24 Jul '23

From: Uday M Bhat <uday.m.bhat(a)intel.com> [ Upstream commit 164e5dc17525181c05563f0a06796f1a363801d5 ] Add rex entry in the soundwire quirk table Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Uday M Bhat <uday.m.bhat(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230602202225.249209-28-pierre-louis.bossart@lin… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index da19bfdac1ea7..046f464739271 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -382,6 +382,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2_100K), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Rex"), + }, + .driver_data = (void *)(SOF_SDW_PCH_DMIC), + }, /* LunarLake devices */ { .callback = sof_sdw_quirk_cb, -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 15/41] ASoC: Intel: sof_sdw_rt_sdca_jack_common: test SOF_JACK_JDSRC in _exit
by Sasha Levin 24 Jul '23

24 Jul '23

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 526a1876fc48e2d0c0ea8ad63b58bdb2cc13047f ] if (!SOF_RT711_JDSRC(sof_sdw_quirk)) is tested in rt711_sdca_add_codec_ device_props(), and we don't add software node to the device if jack source is not set. We need to do the same test in sof_sdw_rt711_sdca_exit(), and avoid removing software node if jack source is not set. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230602202225.249209-8-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw_rt711_sdca.c b/sound/soc/intel/boards/sof_sdw_rt711_sdca.c index 7f16304d025be..cf8b9793fe0e5 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt711_sdca.c @@ -143,6 +143,9 @@ int sof_sdw_rt711_sdca_exit(struct snd_soc_card *card, struct snd_soc_dai_link * if (!ctx->headset_codec_dev) return 0; + if (!SOF_RT711_JDSRC(sof_sdw_quirk)) + return 0; + device_remove_software_node(ctx->headset_codec_dev); put_device(ctx->headset_codec_dev); -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 13/41] ASoC: amd: vangogh: Add check for acp config flags in vangogh platform
by Sasha Levin 24 Jul '23

24 Jul '23

From: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> [ Upstream commit e89f45edb747ed88e97a5771dd6d3dd1eb517873 ] We have SOF and generic ACP support enabled for Vangogh platform on some machines. Since we have same PCI id used for probing, add check for machine configuration flag to avoid conflict with newer pci drivers. Such machine flag has been initialized via dmi match on few Vangogh based machines. If no flag is specified probe and register older platform device. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Link: https://lore.kernel.org/r/20230530110802.674939-1-venkataprasad.potturu@amd… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/vangogh/acp5x.h | 2 ++ sound/soc/amd/vangogh/pci-acp5x.c | 7 ++++++- 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/vangogh/acp5x.h b/sound/soc/amd/vangogh/acp5x.h index bd9f1c5684d17..ac1936a8c43ff 100644 --- a/sound/soc/amd/vangogh/acp5x.h +++ b/sound/soc/amd/vangogh/acp5x.h @@ -147,6 +147,8 @@ static inline void acp_writel(u32 val, void __iomem *base_addr) writel(val, base_addr - ACP5x_PHY_BASE_ADDRESS); } +int snd_amd_acp_find_config(struct pci_dev *pci); + static inline u64 acp_get_byte_count(struct i2s_stream_instance *rtd, int direction) { diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index e0df17c88e8e0..c4634a8a17cdc 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -125,10 +125,15 @@ static int snd_acp5x_probe(struct pci_dev *pci, { struct acp5x_dev_data *adata; struct platform_device_info pdevinfo[ACP5x_DEVS]; - unsigned int irqflags; + unsigned int irqflags, flag; int ret, i; u32 addr, val; + /* Return if acp config flag is defined */ + flag = snd_amd_acp_find_config(pci); + if (flag) + return -ENODEV; + irqflags = IRQF_SHARED; if (pci->revision != 0x50) return -ENODEV; -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 10/41] ASoC: SOF: amd: Add pci revision id check
by Sasha Levin 24 Jul '23

24 Jul '23

From: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> [ Upstream commit 1d4a84632b90d88316986b05bcdfe715399a33db ] Add pci revision id check for renoir and rembrandt platforms. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Link: https://lore.kernel.org/r/20230523072009.2379198-1-venkataprasad.potturu@am… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/amd/acp.h | 3 +++ sound/soc/sof/amd/pci-rmb.c | 3 +++ sound/soc/sof/amd/pci-rn.c | 3 +++ 3 files changed, 9 insertions(+) diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index dd3c072d01721..14148c311f504 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -54,6 +54,9 @@ #define ACP_DSP_TO_HOST_IRQ 0x04 +#define ACP_RN_PCI_ID 0x01 +#define ACP_RMB_PCI_ID 0x6F + #define HOST_BRIDGE_CZN 0x1630 #define HOST_BRIDGE_RMB 0x14B5 #define ACP_SHA_STAT 0x8000 diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index 4e1de462b431b..5698d910b26f3 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -90,6 +90,9 @@ static int acp_pci_rmb_probe(struct pci_dev *pci, const struct pci_device_id *pc unsigned int flag, i, addr; int ret; + if (pci->revision != ACP_RMB_PCI_ID) + return -ENODEV; + flag = snd_amd_acp_find_config(pci); if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) return -ENODEV; diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index fca40b261671b..9189f63632789 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -90,6 +90,9 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci unsigned int flag, i, addr; int ret; + if (pci->revision != ACP_RN_PCI_ID) + return -ENODEV; + flag = snd_amd_acp_find_config(pci); if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) return -ENODEV; -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 08/41] ASoC: Intel: sof_sdw: add quirk for LNL RVP
by Sasha Levin 24 Jul '23

24 Jul '23

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit dfe25fea968dc4884e12d471c8263f0f611b380a ] We should use RT711_JD2_100K for on board rt711 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Link: https://lore.kernel.org/r/20230512173305.65399-9-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index a06693827ae5b..da19bfdac1ea7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -382,6 +382,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2_100K), }, + /* LunarLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Lunar Lake Client Platform"), + }, + .driver_data = (void *)(RT711_JD2_100K), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 07/41] ASoC: Intel: sof_sdw: add quirk for MTL RVP
by Sasha Levin 24 Jul '23

24 Jul '23

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 289e1df00e49a229a1c924c059242e759a552f01 ] We should use RT711_JD2_100K for on board rt711. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Link: https://lore.kernel.org/r/20230512173305.65399-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d4f92bb5e29f8..a06693827ae5b 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -374,6 +374,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD1 | SOF_SDW_TGL_HDMI), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Meteor Lake Client Platform"), + }, + .driver_data = (void *)(RT711_JD2_100K), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 6.1 06/41] ALSA: emu10k1: roll up loops in DSP setup code for Audigy
by Sasha Levin 24 Jul '23

24 Jul '23

From: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> [ Upstream commit 8cabf83c7aa54530e699be56249fb44f9505c4f3 ] There is no apparent reason for the massive code duplication. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> Link: https://lore.kernel.org/r/20230510173917.3073107-3-oswald.buddenhagen@gmx.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/emu10k1/emufx.c | 112 +++----------------------------------- 1 file changed, 9 insertions(+), 103 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 6cf7c8b1de47d..4ca65f425f39c 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1563,14 +1563,8 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr += 2; /* Master volume (will be renamed later) */ - A_OP(icode, &ptr, iMAC0, A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS)); + for (z = 0; z < 8; z++) + A_OP(icode, &ptr, iMAC0, A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS)); snd_emu10k1_init_mono_control(&controls[nctl++], "Wave Master Playback Volume", gpr, 0); gpr += 2; @@ -1654,102 +1648,14 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) dev_dbg(emu->card->dev, "emufx.c: gpr=0x%x, tmp=0x%x\n", gpr, tmp); */ - /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ - /* A_P16VIN(0) is delayed by one sample, - * so all other A_P16VIN channels will need to also be delayed - */ - /* Left ADC in. 1 of 2 */ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); - /* Right ADC in 1 of 2 */ - gpr_map[gpr++] = 0x00000000; - /* Delaying by one sample: instead of copying the input - * value A_P16VIN to output A_FXBUS2 as in the first channel, - * we use an auxiliary register, delaying the value by one - * sample - */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); - /* For 96kHz mode */ - /* Left ADC in. 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); - /* Right ADC in 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); - /* Pavel Hofman - we still have voices, A_FXBUS2s, and - * A_P16VINs available - - * let's add 8 more capture channels - total of 16 - */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x10)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x12)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x14)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x16)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x18)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1a)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1c)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1e)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), - A_C_00000000, A_C_00000000); + /* A_P16VIN(0) is delayed by one sample, so all other A_P16VIN channels + * will need to also be delayed; we use an auxiliary register for that. */ + for (z = 1; z < 0x10; z++) { + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr), A_FXBUS2(z * 2) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr), A_P16VIN(z), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + } } #if 0 -- 2.39.2

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[PATCH AUTOSEL 6.4 54/58] ALSA: hda/realtek: Add quirks for ROG ALLY CS35l41 audio
by Sasha Levin 24 Jul '23

24 Jul '23

From: Matthew Anderson <ruinairas1992(a)gmail.com> [ Upstream commit 724418b84e6248cd27599607b7e5fac365b8e3f5 ] This requires a patched ACPI table or a firmware from ASUS to work because the system does not come with the _DSD field for the CSC3551. Link: https://bugzilla.kernel.org/show_bug.cgi?id=217550 Signed-off-by: Matthew Anderson <ruinairas1992(a)gmail.com> Tested-by: Philip Mueller <philm(a)manjaro.org> Link: https://lore.kernel.org/r/20230621161714.9442-1-ruinairas1992@gmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/patch_realtek.c | 46 +++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dabfdecece264..9310014c70b81 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7120,6 +7120,10 @@ enum { ALC294_FIXUP_ASUS_DUAL_SPK, ALC285_FIXUP_THINKPAD_X1_GEN7, ALC285_FIXUP_THINKPAD_HEADSET_JACK, + ALC294_FIXUP_ASUS_ALLY, + ALC294_FIXUP_ASUS_ALLY_PINS, + ALC294_FIXUP_ASUS_ALLY_VERBS, + ALC294_FIXUP_ASUS_ALLY_SPEAKER, ALC294_FIXUP_ASUS_HPE, ALC294_FIXUP_ASUS_COEF_1B, ALC294_FIXUP_ASUS_GX502_HP, @@ -8432,6 +8436,47 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC294_FIXUP_SPK2_TO_DAC1 }, + [ALC294_FIXUP_ASUS_ALLY] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_i2c_two, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_PINS + }, + [ALC294_FIXUP_ASUS_ALLY_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x03a11050 }, + { 0x1a, 0x03a11c30 }, + { 0x21, 0x03211420 }, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_VERBS + }, + [ALC294_FIXUP_ASUS_ALLY_VERBS] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x46 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0004 }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x47 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0xa47a }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x49 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x0049}, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x4a }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x201b }, + { 0x20, AC_VERB_SET_COEF_INDEX, 0x6b }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x4278}, + { } + }, + .chained = true, + .chain_id = ALC294_FIXUP_ASUS_ALLY_SPEAKER + }, + [ALC294_FIXUP_ASUS_ALLY_SPEAKER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_speaker2_to_dac1, + }, [ALC285_FIXUP_THINKPAD_X1_GEN7] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_thinkpad_x1_gen7, @@ -9535,6 +9580,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1740, "ASUS UX430UA", ALC295_FIXUP_ASUS_DACS), SND_PCI_QUIRK(0x1043, 0x17d1, "ASUS UX431FL", ALC294_FIXUP_ASUS_DUAL_SPK), + SND_PCI_QUIRK(0x1043, 0x17f3, "ROG Ally RC71L_RC71L", ALC294_FIXUP_ASUS_ALLY), SND_PCI_QUIRK(0x1043, 0x1881, "ASUS Zephyrus S/M", ALC294_FIXUP_ASUS_GX502_PINS), SND_PCI_QUIRK(0x1043, 0x18b1, "Asus MJ401TA", ALC256_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x18f1, "Asus FX505DT", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.39.2

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[PATCH AUTOSEL 6.4 50/58] ASoC: SOF: core: Free the firmware trace before calling snd_sof_shutdown()
by Sasha Levin 24 Jul '23

24 Jul '23

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit d389dcb3a48cec4f03c16434c0bf98a4c635372a ] The shutdown is called on reboot/shutdown of the machine. At this point the firmware tracing cannot be used anymore but in case of IPC3 it is using and keeping a DMA channel active (dtrace). For Tiger Lake platforms we have a quirk in place to fix rare reboot issues when a DMA was active before rebooting the system. If the tracing is enabled this quirk will be always used and a print appears on the kernel log which might be misleading or not even correct. Release the fw tracing before executing the shutdown to make sure that this known DMA user is cleared away. Reviewed-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230616100039.378150-4-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 9a9d82220fd0d..30db685cc5f4b 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -504,8 +504,10 @@ int snd_sof_device_shutdown(struct device *dev) if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); - if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) + if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) { + sof_fw_trace_free(sdev); return snd_sof_shutdown(sdev); + } return 0; } -- 2.39.2

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[PATCH AUTOSEL 6.4 29/58] ASoC: SOF: Intel: fix SoundWire/HDaudio mutual exclusion
by Sasha Levin 24 Jul '23

24 Jul '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit f751b99255cacd9ffe8c4bbf99767ad670cee1f7 ] The functionality described in Commit 61bef9e68dca ("ASoC: SOF: Intel: hda: enforce exclusion between HDaudio and SoundWire") does not seem to be properly implemented with two issues that need to be corrected. a) The test used is incorrect when DisplayAudio codecs are not supported. b) Conversely when only Display Audio codecs can be found, we do want to start the SoundWire links, if any. That will help add the relevant topologies and machine descriptors, and identify cases where the SoundWire information in ACPI needs to be modified with a quirk. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/20230606222529.57156-2-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda.c | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 3153e21f100ab..3853582e32e12 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -1343,12 +1343,22 @@ static void hda_generic_machine_select(struct snd_sof_dev *sdev, hda_mach->mach_params.dmic_num = dmic_num; pdata->tplg_filename = tplg_filename; - if (codec_num == 2) { + if (codec_num == 2 || + (codec_num == 1 && !HDA_IDISP_CODEC(bus->codec_mask))) { /* * Prevent SoundWire links from starting when an external * HDaudio codec is used */ hda_mach->mach_params.link_mask = 0; + } else { + /* + * Allow SoundWire links to start when no external HDaudio codec + * was detected. This will not create a SoundWire card but + * will help detect if any SoundWire codec reports as ATTACHED. + */ + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; + + hda_mach->mach_params.link_mask = hdev->info.link_mask; } *mach = hda_mach; -- 2.39.2

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[PATCH AUTOSEL 6.4 24/58] ASoC: Intel: sof_sdw: Add support for Rex soundwire
by Sasha Levin 24 Jul '23

24 Jul '23

From: Uday M Bhat <uday.m.bhat(a)intel.com> [ Upstream commit 164e5dc17525181c05563f0a06796f1a363801d5 ] Add rex entry in the soundwire quirk table Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Yong Zhi <yong.zhi(a)intel.com> Signed-off-by: Uday M Bhat <uday.m.bhat(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230602202225.249209-28-pierre-louis.bossart@lin… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 1238a13744e2e..f0de85edb1de6 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -433,6 +433,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2_100K), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Google"), + DMI_MATCH(DMI_PRODUCT_NAME, "Rex"), + }, + .driver_data = (void *)(SOF_SDW_PCH_DMIC), + }, /* LunarLake devices */ { .callback = sof_sdw_quirk_cb, -- 2.39.2

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[PATCH AUTOSEL 6.4 23/58] ASoC: Intel: sof_sdw: add quick for Dell SKU 0BDA
by Sasha Levin 24 Jul '23

24 Jul '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 3daf02819ac3fd8d7605804a00213cf123ac880d ] The SKU numbering isn't quite consistent with the existing RaptorLake SKUs but the PCI ID is definitively RaptorLake. Closes: https://github.com/thesofproject/linux/issues/4380 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230602202225.249209-17-pierre-louis.bossart@lin… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 342d4f23a8dc6..1238a13744e2e 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -367,6 +367,16 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { RT711_JD2), }, /* RaptorLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), + DMI_EXACT_MATCH(DMI_PRODUCT_SKU, "0BDA") + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + RT711_JD2 | + SOF_SDW_FOUR_SPK), + }, { .callback = sof_sdw_quirk_cb, .matches = { -- 2.39.2

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[PATCH AUTOSEL 6.4 22/58] ASoC: Intel: sof_sdw_rt_sdca_jack_common: test SOF_JACK_JDSRC in _exit
by Sasha Levin 24 Jul '23

24 Jul '23

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 526a1876fc48e2d0c0ea8ad63b58bdb2cc13047f ] if (!SOF_RT711_JDSRC(sof_sdw_quirk)) is tested in rt711_sdca_add_codec_ device_props(), and we don't add software node to the device if jack source is not set. We need to do the same test in sof_sdw_rt711_sdca_exit(), and avoid removing software node if jack source is not set. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20230602202225.249209-8-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw_rt711_sdca.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw_rt711_sdca.c b/sound/soc/intel/boards/sof_sdw_rt711_sdca.c index 7f16304d025be..cf8b9793fe0e5 100644 --- a/sound/soc/intel/boards/sof_sdw_rt711_sdca.c +++ b/sound/soc/intel/boards/sof_sdw_rt711_sdca.c @@ -143,6 +143,9 @@ int sof_sdw_rt711_sdca_exit(struct snd_soc_card *card, struct snd_soc_dai_link * if (!ctx->headset_codec_dev) return 0; + if (!SOF_RT711_JDSRC(sof_sdw_quirk)) + return 0; + device_remove_software_node(ctx->headset_codec_dev); put_device(ctx->headset_codec_dev); -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 19/58] ASoC: amd: vangogh: Add check for acp config flags in vangogh platform
by Sasha Levin 24 Jul '23

24 Jul '23

From: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> [ Upstream commit e89f45edb747ed88e97a5771dd6d3dd1eb517873 ] We have SOF and generic ACP support enabled for Vangogh platform on some machines. Since we have same PCI id used for probing, add check for machine configuration flag to avoid conflict with newer pci drivers. Such machine flag has been initialized via dmi match on few Vangogh based machines. If no flag is specified probe and register older platform device. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Link: https://lore.kernel.org/r/20230530110802.674939-1-venkataprasad.potturu@amd… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/amd/vangogh/acp5x.h | 2 ++ sound/soc/amd/vangogh/pci-acp5x.c | 7 ++++++- 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/vangogh/acp5x.h b/sound/soc/amd/vangogh/acp5x.h index bd9f1c5684d17..ac1936a8c43ff 100644 --- a/sound/soc/amd/vangogh/acp5x.h +++ b/sound/soc/amd/vangogh/acp5x.h @@ -147,6 +147,8 @@ static inline void acp_writel(u32 val, void __iomem *base_addr) writel(val, base_addr - ACP5x_PHY_BASE_ADDRESS); } +int snd_amd_acp_find_config(struct pci_dev *pci); + static inline u64 acp_get_byte_count(struct i2s_stream_instance *rtd, int direction) { diff --git a/sound/soc/amd/vangogh/pci-acp5x.c b/sound/soc/amd/vangogh/pci-acp5x.c index e0df17c88e8e0..c4634a8a17cdc 100644 --- a/sound/soc/amd/vangogh/pci-acp5x.c +++ b/sound/soc/amd/vangogh/pci-acp5x.c @@ -125,10 +125,15 @@ static int snd_acp5x_probe(struct pci_dev *pci, { struct acp5x_dev_data *adata; struct platform_device_info pdevinfo[ACP5x_DEVS]; - unsigned int irqflags; + unsigned int irqflags, flag; int ret, i; u32 addr, val; + /* Return if acp config flag is defined */ + flag = snd_amd_acp_find_config(pci); + if (flag) + return -ENODEV; + irqflags = IRQF_SHARED; if (pci->revision != 0x50) return -ENODEV; -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 15/58] ASoC: SOF: amd: Add pci revision id check
by Sasha Levin 24 Jul '23

24 Jul '23

From: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> [ Upstream commit 1d4a84632b90d88316986b05bcdfe715399a33db ] Add pci revision id check for renoir and rembrandt platforms. Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> Link: https://lore.kernel.org/r/20230523072009.2379198-1-venkataprasad.potturu@am… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/amd/acp.h | 3 +++ sound/soc/sof/amd/pci-rmb.c | 3 +++ sound/soc/sof/amd/pci-rn.c | 3 +++ 3 files changed, 9 insertions(+) diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 1c535cc6c3a95..dc624f727aa37 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -55,6 +55,9 @@ #define ACP_DSP_TO_HOST_IRQ 0x04 +#define ACP_RN_PCI_ID 0x01 +#define ACP_RMB_PCI_ID 0x6F + #define HOST_BRIDGE_CZN 0x1630 #define HOST_BRIDGE_RMB 0x14B5 #define ACP_SHA_STAT 0x8000 diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index eaf70ea6e556e..58b3092425f1a 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -65,6 +65,9 @@ static int acp_pci_rmb_probe(struct pci_dev *pci, const struct pci_device_id *pc { unsigned int flag; + if (pci->revision != ACP_RMB_PCI_ID) + return -ENODEV; + flag = snd_amd_acp_find_config(pci); if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) return -ENODEV; diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 4809cb644152b..7409e21ce5aa7 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -65,6 +65,9 @@ static int acp_pci_rn_probe(struct pci_dev *pci, const struct pci_device_id *pci { unsigned int flag; + if (pci->revision != ACP_RN_PCI_ID) + return -ENODEV; + flag = snd_amd_acp_find_config(pci); if (flag != FLAG_AMD_SOF && flag != FLAG_AMD_SOF_ONLY_DMIC) return -ENODEV; -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 13/58] ASoC: cs35l56: Move DSP part string generation so that it is done only once
by Sasha Levin 24 Jul '23

24 Jul '23

From: Simon Trimmer <simont(a)opensource.cirrus.com> [ Upstream commit 608f1b0dbddec6b2fd766c10bcce2b651995e936 ] Each time we go through dsp_work() it does a devm_kasprintf() to allocate memory to hold the part name string. It's not strictly a memory leak because devm will free it all if the driver is removed. But we keep allocating more and more memory to hold the same string. Move the allocation so that it is performed after the version and secured state information is gathered and handle allocation errors. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Link: https://lore.kernel.org/r/Message-Id: <20230518150250.1121006-2-rf(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/cs35l56.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index e0d2b9bb23262..f3fee448d759e 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -834,12 +834,6 @@ static void cs35l56_dsp_work(struct work_struct *work) if (!cs35l56->init_done) return; - cs35l56->dsp.part = devm_kasprintf(cs35l56->dev, GFP_KERNEL, "cs35l56%s-%02x", - cs35l56->secured ? "s" : "", cs35l56->rev); - - if (!cs35l56->dsp.part) - return; - pm_runtime_get_sync(cs35l56->dev); /* @@ -1505,6 +1499,12 @@ int cs35l56_init(struct cs35l56_private *cs35l56) dev_info(cs35l56->dev, "Cirrus Logic CS35L56%s Rev %02X OTP%d\n", cs35l56->secured ? "s" : "", cs35l56->rev, otpid); + /* Populate the DSP information with the revision and security state */ + cs35l56->dsp.part = devm_kasprintf(cs35l56->dev, GFP_KERNEL, "cs35l56%s-%02x", + cs35l56->secured ? "s" : "", cs35l56->rev); + if (!cs35l56->dsp.part) + return -ENOMEM; + /* Wake source and *_BLOCKED interrupts default to unmasked, so mask them */ regmap_write(cs35l56->regmap, CS35L56_IRQ1_MASK_20, 0xffffffff); regmap_update_bits(cs35l56->regmap, CS35L56_IRQ1_MASK_1, -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 11/58] ASoC: Intel: sof_sdw: add quirk for LNL RVP
by Sasha Levin 24 Jul '23

24 Jul '23

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> [ Upstream commit dfe25fea968dc4884e12d471c8263f0f611b380a ] We should use RT711_JD2_100K for on board rt711 Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Link: https://lore.kernel.org/r/20230512173305.65399-9-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index db99f7fefdfc7..342d4f23a8dc6 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -423,6 +423,15 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD2_100K), }, + /* LunarLake devices */ + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Lunar Lake Client Platform"), + }, + .driver_data = (void *)(RT711_JD2_100K), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 10/58] ASoC: Intel: sof_sdw: add quirk for MTL RVP
by Sasha Levin 24 Jul '23

24 Jul '23

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> [ Upstream commit 289e1df00e49a229a1c924c059242e759a552f01 ] We should use RT711_JD2_100K for on board rt711. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Link: https://lore.kernel.org/r/20230512173305.65399-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_sdw.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 144f082c63fda..db99f7fefdfc7 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -415,6 +415,14 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { }, .driver_data = (void *)(RT711_JD1 | SOF_SDW_TGL_HDMI), }, + { + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_PRODUCT_NAME, "Meteor Lake Client Platform"), + }, + .driver_data = (void *)(RT711_JD2_100K), + }, {} }; -- 2.39.2

1 0

[PATCH AUTOSEL 6.4 09/58] ALSA: emu10k1: roll up loops in DSP setup code for Audigy
by Sasha Levin 24 Jul '23

24 Jul '23

From: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> [ Upstream commit 8cabf83c7aa54530e699be56249fb44f9505c4f3 ] There is no apparent reason for the massive code duplication. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> Link: https://lore.kernel.org/r/20230510173917.3073107-3-oswald.buddenhagen@gmx.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/emu10k1/emufx.c | 112 +++----------------------------------- 1 file changed, 9 insertions(+), 103 deletions(-) diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 3f64ccab0e632..fba19a854f27a 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -1559,14 +1559,8 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) gpr += 2; /* Master volume (will be renamed later) */ - A_OP(icode, &ptr, iMAC0, A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+0+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+1+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+2+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+3+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+4+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+5+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+6+SND_EMU10K1_PLAYBACK_CHANNELS)); - A_OP(icode, &ptr, iMAC0, A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+7+SND_EMU10K1_PLAYBACK_CHANNELS)); + for (z = 0; z < 8; z++) + A_OP(icode, &ptr, iMAC0, A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS), A_C_00000000, A_GPR(gpr), A_GPR(playback+z+SND_EMU10K1_PLAYBACK_CHANNELS)); snd_emu10k1_init_mono_control(&controls[nctl++], "Wave Master Playback Volume", gpr, 0); gpr += 2; @@ -1653,102 +1647,14 @@ A_OP(icode, &ptr, iMAC0, A_GPR(var), A_GPR(var), A_GPR(vol), A_EXTIN(input)) dev_dbg(emu->card->dev, "emufx.c: gpr=0x%x, tmp=0x%x\n", gpr, tmp); */ - /* For the EMU1010: How to get 32bit values from the DSP. High 16bits into L, low 16bits into R. */ - /* A_P16VIN(0) is delayed by one sample, - * so all other A_P16VIN channels will need to also be delayed - */ - /* Left ADC in. 1 of 2 */ snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_P16VIN(0x0), A_FXBUS2(0) ); - /* Right ADC in 1 of 2 */ - gpr_map[gpr++] = 0x00000000; - /* Delaying by one sample: instead of copying the input - * value A_P16VIN to output A_FXBUS2 as in the first channel, - * we use an auxiliary register, delaying the value by one - * sample - */ - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(2) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x1), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(4) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x2), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(6) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x3), A_C_00000000, A_C_00000000); - /* For 96kHz mode */ - /* Left ADC in. 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0x8) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x4), A_C_00000000, A_C_00000000); - /* Right ADC in 2 of 2 */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xa) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x5), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xc) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x6), A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr - 1), A_FXBUS2(0xe) ); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x7), A_C_00000000, A_C_00000000); - /* Pavel Hofman - we still have voices, A_FXBUS2s, and - * A_P16VINs available - - * let's add 8 more capture channels - total of 16 - */ - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x10)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x8), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x12)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0x9), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x14)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xa), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x16)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xb), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x18)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xc), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1a)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xd), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1c)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xe), - A_C_00000000, A_C_00000000); - gpr_map[gpr++] = 0x00000000; - snd_emu10k1_audigy_dsp_convert_32_to_2x16(icode, &ptr, tmp, - bit_shifter16, - A_GPR(gpr - 1), - A_FXBUS2(0x1e)); - A_OP(icode, &ptr, iACC3, A_GPR(gpr - 1), A_P16VIN(0xf), - A_C_00000000, A_C_00000000); + /* A_P16VIN(0) is delayed by one sample, so all other A_P16VIN channels + * will need to also be delayed; we use an auxiliary register for that. */ + for (z = 1; z < 0x10; z++) { + snd_emu10k1_audigy_dsp_convert_32_to_2x16( icode, &ptr, tmp, bit_shifter16, A_GPR(gpr), A_FXBUS2(z * 2) ); + A_OP(icode, &ptr, iACC3, A_GPR(gpr), A_P16VIN(z), A_C_00000000, A_C_00000000); + gpr_map[gpr++] = 0x00000000; + } } #if 0 -- 2.39.2

1 0

[PATCH] ASoC: rt5668: add missed regulator_bulk_disable
by Zhang Shurong 23 Jul '23

23 Jul '23

The driver forgets to call regulator_bulk_disable() Add the missed call to fix it. Fixes: d59fb2856223 ("ASoC: rt5668: add rt5668B codec driver") Signed-off-by: Zhang Shurong <zhang_shurong(a)foxmail.com> --- sound/soc/codecs/rt5668.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c index f04c810fd710..d66c9e9b9185 100644 --- a/sound/soc/codecs/rt5668.c +++ b/sound/soc/codecs/rt5668.c @@ -2452,6 +2452,13 @@ static void rt5668_calibrate(struct rt5668_priv *rt5668) } +static void rt5668_i2c_disable_regulators(void *data) +{ + struct rt5668_priv *rt5668 = data; + + regulator_bulk_disable(ARRAY_SIZE(rt5668->supplies), rt5668->supplies); +} + static int rt5668_i2c_probe(struct i2c_client *i2c) { struct rt5668_platform_data *pdata = dev_get_platdata(&i2c->dev); @@ -2490,6 +2497,11 @@ static int rt5668_i2c_probe(struct i2c_client *i2c) return ret; } + ret = devm_add_action_or_reset(&i2c->dev, rt5668_i2c_disable_regulators, + rt5668); + if (ret) + return ret; + ret = regulator_bulk_enable(ARRAY_SIZE(rt5668->supplies), rt5668->supplies); if (ret != 0) { -- 2.41.0

1 0

alsactl: add define to compile with glibc 2.38
by GitHub pull_request - opened 22 Jul '23

22 Jul '23

alsa-project/alsa-utils pull request #225 was opened from heitbaum: strlcat and strlcpy have been added to glibc 2.38. update the defines to use the glibc versions, and not conflict with string.h. ref: - https://sourceware.org/git/?p=glibc.git;a=commit;h=454a20c8756c9c1d55419153… Request URL : https://github.com/alsa-project/alsa-utils/pull/225 Patch URL : https://github.com/alsa-project/alsa-utils/pull/225.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

USB soundcard no longer working after update from 1.2.8 to 1.2.9
by GitHub issues - opened 22 Jul '23

22 Jul '23

alsa-project/alsa-lib issue #336 was opened from bothesven: I use a "Teufel Cinebar Ultima" as a usb sound card with driver snd-usb-audio. This works with alsa-lib, alsa-ucm-conf and lib32-alsa-lib in version 1.2.8. When these libraries are updated to 1.2.9, any player just hangs when this sound card is selected as output. [alsa-info.txt.gz](https://github.com/alsa-project/alsa-lib/files/12134686/a… Issue URL : https://github.com/alsa-project/alsa-lib/issues/336 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: fsl_spdif: Silence output on stop
by Matus Gajdos 21 Jul '23

21 Jul '23

Clear TX registers on stop to prevent the SPDIF interface from sending last written word over and over again. Fixes: a2388a498ad2 ("ASoC: fsl: Add S/PDIF CPU DAI driver") Signed-off-by: Matus Gajdos <matuszpd(a)gmail.com> --- sound/soc/fsl/fsl_spdif.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 95e639711eba..95bb8b10494a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -755,6 +755,8 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_PAUSE_PUSH: regmap_update_bits(regmap, REG_SPDIF_SCR, dmaen, 0); regmap_update_bits(regmap, REG_SPDIF_SIE, intr, 0); + regmap_write(regmap, REG_SPDIF_STL, 0x0); + regmap_write(regmap, REG_SPDIF_STR, 0x0); break; default: return -EINVAL; -- 2.25.1

3 2

[PATCH v5 0/6] Add cs42l43 PC focused SoundWire CODEC
by Charles Keepax 21 Jul '23

21 Jul '23

This patch chain adds support for the Cirrus Logic cs42l43 PC focused SoundWire CODEC. The chain is currently based of Lee's for-mfd-next branch. Thanks, Charles Charles Keepax (4): dt-bindings: mfd: cirrus,cs42l43: Add initial DT binding mfd: cs42l43: Add support for cs42l43 core driver pinctrl: cs42l43: Add support for the cs42l43 ASoC: cs42l43: Add support for the cs42l43 Lucas Tanure (2): soundwire: bus: Allow SoundWire peripherals to register IRQ handlers spi: cs42l43: Add SPI controller support .../bindings/sound/cirrus,cs42l43.yaml | 313 +++ MAINTAINERS | 4 + drivers/mfd/Kconfig | 23 + drivers/mfd/Makefile | 3 + drivers/mfd/cs42l43-i2c.c | 98 + drivers/mfd/cs42l43-sdw.c | 239 ++ drivers/mfd/cs42l43.c | 1189 +++++++++ drivers/mfd/cs42l43.h | 28 + drivers/pinctrl/cirrus/Kconfig | 11 + drivers/pinctrl/cirrus/Makefile | 2 + drivers/pinctrl/cirrus/pinctrl-cs42l43.c | 609 +++++ drivers/soundwire/bus.c | 32 + drivers/soundwire/bus_type.c | 12 + drivers/spi/Kconfig | 7 + drivers/spi/Makefile | 1 + drivers/spi/spi-cs42l43.c | 281 ++ include/linux/mfd/cs42l43-regs.h | 1184 +++++++++ include/linux/mfd/cs42l43.h | 102 + include/linux/soundwire/sdw.h | 9 + include/sound/cs42l43.h | 17 + sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42l43-jack.c | 969 +++++++ sound/soc/codecs/cs42l43-sdw.c | 74 + sound/soc/codecs/cs42l43.c | 2278 +++++++++++++++++ sound/soc/codecs/cs42l43.h | 131 + 26 files changed, 7636 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs42l43.yaml create mode 100644 drivers/mfd/cs42l43-i2c.c create mode 100644 drivers/mfd/cs42l43-sdw.c create mode 100644 drivers/mfd/cs42l43.c create mode 100644 drivers/mfd/cs42l43.h create mode 100644 drivers/pinctrl/cirrus/pinctrl-cs42l43.c create mode 100644 drivers/spi/spi-cs42l43.c create mode 100644 include/linux/mfd/cs42l43-regs.h create mode 100644 include/linux/mfd/cs42l43.h create mode 100644 include/sound/cs42l43.h create mode 100644 sound/soc/codecs/cs42l43-jack.c create mode 100644 sound/soc/codecs/cs42l43-sdw.c create mode 100644 sound/soc/codecs/cs42l43.c create mode 100644 sound/soc/codecs/cs42l43.h -- 2.30.2

2 11

[PATCH 2/2] ASoC: rt1316: fix key tone missing
by shumingf＠realtek.com 21 Jul '23

21 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> This patch disables the soft ramp to output immediately. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1316-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index ce2246a2591c..feea11149d93 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -134,7 +134,7 @@ static const struct reg_sequence rt1316_blind_write[] = { { 0xc093, 0x80 }, { 0xc09d, 0x80 }, { 0xc0b0, 0x77 }, - { 0xc010, 0xa5 }, + { 0xc010, 0x25 }, { 0xc050, 0x83 }, { 0x2f55, 0x03 }, { 0x3217, 0xb5 }, -- 2.34.1

3 2

[PATCH v1 00/11] Fix support for System Suspend for CS35L41 HDA
by Stefan Binding 21 Jul '23

21 Jul '23

There have been a couple of customer reports of intermittant issues after system resume, where sometimes the DSP firmware stops responding. Investigations into this issue show that there is a race between receiving a prepare from the HDA core, and the firmware reload which is started by the system resume. This can causes the Global Enable on the CS35L41 to be enabled during the firmware load, which can sometimes cause issues in the DSP. The existing system resume behaviour also did not resume the audio, if audio was previously playing when it was suspended. In addition, during investigation, it was found there were additional problems in the System Resume sequence, as well as the Playback sequence with External Boost, where the driver does not correctly follow its enable sequence for this mode. This can cause additional issues such as pops and clicks. This chain intends to correct the sequences for playback and system suspend/resume so that the driver: obeys the external boost enable sequence; resumes audio on system resume; and avoids the race condition on firmware load and playback during system resume. Stefan Binding (11): ALSA: cs35l41: Use mbox command to enable speaker output for external boost ALSA: hda: cs35l41: Check mailbox status of pause command after firmware load ALSA: hda: cs35l41: Ensure we correctly re-sync regmap before system suspending. ALSA: hda: cs35l41: Ensure we pass up any errors during system suspend. ALSA: hda: cs35l41: Move Play and Pause into separate functions ALSA: hda: hda_component: Add pre and post playback hooks to hda_component ALSA: hda: cs35l41: Use pre and post playback hooks ALSA: hda: cs35l41: Rework System Suspend to ensure correct call separation ALSA: hda/realtek: Support pre-/post- playback hooks for cs35l41 ALSA: hda: cs35l41: Add device_link between HDA and cs35l41_hda ALSA: hda: cs35l41: Ensure amp is only unmuted during playback include/sound/cs35l41.h | 5 +- sound/pci/hda/cs35l41_hda.c | 288 +++++++++++++++++++++++++-------- sound/pci/hda/hda_component.h | 2 + sound/pci/hda/patch_realtek.c | 10 +- sound/soc/codecs/cs35l41-lib.c | 118 ++++++++++++-- sound/soc/codecs/cs35l41.c | 18 +-- 6 files changed, 343 insertions(+), 98 deletions(-) -- 2.34.1

3 16

[PATCH 0/2] ALSA: hda/hdmi: Add Intel Lunar Lake support
by Peter Ujfalusi 21 Jul '23

21 Jul '23

Hi, after fixing the ordering, add the HDMI codec ID for LNL Regards, Peter --- Kai Vehmanen (2): ALSA: hda/hdmi: keep codec entries in numerical order ALSA: hda: add HDMI codec ID for Intel LNL sound/pci/hda/patch_hdmi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) -- 2.41.0

2 3

[PATCH v1 1/1] ALSA: korg1212: Re-use sockptr_t and respective APIs
by Andy Shevchenko 21 Jul '23

21 Jul '23

The sockptr_t (despite the naming) is a generic type to hold kernel or user pointer and there are respective APIs to copy data to or from it. Replace open coded variants in the driver by them. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/pci/korg1212/korg1212.c | 33 +++++++++++++-------------------- 1 file changed, 13 insertions(+), 20 deletions(-) diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c index 33b4f95d65b3..92c3eab4d12c 100644 --- a/sound/pci/korg1212/korg1212.c +++ b/sound/pci/korg1212/korg1212.c @@ -10,6 +10,7 @@ #include <linux/interrupt.h> #include <linux/pci.h> #include <linux/slab.h> +#include <linux/sockptr.h> #include <linux/wait.h> #include <linux/module.h> #include <linux/mutex.h> @@ -1285,8 +1286,7 @@ static int snd_korg1212_silence(struct snd_korg1212 *korg1212, int pos, int coun } static int snd_korg1212_copy_to(struct snd_pcm_substream *substream, - void __user *dst, int pos, int count, - bool in_kernel) + sockptr_t dst, int pos, int count) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); @@ -1306,24 +1306,21 @@ static int snd_korg1212_copy_to(struct snd_pcm_substream *substream, #if K1212_DEBUG_LEVEL > 0 if ( (void *) src < (void *) korg1212->recordDataBufsPtr || (void *) src > (void *) korg1212->recordDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_to KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + printk(KERN_DEBUG "K1212_DEBUG: %s KERNEL EFAULT, src=%p dst=%p iter=%d\n", + __func__, src, sockptr_is_kernel(dst) ? dst.kernel : dst.user, i); return -EFAULT; } #endif - if (in_kernel) - memcpy((__force void *)dst, src, size); - else if (copy_to_user(dst, src, size)) + if (copy_to_sockptr_offset(dst, i * size, src, size)) return -EFAULT; src++; - dst += size; } return 0; } static int snd_korg1212_copy_from(struct snd_pcm_substream *substream, - void __user *src, int pos, int count, - bool in_kernel) + sockptr_t src, int pos, int count) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_korg1212 *korg1212 = snd_pcm_substream_chip(substream); @@ -1345,16 +1342,14 @@ static int snd_korg1212_copy_from(struct snd_pcm_substream *substream, #if K1212_DEBUG_LEVEL > 0 if ( (void *) dst < (void *) korg1212->playDataBufsPtr || (void *) dst > (void *) korg1212->playDataBufsPtr[8].bufferData ) { - printk(KERN_DEBUG "K1212_DEBUG: snd_korg1212_copy_from KERNEL EFAULT, src=%p dst=%p iter=%d\n", src, dst, i); + printk(KERN_DEBUG "K1212_DEBUG: %s KERNEL EFAULT, src=%p dst=%p iter=%d\n", + __func__, sockptr_is_kernel(src) ? src.kernel : src.user, dst, i); return -EFAULT; } #endif - if (in_kernel) - memcpy(dst, (__force void *)src, size); - else if (copy_from_user(dst, src, size)) + if (copy_from_sockptr_offset(dst, src, i * size, size)) return -EFAULT; dst++; - src += size; } return 0; @@ -1644,15 +1639,14 @@ static int snd_korg1212_playback_copy(struct snd_pcm_substream *substream, int channel, unsigned long pos, void __user *src, unsigned long count) { - return snd_korg1212_copy_from(substream, src, pos, count, false); + return snd_korg1212_copy_from(substream, USER_SOCKPTR(src), pos, count); } static int snd_korg1212_playback_copy_kernel(struct snd_pcm_substream *substream, int channel, unsigned long pos, void *src, unsigned long count) { - return snd_korg1212_copy_from(substream, (void __user *)src, - pos, count, true); + return snd_korg1212_copy_from(substream, KERNEL_SOCKPTR(src), pos, count); } static int snd_korg1212_playback_silence(struct snd_pcm_substream *substream, @@ -1672,15 +1666,14 @@ static int snd_korg1212_capture_copy(struct snd_pcm_substream *substream, int channel, unsigned long pos, void __user *dst, unsigned long count) { - return snd_korg1212_copy_to(substream, dst, pos, count, false); + return snd_korg1212_copy_to(substream, USER_SOCKPTR(dst), pos, count); } static int snd_korg1212_capture_copy_kernel(struct snd_pcm_substream *substream, int channel, unsigned long pos, void *dst, unsigned long count) { - return snd_korg1212_copy_to(substream, (void __user *)dst, - pos, count, true); + return snd_korg1212_copy_to(substream, KERNEL_SOCKPTR(dst), pos, count); } static const struct snd_pcm_ops snd_korg1212_playback_ops = { -- 2.40.0.1.gaa8946217a0b

3 6

[PATCH v2] ASoC: wm8960: Add DAC filter characteristics selection
by Shengjiu Wang 21 Jul '23

21 Jul '23

Support DAC filter characteristics selection: Normal mode and Sloping stopband. Sloping stopband may have better frequency response. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- changes in v2: - modify capitalise sound/soc/codecs/wm8960.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 366f5d769d6d..5db9b0338396 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -155,6 +155,7 @@ static const char *wm8960_adc_data_output_sel[] = { "Left Data = Right ADC; Right Data = Left ADC", }; static const char *wm8960_dmonomix[] = {"Stereo", "Mono"}; +static const char *wm8960_dacslope[] = {"Normal", "Sloping"}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -165,6 +166,7 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), SOC_ENUM_SINGLE(WM8960_ADDCTL1, 2, 4, wm8960_adc_data_output_sel), SOC_ENUM_SINGLE(WM8960_ADDCTL1, 4, 2, wm8960_dmonomix), + SOC_ENUM_SINGLE(WM8960_DACCTL2, 1, 2, wm8960_dacslope), }; static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -307,6 +309,7 @@ SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", SOC_ENUM("ADC Data Output Select", wm8960_enum[6]), SOC_ENUM("DAC Mono Mix", wm8960_enum[7]), +SOC_ENUM("DAC Filter Characteristics", wm8960_enum[8]), }; static const struct snd_kcontrol_new wm8960_lin_boost[] = { -- 2.34.1

2 1

[PATCH -next] ALSA: seq: remove redundant unsigned comparison to zero
by Wang Weiyang 21 Jul '23

21 Jul '23

Since struct snd_ump_block_info::first_group is unsigned char, comparison to zero is redundant Signed-off-by: Wang Weiyang <wangweiyang2(a)huawei.com> --- sound/core/seq/seq_ump_client.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/core/seq/seq_ump_client.c b/sound/core/seq/seq_ump_client.c index fe21c801af74..f26a1812dfa7 100644 --- a/sound/core/seq/seq_ump_client.c +++ b/sound/core/seq/seq_ump_client.c @@ -298,8 +298,7 @@ static void update_group_attrs(struct seq_ump_client *client) } list_for_each_entry(fb, &client->ump->block_list, list) { - if (fb->info.first_group < 0 || - fb->info.first_group + fb->info.num_groups > SNDRV_UMP_MAX_GROUPS) + if (fb->info.first_group + fb->info.num_groups > SNDRV_UMP_MAX_GROUPS) break; group = &client->groups[fb->info.first_group]; for (i = 0; i < fb->info.num_groups; i++, group++) { -- 2.25.1

2 1

[PATCH v1] ASoC: dt-bindings: wm8904: Convert to dtschema
by Francesco Dolcini 21 Jul '23

21 Jul '23

From: Francesco Dolcini <francesco.dolcini(a)toradex.com> Convert the WM8904 audio CODEC bindings to DT schema. Compared to the original binding #sound-dai-cells and the missing power supplies are added. The latter are all required as described in the datasheet. Datasheet: https://statics.cirrus.com/pubs/proDatasheet/WM8904_Rev4.1.pdf Signed-off-by: Francesco Dolcini <francesco.dolcini(a)toradex.com> --- .../devicetree/bindings/sound/wlf,wm8904.yaml | 74 +++++++++++++++++++ .../devicetree/bindings/sound/wm8904.txt | 33 --------- 2 files changed, 74 insertions(+), 33 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/wlf,wm8904.yaml delete mode 100644 Documentation/devicetree/bindings/sound/wm8904.txt diff --git a/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml new file mode 100644 index 000000000000..329260cf0fa0 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/wlf,wm8904.yaml @@ -0,0 +1,74 @@ +# SPDX-License-Identifier: GPL-2.0-only OR BSD-2-Clause +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/wlf,wm8904.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Wolfson WM8904/WM8912 audio codecs + +maintainers: + - patches(a)opensource.cirrus.com + +description: | + Pins on the device (for linking into audio routes): + IN1L, IN1R, IN2L, IN2R, IN3L, IN3R, HPOUTL, HPOUTR, LINEOUTL, LINEOUTR, + MICBIAS + +properties: + compatible: + enum: + - wlf,wm8904 + - wlf,wm8912 + + reg: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + clocks: + maxItems: 1 + + clock-names: + const: mclk + + AVDD-supply: true + CPVDD-supply: true + DBVDD-supply: true + DCVDD-supply: true + MICVDD-supply: true + +required: + - compatible + - reg + - clocks + - clock-names + - AVDD-supply + - CPVDD-supply + - DBVDD-supply + - DCVDD-supply + - MICVDD-supply + +allOf: + - $ref: dai-common.yaml# + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + codec@1a { + compatible = "wlf,wm8904"; + reg = <0x1a>; + clocks = <&pck0>; + clock-names = "mclk"; + AVDD-supply = <&reg_1p8v>; + CPVDD-supply = <&reg_1p8v>; + DBVDD-supply = <&reg_1p8v>; + DCVDD-supply = <&reg_1p8v>; + MICVDD-supply = <&reg_1p8v>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/wm8904.txt b/Documentation/devicetree/bindings/sound/wm8904.txt deleted file mode 100644 index 66bf261423b9..000000000000 --- a/Documentation/devicetree/bindings/sound/wm8904.txt +++ /dev/null @@ -1,33 +0,0 @@ -WM8904 audio CODEC - -This device supports I2C only. - -Required properties: - - compatible: "wlf,wm8904" or "wlf,wm8912" - - reg: the I2C address of the device. - - clock-names: "mclk" - - clocks: reference to - <Documentation/devicetree/bindings/clock/clock-bindings.txt> - -Pins on the device (for linking into audio routes): - - * IN1L - * IN1R - * IN2L - * IN2R - * IN3L - * IN3R - * HPOUTL - * HPOUTR - * LINEOUTL - * LINEOUTR - * MICBIAS - -Examples: - -codec: wm8904@1a { - compatible = "wlf,wm8904"; - reg = <0x1a>; - clocks = <&pck0>; - clock-names = "mclk"; -}; -- 2.25.1

3 2

[PATCH v1 0/2] ASoC: dt-bindings: nau8822: minor updates
by Francesco Dolcini 21 Jul '23

21 Jul '23

From: Francesco Dolcini <francesco.dolcini(a)toradex.com> Minor updates to NAU8822 DT bindings: - Add #sound-dai-cells - Add MCLK Francesco Dolcini (2): ASoC: dt-bindings: nau8822: Add #sound-dai-cells ASoC: dt-bindings: nau8822: Add MCLK clock .../devicetree/bindings/sound/nuvoton,nau8822.yaml | 12 ++++++++++++ 1 file changed, 12 insertions(+) -- 2.25.1

3 5

Fwd: 6.4 and higher causes audio distortion
by Bagas Sanjaya 21 Jul '23

21 Jul '23

Hi, I notice a regression report on Bugzilla [1]. Quoting from it: > I have a Lenovo ThinkPad X1 Yoga Gen 7 running Arch Linux. Linux 6.4 and higher, cause audio distortion. Sometimes, this occurs to the point that nearly nothing is discernible. This carries over to wired headphones. The issue occurs on the entire mainline 6.4.x kernel series and also the 6.4.3 stable and 6.5 RC1 kernel, which are the latest at the time of writing. The issue occurs on both the Arch distributed kernels, and the mainline kernels. > > Linux kernels 6.3.x are not affected and neither is the 6.1 LTS kernel series which is what I am temporarily using. On Windows 10/11 too, the audio works as it should. This indicates that my hardware is not at fault. Bluetooth audio is not impacted from my testing, either. > > The distortion doesn't start immediately. It either occurs automatically after a random amount of time, or when I increase/decrease the volume, or when I skip forward/backward to a section. In order to stop the distortion, I have to either increase/decrease the volume until it stops, or skip forward/backward until it stops, or restart Pipewire via systemd, however it starts again due to one of the aforementioned reasons. > > At the time of this report, I am running Pipewire 0.3.74 and Wireplumber 0.4.14. This also doesn't seem like a Pipewire/Wireplumber issue, since these same versions work fine on the 6.1 LTS kernels without causing any audio distortion. > > I wrote about this on the Arch Linux forums, too, and seems like at least two other people are facing this issue. Here's the forum post: https://bbs.archlinux.org/viewtopic.php?id=287068 > > Furthermore, I filed a bug report on the Arch Linux Bug Reporter, where they suggested that the issue is a kernel regression and should be reported upstream, here. Here's the bug report that I filed on the Arch Linux Bug Reporter for anyone interested: https://bugs.archlinux.org/task/79081?project=1&pagenum=10 > > I have attached the dmesg outputs of the mainline 6.5 RC1 kernel. > > Here's some audio related hardware information from my device: > > inxi -A > > Audio: > Device-1: Intel Alder Lake PCH-P High Definition Audio > driver: sof-audio-pci-intel-tgl > API: ALSA v: k6.5.0-rc1-1-mainline status: kernel-api > > > pactl info > > Server String: /run/user/1000/pulse/native > Library Protocol Version: 35 > Server Protocol Version: 35 > Is Local: yes > Client Index: 138 > Tile Size: 65472 > User Name: tux > Host Name: NSA-Terminal-4 > Server Name: PulseAudio (on PipeWire 0.3.74) > Server Version: 15.0.0 > Default Sample Specification: float32le 2ch 48000Hz > Default Channel Map: front-left,front-right > Default Sink: alsa_output.pci-0000_00_1f.3-platform-skl_hda_dsp_generic.HiFi__hw_sofhdadsp__sink > Default Source: alsa_input.pci-0000_00_1f.3-platform-skl_hda_dsp_generic.HiFi__hw_sofhdadsp_6__source > Cookie: f9dc:5e7a > > > I can't figure out why this is happening. Kindly ask for any more information that is necessary. Thank you. See Bugzilla for the full thread and attached dmesg. Anyway, I'm adding it to regzbot so that it doesn't fall through cracks unnoticed: #regzbot introduced: v6.3..v6.4 https://bugzilla.kernel.org/show_bug.cgi?id=217673 #regzbot title: PipeWire + Wireplumber audio distortion on Lenovo ThinkPad X1 Yoga Gen 7 #regzbot link: https://bbs.archlinux.org/viewtopic.php?id=287068 #regzbot link: https://bugs.archlinux.org/task/79081?project=1&pagenum=10 Thanks. [1]: https://bugzilla.kernel.org/show_bug.cgi?id=217673 -- An old man doll... just what I always wanted! - Clara

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[PATCH v2 0/5] fix for JD event handling in ClockStop Mode0
by shumingf＠realtek.com 21 Jul '23

21 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> Currently, the codec driver could re-enables SDCA interrupt mask and the bus driver re-enables Imp-defined interrupt mask when the devices are re-attached. If the manager uses ClockStop Mode0 to peripherals, the SDCA/Imp-defined interrupt mask will not be enabled when the system resumes. These patches will fix this issue. -- v2 - modify the commit messages Shuming Fan (5): ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0 ASoC: rt711: fix for JD event handling in ClockStop Mode0 ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0 ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0 ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0 sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- sound/soc/codecs/rt712-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/rt722-sdca-sdw.c | 10 +++++++++- 5 files changed, 43 insertions(+), 5 deletions(-) -- 2.34.1

1 0

[PATCH] ASoC: wm8960: Add DAC filter characteristics selection
by Shengjiu Wang 21 Jul '23

21 Jul '23

Support DAC filter characteristics selection: Normal mode and Sloping stopband. Sloping stopband may have better frequency response. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/wm8960.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 366f5d769d6d..ff1355306b43 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -155,6 +155,7 @@ static const char *wm8960_adc_data_output_sel[] = { "Left Data = Right ADC; Right Data = Left ADC", }; static const char *wm8960_dmonomix[] = {"Stereo", "Mono"}; +static const char *wm8960_dacslope[] = {"Normal", "Sloping"}; static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_DACCTL1, 5, 4, wm8960_polarity), @@ -165,6 +166,7 @@ static const struct soc_enum wm8960_enum[] = { SOC_ENUM_SINGLE(WM8960_ALC3, 8, 2, wm8960_alcmode), SOC_ENUM_SINGLE(WM8960_ADDCTL1, 2, 4, wm8960_adc_data_output_sel), SOC_ENUM_SINGLE(WM8960_ADDCTL1, 4, 2, wm8960_dmonomix), + SOC_ENUM_SINGLE(WM8960_DACCTL2, 1, 2, wm8960_dacslope), }; static const int deemph_settings[] = { 0, 32000, 44100, 48000 }; @@ -307,6 +309,7 @@ SOC_SINGLE_TLV("Right Output Mixer RINPUT3 Volume", SOC_ENUM("ADC Data Output Select", wm8960_enum[6]), SOC_ENUM("DAC Mono Mix", wm8960_enum[7]), +SOC_ENUM("DAC filter characteristics", wm8960_enum[8]), }; static const struct snd_kcontrol_new wm8960_lin_boost[] = { -- 2.34.1

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[PATCH 2/2] ASoC: rt711-sdca: check pin sense when system resumes
by shumingf＠realtek.com 21 Jul '23

21 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> If the jack is unplugged when the system was suspended, the system will believe the jack exists for a while after the system resumes. The codec needs some time, then alerts the host to detect whether the jack exists or not. This patch wants to use the pin sense to detect whether the jack exists or not when the system resumes. It could report the status immediately if the jack was unplugged. It will improve the user experience. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca-sdw.c | 18 ++++++++++++++++++ sound/soc/codecs/rt711-sdca.c | 2 +- sound/soc/codecs/rt711-sdca.h | 2 ++ 3 files changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index cb27b4a98fb6..13a5316ba999 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -12,6 +12,7 @@ #include <linux/soundwire/sdw_registers.h> #include <linux/module.h> #include <linux/pm_runtime.h> +#include <sound/jack.h> #include "rt711-sdca.h" #include "rt711-sdca-sdw.h" @@ -435,6 +436,8 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) struct sdw_slave *slave = dev_to_sdw_dev(dev); struct rt711_sdca_priv *rt711 = dev_get_drvdata(dev); unsigned long time; + unsigned int pin_sense; + int ret; if (!rt711->first_hw_init) return 0; @@ -457,6 +460,21 @@ static int __maybe_unused rt711_sdca_dev_resume(struct device *dev) regcache_sync(rt711->regmap); regcache_cache_only(rt711->mbq_regmap, false); regcache_sync(rt711->mbq_regmap); + + /* + * detect whether jack exists or not + * if the jack is unplugged, we will report it immediately + * if the jack exists, it will take some time to detect, then alert the host + */ + ret = rt711_sdca_index_read(rt711, RT711_VENDOR_REG, RT711_IRQ_FLAG_TABLE2, &pin_sense); + pin_sense = ((pin_sense & 0x100) >> 8); + if (ret < 0) { + dev_err(&slave->dev, "Pin Sense read failed:%d\n", ret); + return ret; + } + if (rt711->hs_jack && !pin_sense) + snd_soc_jack_report(rt711->hs_jack, 0, SND_JACK_HEADSET); + return 0; } diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index f1a93e3ffa1c..29dce5173f90 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -42,7 +42,7 @@ static int rt711_sdca_index_write(struct rt711_sdca_priv *rt711, return ret; } -static int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, +int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, unsigned int nid, unsigned int reg, unsigned int *value) { int ret; diff --git a/sound/soc/codecs/rt711-sdca.h b/sound/soc/codecs/rt711-sdca.h index 11d421e8ab2b..aeedb38a0260 100644 --- a/sound/soc/codecs/rt711-sdca.h +++ b/sound/soc/codecs/rt711-sdca.h @@ -240,4 +240,6 @@ int rt711_sdca_init(struct device *dev, struct regmap *regmap, struct regmap *mbq_regmap, struct sdw_slave *slave); int rt711_sdca_jack_detect(struct rt711_sdca_priv *rt711, bool *hp, bool *mic); +int rt711_sdca_index_read(struct rt711_sdca_priv *rt711, + unsigned int nid, unsigned int reg, unsigned int *value); #endif /* __RT711_SDCA_H__ */ -- 2.34.1

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[PATCH 1/2] ASoC: rt711-sdca: fix key tone missing
by shumingf＠realtek.com 21 Jul '23

21 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> This patch uses a bit to check whether the system was hibernated or not. It could avoid repeating the recalibration and CBJ detection. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt711-sdca-sdw.c | 1 + sound/soc/codecs/rt711-sdca.c | 17 ++++++++++++++++- 2 files changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt711-sdca-sdw.c b/sound/soc/codecs/rt711-sdca-sdw.c index 119e1f9605d7..cb27b4a98fb6 100644 --- a/sound/soc/codecs/rt711-sdca-sdw.c +++ b/sound/soc/codecs/rt711-sdca-sdw.c @@ -101,6 +101,7 @@ static bool rt711_sdca_mbq_volatile_register(struct device *dev, unsigned int re case 0x2000080: case 0x2000081: case 0x2000083: + case 0x200009f: case 0x5800000: case 0x5800001: case 0x5f00001: diff --git a/sound/soc/codecs/rt711-sdca.c b/sound/soc/codecs/rt711-sdca.c index 07640d2f6e56..f1a93e3ffa1c 100644 --- a/sound/soc/codecs/rt711-sdca.c +++ b/sound/soc/codecs/rt711-sdca.c @@ -488,6 +488,12 @@ static void rt711_sdca_jack_init(struct rt711_sdca_priv *rt711) /* set SCP_SDCA_IntMask2[0]=1 */ sdw_write_no_pm(rt711->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); dev_dbg(&rt711->slave->dev, "in %s enable\n", __func__); + + /* trigger GE interrupt */ + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_GE_MODE_RELATED_CTL, 0x0080, 0x0080); + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, + RT711_GE_MODE_RELATED_CTL, 0x0080, 0x0000); } else { /* disable HID 1/2 event */ rt711_sdca_index_update_bits(rt711, RT711_VENDOR_HDA_CTL, @@ -1493,7 +1499,7 @@ int rt711_sdca_io_init(struct device *dev, struct sdw_slave *slave) { struct rt711_sdca_priv *rt711 = dev_get_drvdata(dev); int ret = 0; - unsigned int val; + unsigned int val, hibernation_flag; rt711->disable_irq = false; @@ -1525,6 +1531,11 @@ int rt711_sdca_io_init(struct device *dev, struct sdw_slave *slave) pm_runtime_get_noresume(&slave->dev); + rt711_sdca_index_read(rt711, RT711_VENDOR_REG, 0x9f, &val); + hibernation_flag = (val >> 8); + if (hibernation_flag) + goto _preset_ready_; + rt711_sdca_reset(rt711); rt711_sdca_index_read(rt711, RT711_VENDOR_REG, RT711_JD_PRODUCT_NUM, &val); @@ -1559,6 +1570,10 @@ int rt711_sdca_io_init(struct device *dev, struct sdw_slave *slave) if (rt711->hs_jack) rt711_sdca_jack_init(rt711); + if (!hibernation_flag) + rt711_sdca_index_update_bits(rt711, RT711_VENDOR_REG, 0x9f, 0x0100, 0x0100); + +_preset_ready_: if (rt711->first_hw_init) { regcache_cache_bypass(rt711->regmap, false); regcache_mark_dirty(rt711->regmap); -- 2.34.1

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[PATCH 1/2] ASoC: rt1316: support the different base clock frequency
by shumingf＠realtek.com 21 Jul '23

21 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> This patch configures all PLL settings to support the all of base clock frequencies. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> --- sound/soc/codecs/rt1316-sdw.c | 23 ++++++++++++++++++++++- 1 file changed, 22 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 721821d9e9af..ce2246a2591c 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -76,6 +76,27 @@ static const struct reg_sequence rt1316_blind_write[] = { { 0xc713, 0x06 }, { 0xc714, 0x80 }, { 0xc715, 0x06 }, + { 0xc720, 0x17 }, + { 0xc721, 0x00 }, + { 0xc722, 0x3e }, + { 0xc723, 0x06 }, + { 0xc724, 0x80 }, + { 0xc725, 0x06 }, + { 0xc730, 0x06 }, + { 0xc731, 0x00 }, + { 0xc732, 0x3E }, + { 0xc740, 0x13 }, + { 0xc741, 0x80 }, + { 0xc742, 0x30 }, + { 0xc743, 0x06 }, + { 0xc744, 0x05 }, + { 0xc745, 0x1E }, + { 0xc750, 0x17 }, + { 0xc751, 0x00 }, + { 0xc752, 0x2E }, + { 0xc753, 0x06 }, + { 0xc754, 0x80 }, + { 0xc755, 0x06 }, { 0xc702, 0x0a }, { 0xc703, 0x0a }, { 0xc001, 0x45 }, @@ -88,7 +109,7 @@ static const struct reg_sequence rt1316_blind_write[] = { { 0xc008, 0x11 }, { 0xc009, 0x00 }, - { 0x2f0a, 0x00 }, + { 0x2f0a, 0x01 }, { 0xd101, 0xf0 }, { 0xd103, 0x9b }, { 0x2f36, 0x8e }, -- 2.34.1

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[PATCH v2 0/4] ALSA: Cleanup with snd_ctl_add_followers()
by Takashi Iwai 21 Jul '23

21 Jul '23

This is another set of cleanup patches: introduce a new standard helper for adding vmaster control elements, and replace the open code in a few drivers with it. Takashi === v1->v2: Cleanup of ac97 patch to drop unnecessary check of the error condition === Takashi Iwai (4): ALSA: vmaster: Add snd_ctl_add_followers() helper ALSA: ac97: Use the standard snd_ctl_add_followers() helper ALSA: ca0106: Use the standard snd_ctl_add_followers() helper ALSA: ice1712: Use the standard snd_ctl_add_followers() helper include/sound/control.h | 3 +++ sound/core/vmaster.c | 28 ++++++++++++++++++++++++++++ sound/pci/ac97/ac97_patch.c | 16 +--------------- sound/pci/ca0106/ca0106_mixer.c | 19 ++++++------------- sound/pci/ice1712/juli.c | 19 +++---------------- sound/pci/ice1712/quartet.c | 15 +++------------ 6 files changed, 44 insertions(+), 56 deletions(-) -- 2.35.3

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[GIT PULL] ASoC fixes for v6.5-rc1-2
by Mark Brown 20 Jul '23

20 Jul '23

The following changes since commit fdf0eaf11452d72945af31804e2a1048ee1b574c: Linux 6.5-rc2 (2023-07-16 15:10:37 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc1-2 for you to fetch changes up to 269f399dc19f0e5c51711c3ba3bd06e0ef6ef403: ASoC: fsl_sai: Disable bit clock with transmitter (2023-07-17 19:15:07 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.5 A few more fixes for v6.5, just small driver specific ones. ---------------------------------------------------------------- Dan Carpenter (1): ASoC: SOF: ipc3-dtrace: uninitialized data in dfsentry_trace_filter_write() Derek Fang (1): ASoC: rt5640: Fix the issue of speaker noise Fabio Estevam (1): ASoC: fsl_sai: Revert "ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode" Geert Uytterhoeven (1): ASoC: codecs: SND_SOC_WCD934X should select REGMAP_IRQ Johan Hovold (15): ASoC: codecs: wcd938x: fix codec initialisation race ASoC: codecs: wcd938x: fix mbhc impedance loglevel ASoC: codecs: wcd938x: drop inline keywords ASoC: codecs: wcd938x: use dev_printk() for impedance logging ASoC: codecs: wcd934x: demote impedance printk ASoC: codecs: wcd934x: drop inline keywords ASoC: codecs: wcd938x: fix soundwire initialisation race ASoC: codecs: wcd938x: fix missing mbhc init error handling ASoC: qdsp6: audioreach: fix topology probe deferral ASoC: codecs: wcd938x: fix missing clsh ctrl error handling ASoC: codecs: wcd938x: fix resource leaks on component remove ASoC: codecs: wcd934x: fix resource leaks on component remove ASoC: codecs: wcd-mbhc-v2: fix resource leaks on component remove ASoC: topology: suppress probe deferral errors ASoC: core: suppress probe deferral errors Mario Limonciello (1): ASoC: amd: ps: Fix extraneous error messages Mark Brown (3): ASoC: codecs: wcd938x/wcd934x: loglevel fix and ASoC/soundwire/qdsp6/wcd: fix leaks and probe deferral ASoC: Merge v6.5-rc2 Martin Povišer (1): MAINTAINERS: Redo addition of ssm3515 to APPLE SOUND Mastan Katragadda (1): ASoC: SOF: amd: add revision check for sending sha dma completion command Matthias Reichl (1): ASoC: hdmi-codec: fix channel info for compressed formats Matus Gajdos (1): ASoC: fsl_sai: Disable bit clock with transmitter Nathan Chancellor (1): ASoC: cs35l45: Select REGMAP_IRQ Rob Herring (1): ASoC: dt-bindings: audio-graph-card2: Drop incomplete example Rohit kumar (1): ASoC: dt-bindings: Update maintainer email id Sameer Pujar (1): ASoC: rt5640: Fix sleep in atomic context Sheetal (2): ASoC: tegra: Fix AMX byte map ASoC: tegra: Fix ADX byte map Shuming Fan (1): ASoC: rt5645: check return value after reading device id Srinivas Kandagatla (4): ASoC: qdsp6: q6apm: use dai link pcm id as pcm device number ASoC: qcom: q6afe-dai: fix Display Port Playback stream name ASoC: codecs: wcd938x: fix dB range for HPHL and HPHR ASoC: qcom: q6apm: do not close GPR port before closing graph Thomas Petazzoni (1): ASoC: cs42l51: fix driver to properly autoload with automatic module loading Vijendar Mukunda (6): ASoC: amd: ps: add comments for DMA irq bits mapping ASoC: amd: ps: add fix for dma irq mask for rx streams for SDW0 instance ASoC: amd: ps: fix for position register set for AUDIO0 RX stream ASoC: amd: ps: add comments for DMA register mapping ASoC: amd: ps: fix byte count return value for invalid SoundWire manager instance ASoC: amd: acp: fix for invalid dai id handling in acp_get_byte_count() .../bindings/sound/audio-graph-card2.yaml | 20 +---- .../bindings/sound/google,sc7180-trogdor.yaml | 2 +- .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 2 +- MAINTAINERS | 2 + sound/soc/amd/acp/amd.h | 7 +- sound/soc/amd/ps/acp63.h | 22 ++++- sound/soc/amd/ps/pci-ps.c | 4 +- sound/soc/amd/ps/ps-sdw-dma.c | 16 +++- sound/soc/codecs/Kconfig | 2 + sound/soc/codecs/cs42l51-i2c.c | 6 ++ sound/soc/codecs/cs42l51.c | 7 -- sound/soc/codecs/cs42l51.h | 1 - sound/soc/codecs/rt5640.c | 13 +-- sound/soc/codecs/rt5645.c | 6 +- sound/soc/codecs/wcd-mbhc-v2.c | 57 +++++++++---- sound/soc/codecs/wcd934x.c | 20 ++++- sound/soc/codecs/wcd938x.c | 99 ++++++++++++++++++---- sound/soc/fsl/fsl_sai.c | 8 +- sound/soc/fsl/fsl_sai.h | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- sound/soc/qcom/qdsp6/q6apm-dai.c | 1 + sound/soc/qcom/qdsp6/q6apm.c | 7 +- sound/soc/qcom/qdsp6/topology.c | 4 +- sound/soc/soc-core.c | 6 +- sound/soc/soc-topology.c | 10 ++- sound/soc/sof/amd/acp.c | 10 ++- sound/soc/sof/ipc3-dtrace.c | 9 +- sound/soc/tegra/tegra210_adx.c | 34 +++++--- sound/soc/tegra/tegra210_amx.c | 40 +++++---- 29 files changed, 282 insertions(+), 136 deletions(-)

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[PATCH 00/11] ALSA: Cleanup with snd_ctl_find_id_mixer()
by Takashi Iwai 20 Jul '23

20 Jul '23

Quite a few drivers have the very same open code to find a control element from the given name string. Provide a standard helper and replace with it for cleanup. Takashi === Takashi Iwai (11): ALSA: control: Introduce snd_ctl_find_id_mixer() ALSA: ca0106: Simplify with snd_ctl_find_id_mixer() ALSA: cs46xx: Simplify with snd_ctl_find_id_mixer() ALSA: emu10k1: Simplify with snd_ctl_find_id_mixer() ALSA: es1968: Simplify with snd_ctl_find_id_mixer() ALSA: ice1712: Simplify with snd_ctl_find_id_mixer() ALSA: maestro3: Simplify with snd_ctl_find_id_mixer() ALSA: via82xx: Simplify with snd_ctl_find_id_mixer() ALSA: cmipci: Simplify with snd_ctl_find_id_mixer() ASoC: mediatek: mt8188: Simplify with snd_ctl_find_id_mixer() ALSA: ac97: Simplify with snd_ctl_find_id_mixer() include/sound/control.h | 22 ++++++++++++++++++++++ sound/pci/ac97/ac97_patch.c | 6 +----- sound/pci/ca0106/ca0106_mixer.c | 15 +++------------ sound/pci/cmipci.c | 6 +----- sound/pci/cs46xx/cs46xx_lib.c | 7 ++----- sound/pci/emu10k1/emumixer.c | 11 +---------- sound/pci/es1968.c | 15 ++++----------- sound/pci/ice1712/juli.c | 13 ++----------- sound/pci/ice1712/psc724.c | 19 ++++++++----------- sound/pci/ice1712/quartet.c | 13 ++----------- sound/pci/ice1712/wm8776.c | 6 +----- sound/pci/maestro3.c | 15 ++++----------- sound/pci/via82xx.c | 6 +----- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 12 +----------- 14 files changed, 53 insertions(+), 113 deletions(-) -- 2.35.3

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[PATCH v3 0/6] ALSA: emu10k1: add support for high-bitrate modes of E-MU cards
by Oswald Buddenhagen 20 Jul '23

20 Jul '23

This series is what all the work was about: support the "dual-/quad-pumped" modes of the E-MU cards. --- i'm posting this so the final version (unless some actual bugs are found) is on record, without expectation of it being applied. i'll document its availability on the wiki and the related bug report instead. the first three patches appear uncontroversial, and you may apply them if you want to. however, the added value is somewhat limited without the last three patches, so it's probably not worth bloating the driver with them. --- v3: - rebased onto pulled ahead series --- This patch series needs to be applied on top of the patch titled "ALSA: add snd_ctl_add_locked()". Oswald Buddenhagen (6): ALSA: emu10k1: introduce alternative E-MU D.A.S. mode ALSA: emu10k1: improve mixer control naming in E-MU D.A.S. mode ALSA: emu10k1: make playback in E-MU D.A.S. mode 32-bit ALSA: emu10k1: add support for 2x/4x word clocks in E-MU D.A.S. mode ALSA: emu10k1: add high-rate capture in E-MU D.A.S. mode ALSA: emu10k1: add high-rate playback in E-MU D.A.S. mode include/sound/emu10k1.h | 11 + sound/pci/emu10k1/emu10k1.c | 29 +- sound/pci/emu10k1/emu10k1_main.c | 30 +- sound/pci/emu10k1/emufx.c | 109 +++- sound/pci/emu10k1/emumixer.c | 902 +++++++++++++++++++++++++++---- sound/pci/emu10k1/emupcm.c | 417 ++++++++++++-- sound/pci/emu10k1/emuproc.c | 5 + sound/pci/emu10k1/io.c | 30 +- sound/pci/emu10k1/voice.c | 6 + 9 files changed, 1351 insertions(+), 188 deletions(-) base-commit: 3c04ccbdf86e7e93e06e68f1915e39eb55afb0d0 prerequisite-patch-id: 45677e4f73ea654bda3258c1143c8027173e4b5c -- 2.40.0.152.g15d061e6df

1 7

[PATCH v3] ALSA: add snd_ctl_add_locked()
by Oswald Buddenhagen 20 Jul '23

20 Jul '23

This is in fact more symmetrical to snd_ctl_remove() than snd_ctl_add() is - the former could be named snd_ctl_remove_locked() just as well. One may argue that this is going in the wrong direction, as drivers have no business managing the lock. This may be true in principle, but in practice the vast majority of controls is created even before the device was registered, and therefore before any locking is necessary at all. That means that an even more radical approach of changing snd_ctl_add() to do no locking, and converting the call sites that actually need locking to a new function, would better match reality, and would be somewhat more efficient at that. However, that seems a bit risky and way too much work. This will be used to dynamically change the available controls from another control's put() callback, which is already locked. One might want to add snd_ctl_replace_locked() for completeness, but I have no use for it now. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- applying this upstream would simplify applying the emu10k1 high bit-rate patchset locally, as it would limit the affected modules to the driver itself. v3: - fixed typo in commit message v2: - extended commit message --- include/sound/control.h | 1 + sound/core/control.c | 31 +++++++++++++++++++++++++++++++ 2 files changed, 32 insertions(+) diff --git a/include/sound/control.h b/include/sound/control.h index cc3dcc6cfb0f..d4e210831a38 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -134,6 +134,7 @@ void snd_ctl_notify_one(struct snd_card * card, unsigned int mask, struct snd_kc struct snd_kcontrol *snd_ctl_new1(const struct snd_kcontrol_new * kcontrolnew, void * private_data); void snd_ctl_free_one(struct snd_kcontrol * kcontrol); int snd_ctl_add(struct snd_card * card, struct snd_kcontrol * kcontrol); +int snd_ctl_add_locked(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_remove(struct snd_card * card, struct snd_kcontrol * kcontrol); int snd_ctl_replace(struct snd_card *card, struct snd_kcontrol *kcontrol, bool add_on_replace); int snd_ctl_remove_id(struct snd_card * card, struct snd_ctl_elem_id *id); diff --git a/sound/core/control.c b/sound/core/control.c index 8386b53acdcd..1a0bb76dd8e7 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -527,6 +527,27 @@ static int snd_ctl_add_replace(struct snd_card *card, return err; } +static int snd_ctl_add_replace_locked(struct snd_card *card, + struct snd_kcontrol *kcontrol, + enum snd_ctl_add_mode mode) +{ + int err = -EINVAL; + + if (! kcontrol) + return err; + if (snd_BUG_ON(!card || !kcontrol->info)) + goto error; + + err = __snd_ctl_add_replace(card, kcontrol, mode); + if (err < 0) + goto error; + return 0; + + error: + snd_ctl_free_one(kcontrol); + return err; +} + /** * snd_ctl_add - add the control instance to the card * @card: the card instance @@ -547,6 +568,16 @@ int snd_ctl_add(struct snd_card *card, struct snd_kcontrol *kcontrol) } EXPORT_SYMBOL(snd_ctl_add); +/** + * snd_ctl_add_locked - same as snd_ctl_add(), but card->controls_rwsem + * is expected to be already locked if necessary. + */ +int snd_ctl_add_locked(struct snd_card *card, struct snd_kcontrol *kcontrol) +{ + return snd_ctl_add_replace_locked(card, kcontrol, CTL_ADD_EXCLUSIVE); +} +EXPORT_SYMBOL(snd_ctl_add_locked); + /** * snd_ctl_replace - replace the control instance of the card * @card: the card instance -- 2.40.0.152.g15d061e6df

2 2

[PATCH 0/4] ALSA: Cleanup with snd_ctl_add_followers()
by Takashi Iwai 20 Jul '23

20 Jul '23

This is another set of cleanup patches: introduce a new standard helper for adding vmaster control elements, and replace the open code in a few drivers with it. Takashi === Takashi Iwai (4): ALSA: vmaster: Add snd_ctl_add_followers() helper ALSA: ac97: Use the standard snd_ctl_add_followers() helper ALSA: ca0106: Use the standard snd_ctl_add_followers() helper ALSA: ice1712: Use the standard snd_ctl_add_followers() helper include/sound/control.h | 3 +++ sound/core/vmaster.c | 28 ++++++++++++++++++++++++++++ sound/pci/ac97/ac97_patch.c | 16 +++------------- sound/pci/ca0106/ca0106_mixer.c | 19 ++++++------------- sound/pci/ice1712/juli.c | 19 +++---------------- sound/pci/ice1712/quartet.c | 15 +++------------ 6 files changed, 46 insertions(+), 54 deletions(-) -- 2.35.3

2 6

[PATCH 1/2] ALSA: hda/realtek: Fix generic fixup definition for cs35l41 amp
by Vitaly Rodionov 20 Jul '23

20 Jul '23

Generic fixup for CS35L41 amplifies should not have vendor specific chained fixup. For ThinkPad laptops with led issue, we can just add specific fixup. Fixes: a6ac60b36dade (ALSA: hda/realtek: Fix mute led issue on thinkpad with cs35l41 s-codec) Signed-off-by: Vitaly Rodionov <vitalyr(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 25 +++++++++++++++---------- 1 file changed, 15 insertions(+), 10 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e2f8b608de82..56cbf5b38d48 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7213,6 +7213,7 @@ enum { ALC287_FIXUP_YOGA9_14IAP7_BASS_SPK_PIN, ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS, ALC236_FIXUP_DELL_DUAL_CODECS, + ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9145,8 +9146,6 @@ static const struct hda_fixup alc269_fixups[] = { [ALC287_FIXUP_CS35L41_I2C_2] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_i2c_two, - .chained = true, - .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, [ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, @@ -9283,6 +9282,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE, }, + [ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_i2c_two, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9810,14 +9815,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x22be, "Thinkpad X1 Carbon 8th", ALC285_FIXUP_THINKPAD_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x22c1, "Thinkpad P1 Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), SND_PCI_QUIRK(0x17aa, 0x22c2, "Thinkpad X1 Extreme Gen 3", ALC285_FIXUP_THINKPAD_NO_BASS_SPK_HEADSET_JACK), - SND_PCI_QUIRK(0x17aa, 0x22f1, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x22f2, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x22f3, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x2316, "Thinkpad P1 Gen 6", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x2317, "Thinkpad P1 Gen 6", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x2318, "Thinkpad Z13 Gen2", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x2319, "Thinkpad Z16 Gen2", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x17aa, 0x231a, "Thinkpad Z16 Gen2", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x22f1, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x22f2, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x22f3, "Thinkpad", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x2316, "Thinkpad P1 Gen 6", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x2317, "Thinkpad P1 Gen 6", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x2318, "Thinkpad Z13 Gen2", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x2319, "Thinkpad Z16 Gen2", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), + SND_PCI_QUIRK(0x17aa, 0x231a, "Thinkpad Z16 Gen2", ALC287_FIXUP_CS35L41_I2C_2_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.34.1

2 2

[PATCH 1/2] ASoC: codecs: tx-macro: split widgets per different LPASS versions
by Krzysztof Kozlowski 20 Jul '23

20 Jul '23

TX macro codec differs slightly between different Qualcomm Low Power Audio SubSystem (LPASS) block versions. In LPASS version 9.2 the register responsible for TX SMIC MUXn muxes is different, thus to properly support it, the driver needs to register different widgets per different LPASS version. Prepare for supporting this register difference by refactoring existing code: 1. Move few widgets (TX SMIC MUXn, TX SWR_ADCn, TX SWR_DMICn) out of common 'tx_macro_dapm_widgets[]' array to a new per-variant specific array 'tx_macro_dapm_widgets_v9[]'. 2. Move also related audio routes into new array. 3. Store pointers to these variant-specific arrays in new variant-data structure 'tx_macro_data'. 4. Add variant-specific widgets and routes in component probe, instead of driver probe. The change should have no real impact, except re-shuffling code and registering some widgets and audio routes in component probe, instead of driver probe. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-macro-common.h | 5 + sound/soc/codecs/lpass-tx-macro.c | 219 +++++++++++++++++--------- 2 files changed, 147 insertions(+), 77 deletions(-) diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h index 4eb886565ea3..6376a02b373a 100644 --- a/sound/soc/codecs/lpass-macro-common.h +++ b/sound/soc/codecs/lpass-macro-common.h @@ -9,6 +9,11 @@ /* NPL clock is expected */ #define LPASS_MACRO_FLAG_HAS_NPL_CLOCK BIT(0) +enum lpass_version { + LPASS_VER_9_0_0, + LPASS_VER_11_0_0, +}; + struct lpass_macro { struct device *macro_pd; struct device *dcodec_pd; diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index de978c3d70b7..0da0bbf80eee 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -253,8 +253,18 @@ struct hpf_work { struct delayed_work dwork; }; +struct tx_macro_data { + unsigned int flags; + unsigned int ver; + const struct snd_soc_dapm_widget *extra_widgets; + size_t extra_widgets_num; + const struct snd_soc_dapm_route *extra_routes; + size_t extra_routes_num; +}; + struct tx_macro { struct device *dev; + const struct tx_macro_data *data; struct snd_soc_component *component; struct hpf_work tx_hpf_work[NUM_DECIMATORS]; struct tx_mute_work tx_mute_dwork[NUM_DECIMATORS]; @@ -1235,53 +1245,6 @@ static const struct snd_kcontrol_new tx_dec5_mux = SOC_DAPM_ENUM("tx_dec5", tx_d static const struct snd_kcontrol_new tx_dec6_mux = SOC_DAPM_ENUM("tx_dec6", tx_dec6_enum); static const struct snd_kcontrol_new tx_dec7_mux = SOC_DAPM_ENUM("tx_dec7", tx_dec7_enum); -static const char * const smic_mux_text[] = { - "ZERO", "ADC0", "ADC1", "ADC2", "ADC3", "SWR_DMIC0", - "SWR_DMIC1", "SWR_DMIC2", "SWR_DMIC3", "SWR_DMIC4", - "SWR_DMIC5", "SWR_DMIC6", "SWR_DMIC7" -}; - -static SOC_ENUM_SINGLE_DECL(tx_smic0_enum, CDC_TX_INP_MUX_ADC_MUX0_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic1_enum, CDC_TX_INP_MUX_ADC_MUX1_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic2_enum, CDC_TX_INP_MUX_ADC_MUX2_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic3_enum, CDC_TX_INP_MUX_ADC_MUX3_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic4_enum, CDC_TX_INP_MUX_ADC_MUX4_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic5_enum, CDC_TX_INP_MUX_ADC_MUX5_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic6_enum, CDC_TX_INP_MUX_ADC_MUX6_CFG0, - 0, smic_mux_text); - -static SOC_ENUM_SINGLE_DECL(tx_smic7_enum, CDC_TX_INP_MUX_ADC_MUX7_CFG0, - 0, smic_mux_text); - -static const struct snd_kcontrol_new tx_smic0_mux = SOC_DAPM_ENUM_EXT("tx_smic0", tx_smic0_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic1_mux = SOC_DAPM_ENUM_EXT("tx_smic1", tx_smic1_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic2_mux = SOC_DAPM_ENUM_EXT("tx_smic2", tx_smic2_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic3_mux = SOC_DAPM_ENUM_EXT("tx_smic3", tx_smic3_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic4_mux = SOC_DAPM_ENUM_EXT("tx_smic4", tx_smic4_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic5_mux = SOC_DAPM_ENUM_EXT("tx_smic5", tx_smic5_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic6_mux = SOC_DAPM_ENUM_EXT("tx_smic6", tx_smic6_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); -static const struct snd_kcontrol_new tx_smic7_mux = SOC_DAPM_ENUM_EXT("tx_smic7", tx_smic7_enum, - snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); - static const char * const dmic_mux_text[] = { "ZERO", "DMIC0", "DMIC1", "DMIC2", "DMIC3", "DMIC4", "DMIC5", "DMIC6", "DMIC7" @@ -1427,15 +1390,6 @@ static const struct snd_soc_dapm_widget tx_macro_dapm_widgets[] = { SND_SOC_DAPM_MIXER("TX_AIF3_CAP Mixer", SND_SOC_NOPM, TX_MACRO_AIF3_CAP, 0, tx_aif3_cap_mixer, ARRAY_SIZE(tx_aif3_cap_mixer)), - SND_SOC_DAPM_MUX("TX SMIC MUX0", SND_SOC_NOPM, 0, 0, &tx_smic0_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX1", SND_SOC_NOPM, 0, 0, &tx_smic1_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX2", SND_SOC_NOPM, 0, 0, &tx_smic2_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX3", SND_SOC_NOPM, 0, 0, &tx_smic3_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX4", SND_SOC_NOPM, 0, 0, &tx_smic4_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX5", SND_SOC_NOPM, 0, 0, &tx_smic5_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX6", SND_SOC_NOPM, 0, 0, &tx_smic6_mux), - SND_SOC_DAPM_MUX("TX SMIC MUX7", SND_SOC_NOPM, 0, 0, &tx_smic7_mux), - SND_SOC_DAPM_MUX("TX DMIC MUX0", SND_SOC_NOPM, 4, 0, &tx_dmic0_mux), SND_SOC_DAPM_MUX("TX DMIC MUX1", SND_SOC_NOPM, 4, 0, &tx_dmic1_mux), SND_SOC_DAPM_MUX("TX DMIC MUX2", SND_SOC_NOPM, 4, 0, &tx_dmic2_mux), @@ -1445,18 +1399,6 @@ static const struct snd_soc_dapm_widget tx_macro_dapm_widgets[] = { SND_SOC_DAPM_MUX("TX DMIC MUX6", SND_SOC_NOPM, 4, 0, &tx_dmic6_mux), SND_SOC_DAPM_MUX("TX DMIC MUX7", SND_SOC_NOPM, 4, 0, &tx_dmic7_mux), - SND_SOC_DAPM_INPUT("TX SWR_ADC0"), - SND_SOC_DAPM_INPUT("TX SWR_ADC1"), - SND_SOC_DAPM_INPUT("TX SWR_ADC2"), - SND_SOC_DAPM_INPUT("TX SWR_ADC3"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC0"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC1"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC2"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC3"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC4"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC5"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC6"), - SND_SOC_DAPM_INPUT("TX SWR_DMIC7"), SND_SOC_DAPM_INPUT("TX DMIC0"), SND_SOC_DAPM_INPUT("TX DMIC1"), SND_SOC_DAPM_INPUT("TX DMIC2"), @@ -1577,7 +1519,81 @@ static const struct snd_soc_dapm_route tx_audio_map[] = { {"TX DMIC MUX0", "DMIC5", "TX DMIC5"}, {"TX DMIC MUX0", "DMIC6", "TX DMIC6"}, {"TX DMIC MUX0", "DMIC7", "TX DMIC7"}, +}; +/* Controls and routes specific to LPASS <= v9.0.0 */ +static const char * const smic_mux_text_v9[] = { + "ZERO", "ADC0", "ADC1", "ADC2", "ADC3", "SWR_DMIC0", + "SWR_DMIC1", "SWR_DMIC2", "SWR_DMIC3", "SWR_DMIC4", + "SWR_DMIC5", "SWR_DMIC6", "SWR_DMIC7" +}; + +static SOC_ENUM_SINGLE_DECL(tx_smic0_enum_v9, CDC_TX_INP_MUX_ADC_MUX0_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic1_enum_v9, CDC_TX_INP_MUX_ADC_MUX1_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic2_enum_v9, CDC_TX_INP_MUX_ADC_MUX2_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic3_enum_v9, CDC_TX_INP_MUX_ADC_MUX3_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic4_enum_v9, CDC_TX_INP_MUX_ADC_MUX4_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic5_enum_v9, CDC_TX_INP_MUX_ADC_MUX5_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic6_enum_v9, CDC_TX_INP_MUX_ADC_MUX6_CFG0, + 0, smic_mux_text_v9); + +static SOC_ENUM_SINGLE_DECL(tx_smic7_enum_v9, CDC_TX_INP_MUX_ADC_MUX7_CFG0, + 0, smic_mux_text_v9); + +static const struct snd_kcontrol_new tx_smic0_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic0", tx_smic0_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic1_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic1", tx_smic1_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic2_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic2", tx_smic2_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic3_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic3", tx_smic3_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic4_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic4", tx_smic4_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic5_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic5", tx_smic5_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic6_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic6", tx_smic6_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); +static const struct snd_kcontrol_new tx_smic7_mux_v9 = SOC_DAPM_ENUM_EXT("tx_smic7", tx_smic7_enum_v9, + snd_soc_dapm_get_enum_double, tx_macro_put_dec_enum); + +static const struct snd_soc_dapm_widget tx_macro_dapm_widgets_v9[] = { + SND_SOC_DAPM_MUX("TX SMIC MUX0", SND_SOC_NOPM, 0, 0, &tx_smic0_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX1", SND_SOC_NOPM, 0, 0, &tx_smic1_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX2", SND_SOC_NOPM, 0, 0, &tx_smic2_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX3", SND_SOC_NOPM, 0, 0, &tx_smic3_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX4", SND_SOC_NOPM, 0, 0, &tx_smic4_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX5", SND_SOC_NOPM, 0, 0, &tx_smic5_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX6", SND_SOC_NOPM, 0, 0, &tx_smic6_mux_v9), + SND_SOC_DAPM_MUX("TX SMIC MUX7", SND_SOC_NOPM, 0, 0, &tx_smic7_mux_v9), + + SND_SOC_DAPM_INPUT("TX SWR_ADC0"), + SND_SOC_DAPM_INPUT("TX SWR_ADC1"), + SND_SOC_DAPM_INPUT("TX SWR_ADC2"), + SND_SOC_DAPM_INPUT("TX SWR_ADC3"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC0"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC1"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC2"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC3"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC4"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC5"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC6"), + SND_SOC_DAPM_INPUT("TX SWR_DMIC7"), +}; + +static const struct snd_soc_dapm_route tx_audio_map_v9[] = { {"TX DEC0 MUX", "SWR_MIC", "TX SMIC MUX0"}, {"TX SMIC MUX0", NULL, "TX_SWR_CLK"}, {"TX SMIC MUX0", "ADC0", "TX SWR_ADC0"}, @@ -1823,10 +1839,41 @@ static const struct snd_kcontrol_new tx_macro_snd_controls[] = { tx_macro_get_bcs, tx_macro_set_bcs), }; +static int tx_macro_component_extend(struct snd_soc_component *comp) +{ + struct snd_soc_dapm_context *dapm = snd_soc_component_get_dapm(comp); + struct tx_macro *tx = snd_soc_component_get_drvdata(comp); + int ret; + + if (tx->data->extra_widgets_num) { + ret = snd_soc_dapm_new_controls(dapm, tx->data->extra_widgets, + tx->data->extra_widgets_num); + if (ret) { + dev_err(tx->dev, "failed to add extra widgets: %d\n", ret); + return ret; + } + } + + if (tx->data->extra_routes_num) { + ret = snd_soc_dapm_add_routes(dapm, tx->data->extra_routes, + tx->data->extra_routes_num); + if (ret) { + dev_err(tx->dev, "failed to add extra routes: %d\n", ret); + return ret; + } + } + + return 0; +} + static int tx_macro_component_probe(struct snd_soc_component *comp) { struct tx_macro *tx = snd_soc_component_get_drvdata(comp); - int i; + int i, ret; + + ret = tx_macro_component_extend(comp); + if (ret) + return ret; snd_soc_component_init_regmap(comp, tx->regmap); @@ -1954,17 +2001,16 @@ static int tx_macro_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct device_node *np = dev->of_node; - kernel_ulong_t flags; struct tx_macro *tx; void __iomem *base; int ret, reg; - flags = (kernel_ulong_t)device_get_match_data(dev); - tx = devm_kzalloc(dev, sizeof(*tx), GFP_KERNEL); if (!tx) return -ENOMEM; + tx->data = device_get_match_data(dev); + tx->macro = devm_clk_get_optional(dev, "macro"); if (IS_ERR(tx->macro)) return PTR_ERR(tx->macro); @@ -1977,7 +2023,7 @@ static int tx_macro_probe(struct platform_device *pdev) if (IS_ERR(tx->mclk)) return PTR_ERR(tx->mclk); - if (flags & LPASS_MACRO_FLAG_HAS_NPL_CLOCK) { + if (tx->data->flags & LPASS_MACRO_FLAG_HAS_NPL_CLOCK) { tx->npl = devm_clk_get(dev, "npl"); if (IS_ERR(tx->npl)) return PTR_ERR(tx->npl); @@ -2155,21 +2201,40 @@ static const struct dev_pm_ops tx_macro_pm_ops = { SET_RUNTIME_PM_OPS(tx_macro_runtime_suspend, tx_macro_runtime_resume, NULL) }; +static const struct tx_macro_data lpass_ver_9 = { + .flags = LPASS_MACRO_FLAG_HAS_NPL_CLOCK, + .ver = LPASS_VER_9_0_0, + .extra_widgets = tx_macro_dapm_widgets_v9, + .extra_widgets_num = ARRAY_SIZE(tx_macro_dapm_widgets_v9), + .extra_routes = tx_audio_map_v9, + .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9), +}; + +static const struct tx_macro_data lpass_ver_11 = { + .flags = 0, + .ver = LPASS_VER_11_0_0, + .extra_widgets = tx_macro_dapm_widgets_v9, + .extra_widgets_num = ARRAY_SIZE(tx_macro_dapm_widgets_v9), + .extra_routes = tx_audio_map_v9, + .extra_routes_num = ARRAY_SIZE(tx_audio_map_v9), +}; + static const struct of_device_id tx_macro_dt_match[] = { { .compatible = "qcom,sc7280-lpass-tx-macro", - .data = (void *)LPASS_MACRO_FLAG_HAS_NPL_CLOCK, + .data = &lpass_ver_9, }, { .compatible = "qcom,sm8250-lpass-tx-macro", - .data = (void *)LPASS_MACRO_FLAG_HAS_NPL_CLOCK, + .data = &lpass_ver_9, }, { .compatible = "qcom,sm8450-lpass-tx-macro", - .data = (void *)LPASS_MACRO_FLAG_HAS_NPL_CLOCK, + .data = &lpass_ver_9, }, { .compatible = "qcom,sm8550-lpass-tx-macro", + .data = &lpass_ver_11, }, { .compatible = "qcom,sc8280xp-lpass-tx-macro", - .data = (void *)LPASS_MACRO_FLAG_HAS_NPL_CLOCK, + .data = &lpass_ver_9, }, { } }; -- 2.34.1

2 2

[PATCH 00/11] ALSA: Make control API taking controls_rwsem consistently
by Takashi Iwai 20 Jul '23

20 Jul '23

A few ALSA control API helpers like snd_ctl_rename(), snd_ctl_remove() and snd_ctl_find_*() suppose the callers taking card->controls_rwsem. But it's error-prone and fragile. This patch set tries to change those API functions to take the card->controls>rwsem internally by themselves, so that the drivers don't need to take care of lockings. After applying this patch set, only a couple of places still touch card->controls_rwsem (which are OK-ish as they need for traversing the control linked list). Takashi === Takashi Iwai (11): ALSA: control: Take card->controls_rwsem in snd_ctl_rename() staging: greybus: audio_helper: Use snd_ctl_remove_id() ASoC: atmel: mchp-pdmc: Use snd_ctl_remove_id() ALSA: control: Take controls_rwsem lock in snd_ctl_remove() ALSA: control: Add lockdep warning to internal functions ASoC: sigmadsp: Simplify with snd_ctl_activate_id() staging: greybus: Avoid abusing controls_rwsem ALSA: control: Make snd_ctl_find_id() argument const ALSA: control: Introduce unlocked version for snd_ctl_find_*() helpers ALSA: control: Take lock in snd_ctl_find_id() and snd_ctl_find_numid() ALSA: emu10k1: Go back and simplify with snd_ctl_find_id() drivers/staging/greybus/audio_codec.c | 18 ++-- drivers/staging/greybus/audio_codec.h | 1 + drivers/staging/greybus/audio_helper.c | 20 +--- include/sound/control.h | 6 +- sound/core/control.c | 126 ++++++++++++++++++++----- sound/core/control_compat.c | 2 +- sound/core/control_led.c | 2 +- sound/core/jack.c | 2 - sound/core/oss/mixer_oss.c | 10 +- sound/core/pcm.c | 2 - sound/isa/sb/emu8000.c | 2 - sound/isa/sb/sb16_csp.c | 2 - sound/pci/emu10k1/emufx.c | 5 - sound/pci/hda/hda_codec.c | 2 - sound/soc/atmel/mchp-pdmc.c | 12 +-- sound/soc/codecs/sigmadsp.c | 25 +---- sound/soc/soc-topology.c | 3 - 17 files changed, 129 insertions(+), 111 deletions(-) -- 2.35.3

5 19

Re: [BUG REPORT] sound: pci: ctxfi: htimestamp dose not work on EMU20k during recording
by Takashi Iwai 20 Jul '23

20 Jul '23

On Wed, 19 Jul 2023 12:28:28 +0200, Carl Klemm wrote: > > Hi > > so avail seams to stop updateing. i instrumented pipewire here: > https://gitlab.freedesktop.org/pipewire/pipewire/-/blob/master/spa/plugins/… > > see the attached log below, recording starts at 07942.389568, immidatly > avail stops updateing for the playback device and diff changes causing > pw to eventually drop the device. Then the next step would be to check the raw register value (the result of get_ca()) at each atc_pcm_playback_position() call. > interestingly i discovered that also haveing the alsa_time alsa-lib > testing application running is sufficent to avoid this problem, > everything works normally, wierd Or just a matter of a stream assignment? When only one playback or one capture is running, it's the first one. That is, when you run two playbacks, do they both behave correctly? BTW, please keep alsa-devel ML to Cc in your reply. Such information must be shared. thanks, Takashi > With regards > Carl > > > On Wed, 2023-07-19 at 10:43 +0200, Takashi Iwai wrote: > > [Please keep ML in Cc.] > > > > On Wed, 19 Jul 2023 10:10:47 +0200, > > Carl Klemm wrote: > > > > > > Hi, > > > > > > The values only seam nonsensical if playback and record is active > > > at > > > the same time, otherwise eatch on its own is fine > > > > That's an interesting point. You mean "values" as the PCM positions? > > That is, when running in full-duplex mode, only the capture stream > > shows strange PCM positions, while PCM playback shows correctly? > > Or are both screwed up? > > > > And, how strange they are? They jump up/down, random values, or any > > patterns? > > > > > > thanks, > > > > Takashi > > > > > > > > Carl Klemm > > > > > > On Wed, 2023-07-19 at 09:27 +0200, Takashi Iwai wrote: > > > > On Wed, 19 Jul 2023 09:10:12 +0200, > > > > Carl Klemm wrote: > > > > > > > > > > Hi, > > > > > > > > > > see > > > > > https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3352 > > > > > htimestamp returns nonesensical values when recording is active > > > > > on > > > > > a > > > > > EMU20k device > > > > > > > > The driver doesn't provide any own implementation of PCM tstamp, > > > > hence > > > > it must be the standard values. That said, the incorrect values > > > > are > > > > rather the "avail", that is the PCM position, not the timestamp > > > > itself, I suppose. > > > > > > > > Is it only about recording, i.e. playback doesn't show any wrong > > > > values? > > > > > > > > In anyway, try to enable the tracing for snd_pcm, get the > > > > position > > > > and > > > > the system timestamp during recording, and verify whether the > > > > reported > > > > values are reasonable or not. > > > > > > > > > > > > Takashi > > > > > [E][07941.175466] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941210935880 diff: 18446744073709551601 then: 7941211290770 > [E][07941.181068] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7941216741198 diff: 18446744073709551610 then: 7941216894261 > [E][07941.186902] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7944 current_time: 7941222546507 diff: 18446744073709551608 then: 7941222728163 > [E][07941.192886] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941228351809 diff: 18446744073709551601 then: 7941228710780 > [E][07941.198397] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7941234157102 diff: 18446744073709551614 then: 7941234223624 > [E][07941.204180] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941239962387 diff: 18446744073709551615 then: 7941240006443 > [E][07941.210000] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941245767663 diff: 18446744073709551614 then: 7941245825549 > [E][07941.216093] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941251572932 diff: 18446744073709551601 then: 7941251918146 > [E][07941.221913] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941257378195 diff: 18446744073709551601 then: 7941257737228 > [E][07941.227410] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941263183452 diff: 18446744073709551614 then: 7941263235480 > [E][07941.233326] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7941268988704 diff: 18446744073709551609 then: 7941269152326 > [E][07941.239330] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941274793952 diff: 18446744073709551601 then: 7941275154059 > [E][07941.245125] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941280599197 diff: 18446744073709551601 then: 7941280949453 > [E][07941.250629] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941286404439 diff: 18446744073709551614 then: 7941286455133 > [E][07941.256777] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7953 current_time: 7941292209678 diff: 18446744073709551599 then: 7941292601285 > [E][07941.262236] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941298014915 diff: 18446744073709551614 then: 7941298061737 > [E][07941.268330] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7950 current_time: 7941303820150 diff: 18446744073709551602 then: 7941304154202 > [E][07941.273839] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7937 current_time: 7941309625384 diff: 18446744073709551615 then: 7941309665174 > [E][07941.279963] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941315430617 diff: 18446744073709551601 then: 7941315787255 > [E][07941.285746] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7950 current_time: 7941321235848 diff: 18446744073709551602 then: 7941321569953 > [E][07941.291391] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7941327041077 diff: 18446744073709551609 then: 7941327216583 > [E][07941.297389] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941332846304 diff: 18446744073709551600 then: 7941333213145 > [E][07941.302900] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7941338651530 diff: 18446744073709551613 then: 7941338723348 > [E][07941.308700] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7941344456756 diff: 18446744073709551614 then: 7941344523963 > [E][07941.314754] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7950 current_time: 7941350261980 diff: 18446744073709551603 then: 7941350578904 > [E][07941.320740] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7957 current_time: 7941356067201 diff: 18446744073709551595 then: 7941356563621 > [E][07941.326445] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7953 current_time: 7941361872420 diff: 18446744073709551599 then: 7941362269448 > [E][07941.332159] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7949 current_time: 7941367677637 diff: 18446744073709551603 then: 7941367984347 > [E][07941.337710] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941373482853 diff: 18446744073709551614 then: 7941373535566 > [E][07941.344076] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7962 current_time: 7941379288068 diff: 18446744073709551590 then: 7941379900297 > [E][07941.349625] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941385093282 diff: 18446744073709551601 then: 7941385449141 > [E][07941.355731] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941390898496 diff: 18446744073709551587 then: 7941391556174 > [E][07941.360932] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941396703728 diff: 18446744073709551614 then: 7941396757427 > [E][07941.367370] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7966 current_time: 7941402508975 diff: 18446744073709551586 then: 7941403195706 > [E][07941.372849] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941408314234 diff: 18446744073709551601 then: 7941408672490 > [E][07941.378678] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7953 current_time: 7941414119503 diff: 18446744073709551600 then: 7941414501317 > [E][07941.384162] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7941419924779 diff: 18446744073709551614 then: 7941419986057 > [E][07941.390471] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7961 current_time: 7941425730059 diff: 18446744073709551592 then: 7941426294532 > [E][07941.396045] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7950 current_time: 7941431535340 diff: 18446744073709551602 then: 7941431869128 > [E][07941.401553] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7937 current_time: 7941437340623 diff: 18446744073709551615 then: 7941437379595 > [E][07941.407959] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7964 current_time: 7941443145906 diff: 18446744073709551588 then: 7941443784286 > [E][07941.413288] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7941448951190 diff: 18446744073709551609 then: 7941449114419 > [E][07941.419291] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941454756473 diff: 18446744073709551601 then: 7941455114484 > [E][07941.425089] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941460561755 diff: 18446744073709551601 then: 7941460912171 > [E][07941.430733] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7944 current_time: 7941466367035 diff: 18446744073709551608 then: 7941466556234 > [E][07941.436707] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941472172313 diff: 18446744073709551601 then: 7941472530079 > [E][07941.442359] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7945 current_time: 7941477977588 diff: 18446744073709551607 then: 7941478184186 > [E][07941.448774] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7972 current_time: 7941483782860 diff: 18446744073709551581 then: 7941484598957 > [E][07941.454067] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7949 current_time: 7941489588128 diff: 18446744073709551603 then: 7941489890618 > [E][07941.459981] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7954 current_time: 7941495393393 diff: 18446744073709551598 then: 7941495804032 > [E][07941.465514] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7942 current_time: 7941501198655 diff: 18446744073709551610 then: 7941501338799 > [E][07941.471543] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941507003914 diff: 18446744073709551601 then: 7941507365781 > [E][07941.477389] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7954 current_time: 7941512809169 diff: 18446744073709551599 then: 7941513211755 > [E][07941.483126] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941518614420 diff: 18446744073709551602 then: 7941518948997 > [E][07941.488955] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941524419666 diff: 18446744073709551601 then: 7941524778411 > [E][07941.494443] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941530224908 diff: 18446744073709551615 then: 7941530268479 > [E][07941.500748] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7959 current_time: 7941536030145 diff: 18446744073709551593 then: 7941536571231 > [E][07941.506439] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7955 current_time: 7941541835378 diff: 18446744073709551598 then: 7941542262262 > [E][07941.512176] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941547640606 diff: 18446744073709551601 then: 7941547999216 > [E][07941.517884] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7947 current_time: 7941553445831 diff: 18446744073709551605 then: 7941553708319 > [E][07941.524067] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7964 current_time: 7941559251053 diff: 18446744073709551588 then: 7941559890112 > [E][07941.529490] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7947 current_time: 7941565056272 diff: 18446744073709551605 then: 7941565314660 > [E][07941.535422] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7953 current_time: 7941570861489 diff: 18446744073709551600 then: 7941571244838 > [E][07941.540915] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7941576666704 diff: 18446744073709551613 then: 7941576738578 > [E][07941.547071] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7954 current_time: 7941582471917 diff: 18446744073709551598 then: 7941582893934 > [E][07941.552602] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7942 current_time: 7941588277128 diff: 18446744073709551610 then: 7941588426386 > [E][07941.558616] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941594082338 diff: 18446744073709551601 then: 7941594439327 > [E][07941.564233] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7941599887548 diff: 18446744073709551609 then: 7941600057002 > [E][07941.570379] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7958 current_time: 7941605692757 diff: 18446744073709551594 then: 7941606202302 > [E][07941.576034] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941611497966 diff: 18446744073709551601 then: 7941611856973 > [E][07941.581841] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941617303175 diff: 18446744073709551601 then: 7941617663756 > [E][07941.587575] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7948 current_time: 7941623108384 diff: 18446744073709551604 then: 7941623397723 > [E][07941.593446] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941628913592 diff: 18446744073709551601 then: 7941629268165 > [E][07941.599258] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941634718800 diff: 18446744073709551601 then: 7941635081079 > [E][07941.605137] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7955 current_time: 7941640524007 diff: 18446744073709551597 then: 7941640959020 > [E][07941.610999] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7958 current_time: 7941646329213 diff: 18446744073709551595 then: 7941646821854 > [E][07941.616479] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7941652134417 diff: 18446744073709551609 then: 7941652303030 > [E][07941.622474] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941657939620 diff: 18446744073709551601 then: 7941658296736 > [E][07941.628101] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7944 current_time: 7941663744822 diff: 18446744073709551609 then: 7941663924731 > [E][07941.634137] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7954 current_time: 7941669550023 diff: 18446744073709551598 then: 7941669960304 > [E][07941.639590] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7941675355223 diff: 18446744073709551614 then: 7941675413688 > [E][07941.645711] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941681160421 diff: 18446744073709551600 then: 7941681533333 > [E][07941.651310] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7941686965617 diff: 18446744073709551609 then: 7941687134136 > [E][07941.657041] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941692770813 diff: 18446744073709551614 then: 7941692827758 > [E][07941.663178] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7955 current_time: 7941698576008 diff: 18446744073709551598 then: 7941699001535 > [E][07941.668893] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7950 current_time: 7941704381202 diff: 18446744073709551602 then: 7941704715221 > [E][07941.674446] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7940 current_time: 7941710186395 diff: 18446744073709551613 then: 7941710269789 > [E][07941.680593] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7955 current_time: 7941715991587 diff: 18446744073709551598 then: 7941716415042 > [E][07941.686388] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7954 current_time: 7941721796777 diff: 18446744073709551598 then: 7941722210004 > [E][07941.692138] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941727601965 diff: 18446744073709551601 then: 7941727959870 > [E][07941.697635] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941733407151 diff: 18446744073709551614 then: 7941733458411 > [E][07941.704051] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7965 current_time: 7941739212336 diff: 18446744073709551587 then: 7941739873218 > [E][07941.709378] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7944 current_time: 7941745017520 diff: 18446744073709551608 then: 7941745201106 > [E][07941.715386] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941750822704 diff: 18446744073709551600 then: 7941751207207 > [E][07941.721011] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7945 current_time: 7941756627888 diff: 18446744073709551607 then: 7941756834795 > [E][07941.727166] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7960 current_time: 7941762433072 diff: 18446744073709551592 then: 7941762987511 > [E][07941.732826] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7954 current_time: 7941768238257 diff: 18446744073709551598 then: 7941768647630 > [E][07941.738924] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7967 current_time: 7941774043443 diff: 18446744073709551585 then: 7941774747571 > [E][07941.744093] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941779848630 diff: 18446744073709551614 then: 7941779916617 > [E][07941.749981] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7942 current_time: 7941785653819 diff: 18446744073709551610 then: 7941785803979 > [E][07941.755971] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941791459009 diff: 18446744073709551602 then: 7941791793040 > [E][07941.761480] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941797264199 diff: 18446744073709551615 then: 7941797304275 > [E][07941.767323] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7941803069389 diff: 18446744073709551613 then: 7941803144861 > [E][07941.773233] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7944 current_time: 7941808874578 diff: 18446744073709551608 then: 7941809056043 > [E][07941.779192] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941814679768 diff: 18446744073709551602 then: 7941815013533 > [E][07941.785061] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7953 current_time: 7941820484957 diff: 18446744073709551599 then: 7941820883999 > [E][07941.790735] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7948 current_time: 7941826290145 diff: 18446744073709551605 then: 7941826557450 > [E][07941.796628] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941832095332 diff: 18446744073709551601 then: 7941832449467 > [E][07941.802437] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941837900518 diff: 18446744073709551601 then: 7941838258230 > [E][07941.808261] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941843705704 diff: 18446744073709551600 then: 7941844082280 > [E][07941.813795] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7940 current_time: 7941849510890 diff: 18446744073709551612 then: 7941849617096 > [E][07941.819851] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7941855316077 diff: 18446744073709551601 then: 7941855672529 > [E][07941.826059] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7955 current_time: 7941861121263 diff: 18446744073709551583 then: 7941861881393 > [E][07941.831284] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7941866926467 diff: 18446744073709551609 then: 7941867107258 > [E][07941.836962] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941872731686 diff: 18446744073709551614 then: 7941872784957 > [E][07941.843100] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7953 current_time: 7941878536917 diff: 18446744073709551600 then: 7941878921641 > [E][07941.848876] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941884342157 diff: 18446744073709551601 then: 7941884697692 > [E][07941.854375] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941890147405 diff: 18446744073709551614 then: 7941890197324 > [E][07941.860755] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7963 current_time: 7941895952659 diff: 18446744073709551589 then: 7941896576563 > [E][07941.866293] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941901757918 diff: 18446744073709551601 then: 7941902114583 > [E][07941.872100] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941907563181 diff: 18446744073709551601 then: 7941907920957 > [E][07941.877615] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7941913368447 diff: 18446744073709551613 then: 7941913437713 > [E][07941.884059] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7967 current_time: 7941919173715 diff: 18446744073709551585 then: 7941919880467 > [E][07941.889929] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7953 current_time: 7941924978985 diff: 18446744073709551582 then: 7941925751665 > [E][07941.895321] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941930784278 diff: 18446744073709551601 then: 7941931141570 > [E][07941.900831] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7941936589589 diff: 18446744073709551614 then: 7941936652062 > [E][07941.907388] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7962 current_time: 7941942394915 diff: 18446744073709551581 then: 7941943210508 > [E][07941.912713] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7950 current_time: 7941948200264 diff: 18446744073709551602 then: 7941948534041 > [E][07941.918438] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7947 current_time: 7941954005631 diff: 18446744073709551605 then: 7941954261084 > [E][07941.924058] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7941959811012 diff: 18446744073709551614 then: 7941959878628 > [E][07941.930154] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7941965616403 diff: 18446744073709551601 then: 7941965974824 > [E][07941.935935] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7950 current_time: 7941971421800 diff: 18446744073709551602 then: 7941971755591 > [E][07941.941849] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7955 current_time: 7941977227202 diff: 18446744073709551597 then: 7941977669892 > [E][07941.947276] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7941983032606 diff: 18446744073709551614 then: 7941983096492 > [E][07941.953351] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7950 current_time: 7941988838010 diff: 18446744073709551602 then: 7941989172025 > [E][07941.958989] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7941994643412 diff: 18446744073709551609 then: 7941994811156 > [E][07941.964806] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7944 current_time: 7942000448811 diff: 18446744073709551609 then: 7942000627887 > [E][07941.970476] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7942006254206 diff: 18446744073709551615 then: 7942006297822 > [E][07941.976412] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7942012059594 diff: 18446744073709551609 then: 7942012234515 > [E][07941.982090] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7942017864976 diff: 18446744073709551614 then: 7942017912070 > [E][07941.988026] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7944 current_time: 7942023670352 diff: 18446744073709551609 then: 7942023848031 > [E][07941.994067] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7954 current_time: 7942029475720 diff: 18446744073709551598 then: 7942029888272 > [E][07941.999820] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942035281081 diff: 18446744073709551601 then: 7942035640335 > [E][07942.005624] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942041086434 diff: 18446744073709551601 then: 7942041444237 > [E][07942.011114] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7942046891778 diff: 18446744073709551615 then: 7942046936945 > [E][07942.017463] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7962 current_time: 7942052697112 diff: 18446744073709551591 then: 7942053283676 > [E][07942.022832] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7942 current_time: 7942058502437 diff: 18446744073709551610 then: 7942058654921 > [E][07942.028545] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7942064307753 diff: 18446744073709551614 then: 7942064366784 > [E][07942.034656] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942070113060 diff: 18446744073709551600 then: 7942070476097 > [E][07942.040232] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7942 current_time: 7942075918359 diff: 18446744073709551611 then: 7942076054028 > [E][07942.046144] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7947 current_time: 7942081723649 diff: 18446744073709551606 then: 7942081966481 > [E][07942.052470] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7969 current_time: 7942087528931 diff: 18446744073709551583 then: 7942088291577 > [E][07942.057559] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7942093334205 diff: 18446744073709551614 then: 7942093381015 > [E][07942.063466] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7942 current_time: 7942099139473 diff: 18446744073709551610 then: 7942099288116 > [E][07942.069482] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942104944735 diff: 18446744073709551601 then: 7942105302029 > [E][07942.075284] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7942110749991 diff: 18446744073709551601 then: 7942111104396 > [E][07942.080798] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7942116555242 diff: 18446744073709551614 then: 7942116617840 > [E][07942.086694] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7942122360489 diff: 18446744073709551610 then: 7942122517046 > [E][07942.092413] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7942128165731 diff: 18446744073709551614 then: 7942128233228 > [E][07942.098509] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942133970968 diff: 18446744073709551601 then: 7942134329269 > [E][07942.103991] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7937 current_time: 7942139776199 diff: 18446744073709551615 then: 7942139814057 > [E][07942.110108] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7942145581425 diff: 18446744073709551601 then: 7942145928146 > [E][07942.115901] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7942151386648 diff: 18446744073709551602 then: 7942151720732 > [E][07942.121830] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7956 current_time: 7942157191867 diff: 18446744073709551596 then: 7942157650214 > [E][07942.127292] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7941 current_time: 7942162997082 diff: 18446744073709551611 then: 7942163111709 > [E][07942.133355] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942168802295 diff: 18446744073709551600 then: 7942169174366 > [E][07942.139299] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7958 current_time: 7942174607506 diff: 18446744073709551594 then: 7942175118308 > [E][07942.144942] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7942180412715 diff: 18446744073709551601 then: 7942180761795 > [E][07942.150894] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7958 current_time: 7942186217924 diff: 18446744073709551595 then: 7942186707777 > [E][07942.156257] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7942192023131 diff: 18446744073709551614 then: 7942192078529 > [E][07942.162371] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942197828337 diff: 18446744073709551601 then: 7942198190746 > [E][07942.168176] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942203633541 diff: 18446744073709551601 then: 7942203995896 > [E][07942.174050] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7955 current_time: 7942209438742 diff: 18446744073709551597 then: 7942209869601 > [E][07942.179781] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7942215243941 diff: 18446744073709551601 then: 7942215600628 > [E][07942.185583] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7942221049139 diff: 18446744073709551601 then: 7942221402455 > [E][07942.191370] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7942226854336 diff: 18446744073709551602 then: 7942227191285 > [E][07942.197436] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7962 current_time: 7942232659532 diff: 18446744073709551590 then: 7942233257158 > [E][07942.202979] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7950 current_time: 7942238464727 diff: 18446744073709551602 then: 7942238798526 > [E][07942.208807] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942244269921 diff: 18446744073709551601 then: 7942244626091 > [E][07942.214302] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7942250075114 diff: 18446744073709551614 then: 7942250122868 > [E][07942.220511] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7956 current_time: 7942255880306 diff: 18446744073709551597 then: 7942256330904 > [E][07942.226136] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7948 current_time: 7942261685497 diff: 18446744073709551605 then: 7942261956943 > [E][07942.231833] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7942267490686 diff: 18446744073709551609 then: 7942267654646 > [E][07942.237530] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7942273295873 diff: 18446744073709551614 then: 7942273350836 > [E][07942.243834] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7960 current_time: 7942279101060 diff: 18446744073709551592 then: 7942279653624 > [E][07942.249456] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942284906246 diff: 18446744073709551600 then: 7942285276891 > [E][07942.255249] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942290711433 diff: 18446744073709551601 then: 7942291068470 > [E][07942.260760] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7939 current_time: 7942296516619 diff: 18446744073709551614 then: 7942296580042 > [E][07942.266927] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7955 current_time: 7942302321804 diff: 18446744073709551598 then: 7942302745838 > [E][07942.272660] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7942308126986 diff: 18446744073709551601 then: 7942308478941 > [E][07942.278286] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7944 current_time: 7942313932167 diff: 18446744073709551609 then: 7942314107175 > [E][07942.284133] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7945 current_time: 7942319737346 diff: 18446744073709551607 then: 7942319952189 > [E][07942.289880] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7942325542524 diff: 18446744073709551610 then: 7942325700844 > [E][07942.295819] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7949 current_time: 7942331347700 diff: 18446744073709551604 then: 7942331640346 > [E][07942.301504] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7942337152874 diff: 18446744073709551609 then: 7942337324153 > [E][07942.307364] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7946 current_time: 7942342958047 diff: 18446744073709551607 then: 7942343182835 > [E][07942.313342] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7953 current_time: 7942348763218 diff: 18446744073709551599 then: 7942349161408 > [E][07942.319084] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7951 current_time: 7942354568388 diff: 18446744073709551602 then: 7942354902635 > [E][07942.324773] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7945 current_time: 7942360373556 diff: 18446744073709551607 then: 7942360593210 > [E][07942.330728] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942366178723 diff: 18446744073709551600 then: 7942366546680 > [E][07942.336605] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7955 current_time: 7942371983890 diff: 18446744073709551597 then: 7942372423985 > [E][07942.342697] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7968 current_time: 7942377789057 diff: 18446744073709551584 then: 7942378517014 > [E][07942.347975] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7944 current_time: 7942383594225 diff: 18446744073709551608 then: 7942383795091 > [E][07942.353669] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7940 current_time: 7942389399394 diff: 18446744073709551613 then: 7942389488870 > [E][07942.359754] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7952 current_time: 7942395204563 diff: 18446744073709551600 then: 7942395574194 > [E][07942.365824] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7964 current_time: 7942401009733 diff: 18446744073709551589 then: 7942401643079 > [E][07942.371033] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7938 current_time: 7942406814903 diff: 18446744073709551615 then: 7942406853189 > [E][07942.376827] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7937 current_time: 7942412620072 diff: 18446744073709551615 then: 7942412647719 > [E][07942.382766] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7943 current_time: 7942418425241 diff: 18446744073709551609 then: 7942418585673 > [E][07942.388570] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7729 current_time: 7942424230410 diff: 18446744073709551609 then: 7942424390739 > [E][07942.389568] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 248 current_time: 7942425356776 diff: 18446744073709551615 then: 7942425389902 > [E][07942.389732] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 256 current_time: 7942425538181 diff: 0 then: 7942425557061 > [E][07942.394256] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7473 current_time: 7942430035851 diff: 248 then: 7942424390739 > [E][07942.395559] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 256 current_time: 7942431343210 diff: 18446744073709551615 then: 7942431380631 > [E][07942.400108] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942435841788 diff: 504 then: 7942424390739 > [E][07942.401369] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 256 current_time: 7942437148314 diff: 18446744073709551615 then: 7942437189597 > [E][07942.407178] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 256 current_time: 7942442953521 diff: 18446744073709551615 then: 7942442998527 > [E][07942.410564] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942446340654 diff: 967 then: 7942424390739 > [E][07942.412981] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942448758855 diff: 18446744073709551615 then: 7942448802286 > [E][07942.418906] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942454564323 diff: 18446744073709551609 then: 7942454726680 > [E][07942.422720] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942457950631 diff: 1479 then: 7942424390739 > [E][07942.424621] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942460369938 diff: 18446744073709551613 then: 7942460441794 > [E][07942.430390] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 256 current_time: 7942466175705 diff: 18446744073709551615 then: 7942466210537 > [E][07942.433779] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942469560608 diff: 1991 then: 7942424390739 > [E][07942.436201] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 256 current_time: 7942471981635 diff: 18446744073709551615 then: 7942472021711 > [E][07942.442140] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942477787736 diff: 18446744073709551609 then: 7942477960529 > [E][07942.445389] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942481170585 diff: 2503 then: 7942424390739 > [E][07942.447997] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942483594009 diff: 18446744073709551607 then: 7942483818743 > [E][07942.453622] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 256 current_time: 7942489400459 diff: 18446744073709551615 then: 7942489441806 > [E][07942.456990] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942492780562 diff: 3015 then: 7942424390739 > [E][07942.459417] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942495207086 diff: 18446744073709551615 then: 7942495239096 > [E][07942.465759] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942501013881 diff: 18446744073709551592 then: 7942501576696 > [E][07942.468724] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942504390539 diff: 3527 then: 7942424390739 > [E][07942.471162] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942506820843 diff: 18446744073709551609 then: 7942506982920 > [E][07942.476848] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942512627968 diff: 18446744073709551615 then: 7942512667567 > [E][07942.480224] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942516000516 diff: 4039 then: 7942424390739 > [E][07942.482662] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942518435246 diff: 18446744073709551614 then: 7942518482257 > [E][07942.488729] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942524242667 diff: 18446744073709551603 then: 7942524549570 > [E][07942.491987] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942527610493 diff: 4551 then: 7942424390739 > [E][07942.494267] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942530050226 diff: 18446744073709551615 then: 7942530088406 > [E][07942.500230] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942535857913 diff: 18446744073709551608 then: 7942536049507 > [E][07942.503748] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942539220470 diff: 5063 then: 7942424390739 > [E][07942.506243] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942541665727 diff: 18446744073709551599 then: 7942542061333 > [E][07942.512189] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942547473659 diff: 18446744073709551593 then: 7942548007470 > [E][07942.515081] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942550830447 diff: 5575 then: 7942424390739 > [E][07942.517517] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942553281706 diff: 18446744073709551614 then: 7942553335022 > [E][07942.523610] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942559089860 diff: 18446744073709551602 then: 7942559429695 > [E][07942.527048] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942562440424 diff: 6087 then: 7942424390739 > [E][07942.529483] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942564898119 diff: 18446744073709551599 then: 7942565301302 > [E][07942.535043] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942570706473 diff: 18446744073709551610 then: 7942570862457 > [E][07942.538393] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942574050401 diff: 6599 then: 7942424390739 > [E][07942.541131] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942576514920 diff: 18446744073709551597 then: 7942576951394 > [E][07942.546935] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942582323454 diff: 18446744073709551598 then: 7942582753424 > [E][07942.550275] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942585660378 diff: 7111 then: 7942424390739 > [E][07942.552452] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942588132068 diff: 18446744073709551610 then: 7942588274224 > [E][07942.558164] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942593940761 diff: 18446744073709551615 then: 7942593982799 > [E][07942.561601] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942597270355 diff: 7623 then: 7942424390739 > [E][07942.564163] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942599749525 diff: 18446744073709551606 then: 7942599981472 > [E][07942.570113] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942605558355 diff: 18446744073709551600 then: 7942605931029 > [E][07942.573099] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942608880332 diff: 8135 then: 7942424390739 > [E][07942.575883] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942611367242 diff: 18446744073709551602 then: 7942611700828 > [E][07942.581533] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942617176188 diff: 18446744073709551609 then: 7942617351803 > [E][07942.584781] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942620490309 diff: 8647 then: 7942424390739 > [E][07942.587376] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942622985184 diff: 18446744073709551607 then: 7942623193001 > [E][07942.593342] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942628794225 diff: 18446744073709551600 then: 7942629159838 > [E][07942.596328] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942632100286 diff: 9159 then: 7942424390739 > [E][07942.599119] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942634603305 diff: 18446744073709551602 then: 7942634936777 > [E][07942.604762] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942640412417 diff: 18446744073709551609 then: 7942640581620 > [E][07942.608072] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942643710263 diff: 9671 then: 7942424390739 > [E][07942.610438] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942646221557 diff: 18446744073709551615 then: 7942646257759 > [E][07942.616695] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942652030722 diff: 18446744073709551595 then: 7942652513387 > [E][07942.619858] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942655320240 diff: 10183 then: 7942424390739 > [E][07942.622380] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942657839916 diff: 18446744073709551601 then: 7942658197517 > [E][07942.628117] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942663649134 diff: 18446744073709551604 then: 7942663935816 > [E][07942.631286] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942666930217 diff: 10695 then: 7942424390739 > [E][07942.633714] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942669458379 diff: 18446744073709551613 then: 7942669533296 > [E][07942.639813] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942675267651 diff: 18446744073709551600 then: 7942675631568 > [E][07942.642779] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942678540194 diff: 11207 then: 7942424390739 > [E][07942.645617] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942681076946 diff: 18446744073709551601 then: 7942681434631 > [E][07942.651109] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942686886261 diff: 18446744073709551615 then: 7942686927638 > [E][07942.654546] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942690150171 diff: 11719 then: 7942424390739 > [E][07942.657395] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7942692695594 diff: 18446744073709551594 then: 7942693212750 > [E][07942.662745] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942698504940 diff: 18446744073709551614 then: 7942698563067 > [E][07942.666300] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942701760148 diff: 12231 then: 7942424390739 > [E][07942.668667] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942704314301 diff: 18446744073709551609 then: 7942704485264 > [E][07942.674358] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942710123673 diff: 18446744073709551614 then: 7942710176319 > [E][07942.677598] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942713370125 diff: 12743 then: 7942424390739 > [E][07942.680228] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942715933059 diff: 18446744073709551611 then: 7942716047265 > [E][07942.686086] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942721742452 diff: 18446744073709551609 then: 7942721906967 > [E][07942.689519] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942724980102 diff: 13255 then: 7942424390739 > [E][07942.691776] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942727551852 diff: 18446744073709551615 then: 7942727594698 > [E][07942.697743] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942733361259 diff: 18446744073709551608 then: 7942733564463 > [E][07942.700801] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942736590079 diff: 13767 then: 7942424390739 > [E][07942.703848] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7942739170673 diff: 18446744073709551595 then: 7942739666180 > [E][07942.709520] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942744980090 diff: 18446744073709551601 then: 7942745336691 > [E][07942.712425] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942748200056 diff: 14279 then: 7942424390739 > [E][07942.715118] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942750789509 diff: 18446744073709551610 then: 7942750939882 > [E][07942.720844] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942756598929 diff: 18446744073709551614 then: 7942756660733 > [E][07942.724022] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942759810033 diff: 14791 then: 7942424390739 > [E][07942.727158] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7942762408353 diff: 18446744073709551592 then: 7942762974921 > [E][07942.732442] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942768217782 diff: 18446744073709551615 then: 7942768259994 > [E][07942.735761] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942771420010 diff: 15303 then: 7942424390739 > [E][07942.738237] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942774027217 diff: 18446744073709551615 then: 7942774058208 > [E][07942.744268] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942779836661 diff: 18446744073709551605 then: 7942780087049 > [E][07942.747250] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942783029987 diff: 15815 then: 7942424390739 > [E][07942.750274] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7942785646106 diff: 18446744073709551597 then: 7942786091015 > [E][07942.755666] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942791455548 diff: 18446744073709551615 then: 7942791485875 > [E][07942.759146] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942794639964 diff: 16327 then: 7942424390739 > [E][07942.761478] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942797264990 diff: 18446744073709551615 then: 7942797298729 > [E][07942.767293] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942803074425 diff: 18446744073709551615 then: 7942803111955 > [E][07942.770478] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942806249941 diff: 16839 then: 7942424390739 > [E][07942.773091] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 264 current_time: 7942808883847 diff: 18446744073709551615 then: 7942808912461 > [E][07942.779226] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942814693261 diff: 18446744073709551601 then: 7942815044910 > [E][07942.782222] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942817859918 diff: 17351 then: 7942424390739 > [E][07942.784976] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942820502670 diff: 18446744073709551604 then: 7942820795341 > [E][07942.790547] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942826312072 diff: 18446744073709551614 then: 7942826364500 > [E][07942.793812] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942829469895 diff: 17863 then: 7942424390739 > [E][07942.796572] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942832121462 diff: 18446744073709551605 then: 7942832390569 > [E][07942.802153] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942837930839 diff: 18446744073709551615 then: 7942837971873 > [E][07942.805413] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942841079872 diff: 18375 then: 7942424390739 > [E][07942.807952] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942843740199 diff: 18446744073709551615 then: 7942843771503 > [E][07942.813872] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942849549539 diff: 18446744073709551610 then: 7942849690665 > [E][07942.816914] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942852689849 diff: 18887 then: 7942424390739 > [E][07942.819669] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942855358863 diff: 18446744073709551611 then: 7942855490103 > [E][07942.825392] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942861168174 diff: 18446744073709551615 then: 7942861209754 > [E][07942.828709] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942864299826 diff: 19399 then: 7942424390739 > [E][07942.831204] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942866977468 diff: 18446744073709551615 then: 7942867022764 > [E][07942.837002] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942872786743 diff: 18446744073709551615 then: 7942872820786 > [E][07942.840323] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942875909803 diff: 19911 then: 7942424390739 > [E][07942.842946] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942878595998 diff: 18446744073709551609 then: 7942878763660 > [E][07942.848822] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942884405228 diff: 18446744073709551606 then: 7942884639898 > [E][07942.851852] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942887519780 diff: 20423 then: 7942424390739 > [E][07942.854551] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942890214435 diff: 18446744073709551610 then: 7942890369077 > [E][07942.860448] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942896023615 diff: 18446744073709551606 then: 7942896266387 > [E][07942.863794] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942899129757 diff: 20935 then: 7942424390739 > [E][07942.866172] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942901832770 diff: 18446744073709551609 then: 7942901993909 > [E][07942.871867] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942907641897 diff: 18446744073709551615 then: 7942907685468 > [E][07942.875114] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942910739734 diff: 21447 then: 7942424390739 > [E][07942.877777] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942913450999 diff: 18446744073709551610 then: 7942913598882 > [E][07942.884102] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 296 current_time: 7942919260072 diff: 18446744073709551587 then: 7942919918482 > [E][07942.886685] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942922349711 diff: 21959 then: 7942424390739 > [E][07942.889642] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7942925069124 diff: 18446744073709551599 then: 7942925457845 > [E][07942.895272] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942930878156 diff: 18446744073709551607 then: 7942931090532 > [E][07942.898170] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942933959688 diff: 22471 then: 7942424390739 > [E][07942.901062] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942936687169 diff: 18446744073709551608 then: 7942936883190 > [E][07942.906749] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942942496163 diff: 18446744073709551615 then: 7942942538653 > [E][07942.910131] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942945569665 diff: 22983 then: 7942424390739 > [E][07942.912823] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7942948305140 diff: 18446744073709551602 then: 7942948639386 > [E][07942.918632] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7942954114096 diff: 18446744073709551602 then: 7942954448124 > [E][07942.921529] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942957179642 diff: 23495 then: 7942424390739 > [E][07942.924139] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942959923030 diff: 18446744073709551615 then: 7942959959516 > [E][07942.930503] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 296 current_time: 7942965731945 diff: 18446744073709551591 then: 7942966318764 > [E][07942.933322] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942968789619 diff: 24007 then: 7942424390739 > [E][07942.935985] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942971540843 diff: 18446744073709551605 then: 7942971802099 > [E][07942.941698] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7942977349728 diff: 18446744073709551609 then: 7942977516026 > [E][07942.944932] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942980399596 diff: 24519 then: 7942424390739 > [E][07942.947384] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7942983158600 diff: 18446744073709551615 then: 7942983201861 > [E][07942.953577] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7942988967460 diff: 18446744073709551598 then: 7942989392558 > [E][07942.956548] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7942992009573 diff: 25031 then: 7942424390739 > [E][07942.959315] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7942994776312 diff: 18446744073709551601 then: 7942995130448 > [E][07942.965158] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943000585154 diff: 18446744073709551599 then: 7943000973446 > [E][07942.968142] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943003619550 diff: 25543 then: 7942424390739 > [E][07942.970756] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943006393988 diff: 18446744073709551609 then: 7943006571441 > [E][07942.976567] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943012202812 diff: 18446744073709551608 then: 7943012384307 > [E][07942.979562] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943015229527 diff: 26055 then: 7942424390739 > [E][07942.982258] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943018011628 diff: 18446744073709551614 then: 7943018075662 > [E][07942.988621] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 304 current_time: 7943023820428 diff: 18446744073709551589 then: 7943024437694 > [E][07942.991171] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943026839504 diff: 26567 then: 7942424390739 > [E][07942.993852] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7943029629208 diff: 18446744073709551615 then: 7943029669235 > [E][07942.999666] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7943035437972 diff: 18446744073709551614 then: 7943035484273 > [E][07943.002976] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943038449481 diff: 27079 then: 7942424390739 > [E][07943.005833] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943041246725 diff: 18446744073709551599 then: 7943041649926 > [E][07943.011275] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 272 current_time: 7943047055470 diff: 18446744073709551615 then: 7943047092246 > [E][07943.014384] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943050059458 diff: 27591 then: 7942424390739 > [E][07943.017399] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943052864210 diff: 18446744073709551601 then: 7943053214302 > [E][07943.023263] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 296 current_time: 7943058672944 diff: 18446744073709551599 then: 7943059078120 > [E][07943.026234] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943061669435 diff: 28103 then: 7942424390739 > [E][07943.029000] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943064481664 diff: 18446744073709551602 then: 7943064814991 > [E][07943.034710] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943070290371 diff: 18446744073709551606 then: 7943070527286 > [E][07943.037514] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943073279412 diff: 28615 then: 7942424390739 > [E][07943.040479] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943076099060 diff: 18446744073709551608 then: 7943076293892 > [E][07943.046447] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943081907734 diff: 18446744073709551601 then: 7943082262049 > [E][07943.049131] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943084889389 diff: 29127 then: 7942424390739 > [E][07943.051936] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943087716396 diff: 18446744073709551615 then: 7943087753449 > [E][07943.057761] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943093525040 diff: 18446744073709551614 then: 7943093578004 > [E][07943.060782] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943096499366 diff: 29639 then: 7942424390739 > [E][07943.064049] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 296 current_time: 7943099333662 diff: 18446744073709551593 then: 7943099857403 > [E][07943.069661] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943105142267 diff: 18446744073709551602 then: 7943105475978 > [E][07943.072351] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943108109343 diff: 30151 then: 7942424390739 > [E][07943.075277] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943110950856 diff: 18446744073709551610 then: 7943111098321 > [E][07943.080982] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943116759431 diff: 18446744073709551615 then: 7943116798316 > [E][07943.084071] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943119719320 diff: 30663 then: 7942424390739 > [E][07943.087090] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943122567988 diff: 18446744073709551601 then: 7943122909309 > [E][07943.092895] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943128376531 diff: 18446744073709551602 then: 7943128710115 > [E][07943.095848] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943131329297 diff: 31175 then: 7942424390739 > [E][07943.098504] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943134185061 diff: 18446744073709551610 then: 7943134325006 > [E][07943.104223] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943139993581 diff: 18446744073709551614 then: 7943140039389 > [E][07943.107175] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943142939274 diff: 31687 then: 7942424390739 > [E][07943.110223] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943145802088 diff: 18446744073709551606 then: 7943146039347 > [E][07943.115838] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943151610579 diff: 18446744073709551615 then: 7943151654082 > [E][07943.119069] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943154549251 diff: 32199 then: 7942424390739 > [E][07943.121982] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 296 current_time: 7943157419050 diff: 18446744073709551600 then: 7943157801447 > [E][07943.127530] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943163227498 diff: 18446744073709551611 then: 7943163344782 > [E][07943.130738] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943166159228 diff: 32711 then: 7942424390739 > [E][07943.133978] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 312 current_time: 7943169035927 diff: 18446744073709551583 then: 7943169794965 > [E][07943.139363] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943174844334 diff: 18446744073709551602 then: 7943175178113 > [E][07943.142135] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943177769205 diff: 33223 then: 7942424390739 > [E][07943.145005] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 288 current_time: 7943180652724 diff: 18446744073709551609 then: 7943180821564 > [E][07943.150733] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bab1380 device: front:1 avail: 280 current_time: 7943186461093 diff: 18446744073709551613 then: 7943186547875 > [E][07943.153687] spa.alsa | [ alsa-pcm.c: 2016 get_avail()] handle: 0x56197bcad0d0 device: surround40:1 avail: 7217 current_time: 7943189379182 diff: 33735 then: 7942424390739 > [E][07943.153810] spa.alsa | [ alsa-pcm.c: 2024 get_avail()] surround40:1: wrong htimestamps from driver, disabling

2 2

Broken microphone switch on mtk-rt5650
by GitHub issues - edited 20 Jul '23

20 Jul '23

alsa-project/alsa-ucm-conf issue #336 was edited from jenneron: The problem is that both headphones and mic use `JackControl "Headset Jack"` When there are headphones without mic connected it switches to external mic which is not present. In kernel there appears to be only one switch https://github.com/torvalds/linux/blob/3c8b5861850c734add65233e538d4a8c2dff… responsible for handling both headphone and microphone. So, how is userspace supposed to distinguish connection of headset and headphones? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/336 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Broken microphone switch on mtk-rt5650
by GitHub issues - opened 20 Jul '23

20 Jul '23

alsa-project/alsa-ucm-conf issue #336 was opened from jenneron: The problem is that both headphones and mic use `JackControl "Headset Jack"` When there are headphones without mic connected it switches to external mic which is not present. In kernel there appears to be only one switch https://github.com/torvalds/linux/blob/3c8b5861850c734add65233e538d4a8c2dff… responsible for handling both headphone and microphone. So, how is userspace supposed to distinguish connection of headset and headphones without mic? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/336 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: fsl_spdif: Add support for 22.05 kHz sample rate
by Matus Gajdos 19 Jul '23

19 Jul '23

Add support for 22.05 kHz sample rate for TX. Signed-off-by: Matus Gajdos <matuszpd(a)gmail.com> --- sound/soc/fsl/fsl_spdif.c | 8 ++++++-- sound/soc/fsl/fsl_spdif.h | 6 ++++-- 2 files changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 015c3708aa04..95e639711eba 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -514,6 +514,10 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, int ret; switch (sample_rate) { + case 22050: + rate = SPDIF_TXRATE_22050; + csfs = IEC958_AES3_CON_FS_22050; + break; case 32000: rate = SPDIF_TXRATE_32000; csfs = IEC958_AES3_CON_FS_32000; @@ -1422,7 +1426,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, struct clk *clk, u64 savesub, enum spdif_txrate index, bool round) { - static const u32 rate[] = { 32000, 44100, 48000, 88200, 96000, 176400, + static const u32 rate[] = { 22050, 32000, 44100, 48000, 88200, 96000, 176400, 192000, }; bool is_sysclk = clk_is_match(clk, spdif_priv->sysclk); u64 rate_ideal, rate_actual, sub; @@ -1483,7 +1487,7 @@ static u32 fsl_spdif_txclk_caldiv(struct fsl_spdif_priv *spdif_priv, static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, enum spdif_txrate index) { - static const u32 rate[] = { 32000, 44100, 48000, 88200, 96000, 176400, + static const u32 rate[] = { 22050, 32000, 44100, 48000, 88200, 96000, 176400, 192000, }; struct platform_device *pdev = spdif_priv->pdev; struct device *dev = &pdev->dev; diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index 75b42a692c90..2bc1b10c17d4 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -175,7 +175,8 @@ enum spdif_gainsel { /* SPDIF tx rate */ enum spdif_txrate { - SPDIF_TXRATE_32000 = 0, + SPDIF_TXRATE_22050 = 0, + SPDIF_TXRATE_32000, SPDIF_TXRATE_44100, SPDIF_TXRATE_48000, SPDIF_TXRATE_88200, @@ -191,7 +192,8 @@ enum spdif_txrate { #define SPDIF_QSUB_SIZE (SPDIF_UBITS_SIZE / 8) -#define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_32000 | \ +#define FSL_SPDIF_RATES_PLAYBACK (SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | \ SNDRV_PCM_RATE_88200 | \ -- 2.25.1

3 2

[PATCH] ASoC: mediatek: mt8188: DPCM used FE and BE merged parameters
by Trevor Wu 19 Jul '23

19 Jul '23

To ensure that DPCM takes into account the backend hardware limitations when user space queries the hw_params of a device, we need to add dpcm_merged_format, dpcm_merged_chan, and dpcm_merged_rate to the FE dai_links. This patch includes only stereo FE dai_links, since multi-channel FEs may be reserved for specific purposes. Therefore, it may not be appropriate to consider BE conditions. Signed-off-by: Trevor Wu <trevor.wu(a)mediatek.com> --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index ac69c23e0da1..7c9e08e6a4f5 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -723,6 +723,9 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(playback2), }, [DAI_LINK_DL3_FE] = { @@ -734,6 +737,9 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(playback3), }, [DAI_LINK_DL6_FE] = { @@ -745,6 +751,9 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_playback = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(playback6), }, [DAI_LINK_DL7_FE] = { @@ -833,6 +842,9 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(capture4), }, [DAI_LINK_UL5_FE] = { @@ -844,6 +856,9 @@ static struct snd_soc_dai_link mt8188_mt6359_dai_links[] = { }, .dynamic = 1, .dpcm_capture = 1, + .dpcm_merged_chan = 1, + .dpcm_merged_rate = 1, + .dpcm_merged_format = 1, SND_SOC_DAILINK_REG(capture5), }, [DAI_LINK_UL6_FE] = { -- 2.18.0

2 1

[PATCH] ASoC: amd: acp: fix SND_SOC_AMD_ACP_PCI depdenencies
by Arnd Bergmann 19 Jul '23

19 Jul '23

From: Arnd Bergmann <arnd(a)arndb.de> The new PM functions require code that is part of the snd-acp-legacy-common module: x86_64-linux-ld: sound/soc/amd/acp/acp-pci.o: in function `snd_acp_resume': acp-pci.c:(.text+0x23): undefined reference to `acp_init' x86_64-linux-ld: acp-pci.c:(.text+0x58): undefined reference to `acp_enable_interrupts' x86_64-linux-ld: sound/soc/amd/acp/acp-pci.o: in function `snd_acp_suspend': acp-pci.c:(.text+0x89): undefined reference to `acp_deinit' x86_64-linux-ld: sound/soc/amd/acp/acp-pci.o: in function `acp_pci_remove': acp-pci.c:(.text+0xec): undefined reference to `acp_deinit' x86_64-linux-ld: sound/soc/amd/acp/acp-pci.o: in function `acp_pci_probe': acp-pci.c:(.text+0x26b): undefined reference to `acp_init' Select that Kconfig symbol as is done for the other frontends. Fixes: 088a40980efbc ("ASoC: amd: acp: add pm ops support for acp pci driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/amd/acp/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp/Kconfig b/sound/soc/amd/acp/Kconfig index 9e31b5d167908..631cdf96d6376 100644 --- a/sound/soc/amd/acp/Kconfig +++ b/sound/soc/amd/acp/Kconfig @@ -30,6 +30,7 @@ config SND_SOC_AMD_ACP_PCM config SND_SOC_AMD_ACP_PCI tristate "AMD ACP PCI Driver Support" + select SND_SOC_AMD_ACP_LEGACY_COMMON depends on X86 && PCI help This options enables generic PCI driver for ACP device. -- 2.39.2

2 1

[PATCH 0/7] sound: Use -EPROBE_DEFER instead of i915 module loading.
by Maarten Lankhorst 19 Jul '23

19 Jul '23

Explicitly loading i915 becomes a problem when upstreaming the new intel driver for Tiger Lake and higher graphics (xe). By loading i915, it doesn't wait for driver load of xe, and will fail completely before it loads. -EPROBE_DEFER has to be returned before any device is created in probe(), otherwise the removal of the device will cause EPROBE_DEFER to try again in an infinite loop. The conversion is done in gradual steps. First I add an argument to snd_hdac_i915_init to allow for -EPROBE_DEFER so I can convert each driver separately. Then I convert each driver to move snd_hdac_i915_init out of the workqueue. Finally I drop the ability to choose modprobe behavior after the last user is converted. I suspect the avs and skylake drivers used snd_hdac_i915_init purely for the modprobe, but I don't have the hardware to test if it can be safely removed. It can still be done easily in a followup patch to simplify probing. Maarten Lankhorst (7): ALSA: hda/i915: Add an allow_modprobe argument to snd_hdac_i915_init ALSA: hda/i915: Allow xe as match for i915_component_master_match ASoC: Intel: avs: Move snd_hdac_i915_init to before probe_work. ASoC: Intel: Skylake: Move snd_hdac_i915_init to before probe_work. ALSA: hda/intel: Move snd_hdac_i915_init to before probe_work. ASoC: SOF: Intel: Remove deferred probe for SOF ALSA: hda/i915: Remove extra argument from snd_hdac_i915_init Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Bard Liao <yung-chuan.liao(a)linux.intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Cc: sound-open-firmware(a)alsa-project.org sound/hda/hdac_i915.c | 15 +++------ sound/pci/hda/hda_intel.c | 58 +++++++++++++++++++---------------- sound/soc/intel/avs/core.c | 13 +++++--- sound/soc/intel/skylake/skl.c | 31 ++++++------------- sound/soc/sof/Kconfig | 19 ------------ sound/soc/sof/core.c | 38 ++--------------------- sound/soc/sof/intel/Kconfig | 1 - sound/soc/sof/intel/hda.c | 32 +++++++++++-------- sound/soc/sof/sof-pci-dev.c | 3 +- sound/soc/sof/sof-priv.h | 5 --- 10 files changed, 75 insertions(+), 140 deletions(-) -- 2.39.2

5 27

[PATCH v2 0/5] ASoC: Improve coverage in default KUnit runs
by Mark Brown 19 Jul '23

19 Jul '23

We have some KUnit tests for ASoC but they're not being run as much as they should be since ASoC isn't enabled in the configs used by default with KUnit and in the case of the topology tests there is no way to enable them without enabling drivers that use them. This series provides a Kconfig option which KUnit can use directly rather than worry about drivers. Further, since KUnit is typically run in UML but ALSA prevents build with UML we need to remove that Kconfig conflict. As far as I can tell the motiviation for this is that many ALSA drivers use iomem APIs which are not available under UML and it's more trouble than it's worth to go through and add per driver dependencies. In order to avoid these issues we also provide stubs for these APIs so there are no build time issues if a driver relies on iomem but does not depend on it. With these stubs I am able to build all the sound drivers available in a UML defconfig (UML allmodconfig appears to have substantial other issues in a quick test). With this series I am able to run the topology KUnit tests as part of a kunit --alltests run. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Changes in v2: - Add support for building ALSA with UML. - Link to v1: https://lore.kernel.org/r/20230712-asoc-topology-kunit-enable-v1-0-b9f2da9d… --- Mark Brown (5): driver core: Provide stubs for !IOMEM builds platform: Provide stubs for !HAS_IOMEM builds ALSA: Enable build with UML kunit: Enable ASoC in all_tests.config ASoC: topology: Add explicit build option include/linux/device.h | 26 ++++++++++++++++++++++++++ include/linux/platform_device.h | 28 ++++++++++++++++++++++++++++ sound/Kconfig | 4 ---- sound/soc/Kconfig | 11 +++++++++++ tools/testing/kunit/configs/all_tests.config | 5 +++++ 5 files changed, 70 insertions(+), 4 deletions(-) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230701-asoc-topology-kunit-enable-5e8dd50d0ed7 Best regards, -- Mark Brown <broonie(a)kernel.org>

4 14

[PATCH 0/2] ASoC: Improve coverage in default KUnit runs
by Mark Brown 19 Jul '23

19 Jul '23

We have some KUnit tests for ASoC but they're not being run as much as they should be since ASoC isn't enabled in the configs used by default with KUnit and in the case of the topolofy tests there is no way to enable them without enabling drivers that use them. Let's improve that. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (2): kunit: Enable ASoC in all_tests.config ASoC: topology: Add explicit build option sound/soc/Kconfig | 11 +++++++++++ tools/testing/kunit/configs/all_tests.config | 5 +++++ 2 files changed, 16 insertions(+) --- base-commit: 06c2afb862f9da8dc5efa4b6076a0e48c3fbaaa5 change-id: 20230701-asoc-topology-kunit-enable-5e8dd50d0ed7 Best regards, -- Mark Brown <broonie(a)kernel.org>

2 6

[PATCH 0/3] regmap: Add interface for checking if a register is cached
by Mark Brown 19 Jul '23

19 Jul '23

HDA has a use case for checking if a register is present in the cache which it awkwardly open codes with use of _cache_only() and a read, provide a direct API for this. --- Mark Brown (3): regmap: Let users check if a register is cached regmap: Provide test for regcache_reg_present() ALSA: hda: Use regcache_reg_cached() rather than open coding drivers/base/regmap/regcache.c | 23 ++++++++++++++++++++++ drivers/base/regmap/regmap-kunit.c | 40 ++++++++++++++++++++++++++++++++++++++ include/linux/regmap.h | 1 + sound/hda/hdac_regmap.c | 9 +++------ 4 files changed, 67 insertions(+), 6 deletions(-) --- base-commit: 3953d5c79c21defa716624a8623c4157c0f2fee0 change-id: 20230716-regmap-cache-check-6f6939f41ed5 Best regards, -- Mark Brown <broonie(a)kernel.org>

2 6

Fwd: Odd issue with ALSA connecting directly to HDMI audio output
by Ico Bukvic 19 Jul '23

19 Jul '23

Hi all, I have a consumer-level setup audio question. My research project uses a Raspberry Pi in conjunction with a small portable monitor that has built-in loudspeakers that are to be used via HDMI connection. The goal here is portability and minimal cost. Curiously, when trying to connect to monitor's loudspeakers (via HDMI) using ALSA, no matter what settings I pick, at worst I cannot get audio out, and at best, get a buzzy sound whose waveform is in the attached image. The sound is clearly periodic and impulse-like. It happens every 59 samples. Changing sample rate increases pitch, suggesting 59 samples remain constant regardless of the sampling rate. When playing the actual audio, you can hear it in the background, but this buzz is easily overpowering it. Changing sample rate or bit depth, or buffering, makes no difference. Selecting 8 channels for output (one of HDMI's standards, as far as I can tell), sometimes manages to output a tiny chunk of a good audio sample, and then goes silent. Now, even more curiously, when using the portaudio backend and connecting to the default pulseaudio sink, works just fine. What could be causing this discrepancy? Best, Ico -- Ivica Ico Bukvic, D.M.A. Director, Creativity + Innovation Director, Human-Centered Design iPhD Institute for Creativity, Arts, and Technology Virginia Tech Creative Technologies in Music School of Performing Arts – 0141 Blacksburg, VA 24061 (540) 231-6139 ico(a)vt.edu ci.icat.vt.edu l2ork.icat.vt.edu ico.bukvic.net

3 7

[PATCH] selftests: ALSA: Add test-pcmtest-driver to .gitignore
by Takashi Iwai 19 Jul '23

19 Jul '23

It was forgotten to add the new binary to .gitignore. Let's fix it. Fixes: 10b98a4db11a ("selftests: ALSA: Add test for the 'pcmtest' driver") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- tools/testing/selftests/alsa/.gitignore | 1 + 1 file changed, 1 insertion(+) diff --git a/tools/testing/selftests/alsa/.gitignore b/tools/testing/selftests/alsa/.gitignore index 2b0d11797f25..12dc3fcd3456 100644 --- a/tools/testing/selftests/alsa/.gitignore +++ b/tools/testing/selftests/alsa/.gitignore @@ -1,2 +1,3 @@ mixer-test pcm-test +test-pcmtest-driver -- 2.35.3

2 1

[PATCH 0/6] Add audio support in v4l2 framework
by Shengjiu Wang 19 Jul '23

19 Jul '23

Audio signal processing has the requirement for memory to memory similar as Video. This patch is to add this support in v4l2 framework, defined new buffer type V4L2_BUF_TYPE_AUDIO_CAPTURE and V4L2_BUF_TYPE_AUDIO_OUTPUT, defined new format v4l2_audio_format for audio case usage. The created audio device is named "/dev/audioX". And add memory to memory support for two kinds of i.MX ASRC module Shengjiu Wang (6): media: v4l2: Add audio capture and output support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: Add memory to memory driver ASoC: fsl_asrc: enable memory to memory function ASoC: fsl_easrc: enable memory to memory function .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/v4l2-core/v4l2-dev.c | 17 + drivers/media/v4l2-core/v4l2-ioctl.c | 52 ++ include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + include/uapi/linux/videodev2.h | 19 + sound/soc/fsl/Kconfig | 13 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_asrc.c | 175 +++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 54 ++ sound/soc/fsl/fsl_asrc_m2m.c | 878 ++++++++++++++++++ sound/soc/fsl/fsl_asrc_m2m.h | 48 + sound/soc/fsl/fsl_easrc.c | 255 ++++- sound/soc/fsl/fsl_easrc.h | 6 + 15 files changed, 1557 insertions(+), 4 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c create mode 100644 sound/soc/fsl/fsl_asrc_m2m.h -- 2.34.1

7 23

Re: [BUG REPORT] sound: pci: ctxfi: htimestamp dose not work on EMU20k during recording
by Takashi Iwai 19 Jul '23

19 Jul '23

[Please keep ML in Cc.] On Wed, 19 Jul 2023 10:10:47 +0200, Carl Klemm wrote: > > Hi, > > The values only seam nonsensical if playback and record is active at > the same time, otherwise eatch on its own is fine That's an interesting point. You mean "values" as the PCM positions? That is, when running in full-duplex mode, only the capture stream shows strange PCM positions, while PCM playback shows correctly? Or are both screwed up? And, how strange they are? They jump up/down, random values, or any patterns? thanks, Takashi > > Carl Klemm > > On Wed, 2023-07-19 at 09:27 +0200, Takashi Iwai wrote: > > On Wed, 19 Jul 2023 09:10:12 +0200, > > Carl Klemm wrote: > > > > > > Hi, > > > > > > see https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3352 > > > htimestamp returns nonesensical values when recording is active on > > > a > > > EMU20k device > > > > The driver doesn't provide any own implementation of PCM tstamp, > > hence > > it must be the standard values. That said, the incorrect values are > > rather the "avail", that is the PCM position, not the timestamp > > itself, I suppose. > > > > Is it only about recording, i.e. playback doesn't show any wrong > > values? > > > > In anyway, try to enable the tracing for snd_pcm, get the position > > and > > the system timestamp during recording, and verify whether the > > reported > > values are reasonable or not. > > > > > > Takashi >

1 0

[PATCH] ALSA: hda/realtek: Enable Mute LED on HP Laptop 15s-eq2xxx
by Luka Guzenko 19 Jul '23

19 Jul '23

The HP Laptop 15s-eq2xxx uses ALC236 codec and controls the mute LED using COEF 0x07 index 1. No existing quirk covers this configuration. Adds a new quirk and enables it for the device. Signed-off-by: Luka Guzenko <l.guzenko(a)web.de> --- sound/pci/hda/patch_realtek.c | 23 ++++++++++++++++++++++- 1 file changed, 20 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 308ec7034cc9..e8a836596bfb 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4623,6 +4623,21 @@ static void alc236_fixup_hp_mute_led_coefbit(struct hda_codec *codec, } } +static void alc236_fixup_hp_mute_led_coefbit2(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->mute_led_polarity = 0; + spec->mute_led_coef.idx = 0x07; + spec->mute_led_coef.mask = 1; + spec->mute_led_coef.on = 1; + spec->mute_led_coef.off = 0; + snd_hda_gen_add_mute_led_cdev(codec, coef_mute_led_set); + } +} + /* turn on/off mic-mute LED per capture hook by coef bit */ static int coef_micmute_led_set(struct led_classdev *led_cdev, enum led_brightness brightness) @@ -7133,6 +7148,7 @@ enum { ALC285_FIXUP_HP_GPIO_LED, ALC285_FIXUP_HP_MUTE_LED, ALC285_FIXUP_HP_SPECTRE_X360_MUTE_LED, + ALC236_FIXUP_HP_MUTE_LED_COEFBIT2, ALC236_FIXUP_HP_GPIO_LED, ALC236_FIXUP_HP_MUTE_LED, ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF, @@ -8556,6 +8572,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_hp_spectre_x360_mute_led, }, + [ALC236_FIXUP_HP_MUTE_LED_COEFBIT2] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc236_fixup_hp_mute_led_coefbit2, + }, [ALC236_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc236_fixup_hp_gpio_led, @@ -9440,6 +9460,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x886d, "HP ZBook Fury 17.3 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), + SND_PCI_QUIRK(0x103c, 0x887a, "HP Laptop 15s-eq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8895, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED), SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), -- 2.41.0

2 1

[BUG REPORT] sound: pci: ctxfi: htimestamp dose not work on EMU20k during recording
by Carl Klemm 19 Jul '23

19 Jul '23

Hi, see https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3352 htimestamp returns nonesensical values when recording is active on a EMU20k device lspci: 03:00.0 Audio device: Creative Labs EMU20k2 [Sound Blaster X-Fi Titanium Series] (rev 03) Subsystem: Creative Labs EMU20k2 [Sound Blaster X-Fi Titanium Series] Flags: bus master, fast devsel, latency 0, IRQ 17 Memory at d1200000 (64-bit, non-prefetchable) [size=64K] Memory at d1000000 (64-bit, non-prefetchable) [size=2M] Memory at d0000000 (64-bit, non-prefetchable) [size=16M] Capabilities: <access denied> Kernel driver in use: snd_ctxfi Kernel modules: snd_ctxfi With regards, Carl Klemm

2 1

[PATCH v2] Add support for trigger-start/stop for simple audio card
by Maxim Kochetkov 19 Jul '23

19 Jul '23

Support for trigger-start/stop for simple audio card. Maxim Kochetkov (3): ASoC: simple-card-utils: introduce asoc_simple_parse_triggers() ASoC: dt-bindings: simple-card: add triggers properties ASoC: simple-card: add triggers parsing from DT node support .../bindings/sound/simple-card.yaml | 31 +++++++++++++ include/sound/simple_card_utils.h | 3 ++ sound/soc/generic/simple-card-utils.c | 45 +++++++++++++++++++ sound/soc/generic/simple-card.c | 4 ++ 4 files changed, 83 insertions(+) -- 2.40.1

2 5

[PATCH V1 0/5] ASoC: codecs: Add Awinic AW88261 audio amplifier driver
by wangweidong.a＠awinic.com 19 Jul '23

19 Jul '23

From: Weidong Wang <wangweidong.a(a)awinic.com> The AW88261 is an I2S/TDM input, high efficiency digital Smart K audio amplifier with an integrated 10.25V smart boost convert Add a DT schema for describing Awinic aw88261 audio amplifiers. They are controlled using I2C. Weidong Wang (5): ASoC: codecs: Add i2c and codec registration for aw88261 and their associated operation functions ASoC: codecs: aw88261 function for ALSA Audio Driver ASoC: codecs: ACF bin parsing and check library file for aw88261 ASoC: codecs: aw88261 chip register file, data type file and Kconfig Makefile ASoC: dt-bindings: Add schema for "awinic,aw88261" .../bindings/sound/awinic,aw88261.yaml | 53 + sound/soc/codecs/Kconfig | 17 + sound/soc/codecs/Makefile | 5 + sound/soc/codecs/aw88261/aw88261.c | 539 ++++++++++ sound/soc/codecs/aw88261/aw88261.h | 52 + sound/soc/codecs/aw88261/aw88261_data_type.h | 144 +++ sound/soc/codecs/aw88261/aw88261_device.c | 959 +++++++++++++++++ sound/soc/codecs/aw88261/aw88261_device.h | 114 ++ sound/soc/codecs/aw88261/aw88261_lib.c | 997 ++++++++++++++++++ sound/soc/codecs/aw88261/aw88261_lib.h | 91 ++ sound/soc/codecs/aw88261/aw88261_reg.h | 377 +++++++ 11 files changed, 3348 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw88261.yaml create mode 100644 sound/soc/codecs/aw88261/aw88261.c create mode 100644 sound/soc/codecs/aw88261/aw88261.h create mode 100644 sound/soc/codecs/aw88261/aw88261_data_type.h create mode 100644 sound/soc/codecs/aw88261/aw88261_device.c create mode 100644 sound/soc/codecs/aw88261/aw88261_device.h create mode 100644 sound/soc/codecs/aw88261/aw88261_lib.c create mode 100644 sound/soc/codecs/aw88261/aw88261_lib.h create mode 100644 sound/soc/codecs/aw88261/aw88261_reg.h base-commit: 8d7868c41df58edabc4e408d119a1aef58a54d9d -- 2.41.0

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