March 2024 - Alsa-devel - mailman.alsa-project.org

No sound "dummy output" - snd_hda_intel on Pop OS 22.04 LTS on Acer chromebook plus 514 (Google Markath rev 3)
by GitHub issues - opened 23 Mar '24

23 Mar '24

alsa-project/alsa-lib issue #391 was opened from judicandus: Installed a fresh Pop OS installation on an Acer Chromebook Plus 514, but audio is not working with just one interface available "Dummy Output". Alsa-info attached: [alsa-info.txt](https://github.com/alsa-project/alsa-lib/files/14733718/alsa… **1) What I already tried to do** All the instructions in: https://support.system76.com/articles/audio/ , specifically: i. reinstalling pipewire and deleting configurations: systemctl --user restart wireplumber pipewire pipewire-pulse rm -r ~/.config/pulse/* rm -r ~/.local/state/wireplumber/* ii. reinstalling apps sudo apt install --reinstall alsa-base alsa-utils linux-sound-base libasound2 sudo apt reinstall libpipewire-0.3-0 libpipewire-0.3-common libpipewire-0.3-modules pipewire pipewire-audio-client-libraries pipewire-bin pipewire-pulse sudo alsa force-reload iii. snh_hda_intel specific bug fix echo "options snd-hda-intel dmic_detect=0" | sudo tee -a /etc/modprobe.d/alsa-base.conf echo "blacklist snd_soc_skl" | sudo tee -a /etc/modprobe.d/blacklist.conf (reboot) iv. installing SOF driver according to https://github.com/thesofproject/sof-bin from the tarball Nothing worked. **2) Output** aplay -l **** Liste des périphériques matériels PLAYBACK **** carte 0 : Generic [HD-Audio Generic], périphérique 3 : HDMI 0 [HDMI 0] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 7 : HDMI 1 [HDMI 1] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 8 : HDMI 2 [HDMI 2] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 carte 0 : Generic [HD-Audio Generic], périphérique 9 : HDMI 3 [HDMI 3] Sous-périphériques : 1/1 Sous-périphérique #0 : subdevice #0 arecord -l **** Liste des périphériques matériels CAPTURE **** lsmod | grep snd_hda_intel snd_hda_intel 61440 1 snd_intel_dspcfg 36864 2 snd_hda_intel,snd_sof snd_hda_codec 221184 2 snd_hda_codec_hdmi,snd_hda_intel snd_hda_core 151552 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_pcm 196608 14 snd_sof_amd_acp,snd_hda_codec_hdmi,snd_pci_acp6x,snd_hda_intel,snd_hda_codec,snd_sof,snd_compress,snd_soc_core,snd_sof_utils,snd_hda_core,snd_soc_rt1019,snd_pci_ps,snd_soc_rt5682s,snd_pcm_dmaengine snd 147456 16 snd_seq,snd_seq_device,snd_hda_codec_hdmi,snd_hwdep,snd_hda_intel,snd_hda_codec,snd_sof,snd_timer,snd_compress,snd_soc_core,snd_pcm,snd_rawmidi lspci -nnk | grep -A2 Audio 04:00.1 Audio device [0403]: Advanced Micro Devices, Inc. [AMD/ATI] Device [1002:1640] Subsystem: Advanced Micro Devices, Inc. [AMD/ATI] Device [1002:1640] Kernel driver in use: snd_hda_intel -- 04:00.5 Multimedia controller [0480]: Advanced Micro Devices, Inc. [AMD] Raven/Raven2/FireFlight/Renoir Audio Processor [1022:15e2] (rev 6f) DeviceName: Multimedia controller Subsystem: Advanced Micro Devices, Inc. [AMD] Raven/Raven2/FireFlight/Renoir Audio Processor [1022:15e2] Kernel driver in use: snd_sof_amd_rembrandt Kernel modules: snd_pci_acp3x, snd_rn_pci_acp3x, snd_pci_acp5x, snd_pci_acp6x, snd_acp_pci, snd_rpl_pci_acp6x, snd_pci_ps, snd_sof_amd_renoir, snd_sof_amd_rembrandt, snd_sof_amd_vangogh cat /proc/asound/cards 0 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xd06c0000 irq 77 sudo dmesg | grep -i -e audio -e error -e fail [ 0.282615] ACPI: _OSC evaluation for CPUs failed, trying _PDC [ 0.629902] ACPI Error: Could not enable SleepButton event (20230628/evxfevnt-182) [ 6.100083] RAS: Correctable Errors collector initialized. [ 6.327488] hub 6-0:1.0: config failed, hub doesn't have any ports! (err -19) [ 8.787783] system76_acpi: module verification failed: signature and/or required key missing - tainting kernel [ 10.353860] snd_hda_intel 0000:04:00.1: bound 0000:04:00.0 (ops amdgpu_dm_audio_component_bind_ops [amdgpu]) [ 10.454926] input: HD-Audio Generic HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input10 [ 10.461777] input: HD-Audio Generic HDMI/DP,pcm=7 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input11 [ 10.491432] input: HD-Audio Generic HDMI/DP,pcm=8 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input12 [ 10.518137] input: HD-Audio Generic HDMI/DP,pcm=9 as /devices/pci0000:00/0000:00:08.1/0000:04:00.1/sound/card0/input13 [ 10.659161] snd_sof_amd_rembrandt 0000:04:00.5: Direct firmware load for amd/sof/community/sof-rmb.ri failed with error -2 [ 10.659196] snd_sof_amd_rembrandt 0000:04:00.5: error: sof firmware file is missing, you might need to [ 10.659208] snd_sof_amd_rembrandt 0000:04:00.5: error: failed to load DSP firmware -2 [ 10.659441] snd_sof_amd_rembrandt 0000:04:00.5: error: sof_probe_work failed err: -2 systemctl --user status pipewire pipewire.service - PipeWire Multimedia Service Loaded: loaded (/usr/lib/systemd/user/pipewire.service; enabled; vendor preset: enabled) Active: active (running) since Sat 2024-03-23 22:15:36 CET; 58min ago TriggeredBy: ● pipewire.socket Main PID: 1320 (pipewire) Tasks: 3 (limit: 9239) Memory: 6.6M CPU: 335ms CGroup: /user.slice/user-1000.slice/user(a)1000.service/session.slice/pipewire.service └─1320 /usr/bin/pipewire mars 23 22:15:36 pop-os systemd[1310]: Started PipeWire Multimedia Service. mars 23 22:15:36 pop-os pipewire[1320]: mod.jackdbus-detect: Failed to receive jackdbus reply: org.freedesktop.DBus.Error.ServiceUnknown: The name is not activatable I am not sure if this is a bug but since I tried all solutions I could find on the internet, I thought I might try my luck here. Thank you beforehand for any assistance or insight. Issue URL : https://github.com/alsa-project/alsa-lib/issues/391 Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH] amdtp-stream: Checking a variable for zero before dividing
by Andrey Shumilin 23 Mar '24

23 Mar '24

The step variable is initialized to zero. It changes in the loop, but if it doesn’t change it will remain zero. The patch added a variable check. Found by Linux Verification Center (linuxtesting.org) with SVACE. Signed-off-by: Andrey Shumilin <shum.sdl(a)nppct.ru> --- sound/firewire/amdtp-stream.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index c9f153f85ae6..b61aa30f43a9 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -168,7 +168,8 @@ static int apply_constraint_to_size(struct snd_pcm_hw_params *params, if (snd_interval_test(r, amdtp_rate_table[i])) step = max(step, amdtp_syt_intervals[i]); } - + if (step == 0) + return -EINVAL; t.min = roundup(s->min, step); t.max = rounddown(s->max, step); t.integer = 1; -- 2.30.2

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Variable mclk-fs in ASoC audio cards?
by John Watts 22 Mar '24

22 Mar '24

Greetings fellow ALSAers :) Over the past month I have been working on a driver for the 8-channel TDM CS5368 audio codec. However I've run in to an interesting snag that seems to be an open problem. Let me explain. The audio codec generally runs at 8 channels of 32-bit data, so: MCLK = BCLK = LRCK * 256 However it is possible to read 4 or 2 channels instead: MCLK = BCLK = LRCK * 128 MCLK = BCLK = LRCK * 64 This trade-off allows avoiding high frequency clocking at the expense of a reduced channel count. From what I current see the way to indicate the multiplier here is using mclk-fs in the device tree, but this would mean effectively fixing the channel count in the device tree. I have searched around found someone has suggested mapping mclk-fs to rates here: https://lore.kernel.org/all/CAEnQRZDpJu27_pB=-ZYQjsNPC8eA4-Ur-36niT2hCosMWt… This functionality is somewhat adjacent to it, but I was wondering if it would be better to somehow factor dai-tdm-slot-* in to mclk-fs calculation. Perhaps mclk-fs could be divided by dai-tdm-slot-num then multiplied by slots? Has anyone hit this problem before? How did you solve it? John.

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[PATCH] [RESEND] ASoC: amd: simplify soundwire dependencies for
by Vijendar Mukunda 22 Mar '24

22 Mar '24

The CONFIG_SOUNDWIRE_AMD is a user-visible option, it should be never selected by another driver. So replace the extra complexity with a normal Kconfig dependency in SND_SOC_AMD_SOUNDWIRE. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/Kconfig | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index fa74635cee08..3508f5a96b75 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -134,15 +134,14 @@ config SND_SOC_AMD_RPL_ACP6x config SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE tristate - select SOUNDWIRE_AMD if SND_SOC_AMD_SOUNDWIRE != n select SND_AMD_SOUNDWIRE_ACPI if ACPI config SND_SOC_AMD_SOUNDWIRE tristate "Support for SoundWire based AMD platforms" default SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE depends on SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE - depends on ACPI && SOUNDWIRE - depends on !(SOUNDWIRE=m && SND_SOC_AMD_SOUNDWIRE_LINK_BASELINE=y) + depends on ACPI + depends on SOUNDWIRE_AMD help This adds support for SoundWire for AMD platforms. Say Y if you want to enable SoundWire links with SOF. -- 2.34.1

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[GIT PULL] ASoC fixes for v6.9-merge-window
by Mark Brown 21 Mar '24

21 Mar '24

The following changes since commit e8f897f4afef0031fe618a8e94127a0934896aba: Linux 6.8 (2024-03-10 13:38:09 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.9-merge-window for you to fetch changes up to 9a8b202f8cb7ebebc71f1f2a353a21c76d3063a8: ASoC: soc-compress: Fix and add DPCM locking (2024-03-18 14:41:51 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.9 A bunch of fixes that came in during the merge window, probably the most substantial thing is the DPCM locking fix for compressed audio which has been lurking for a while. ---------------------------------------------------------------- Chancel Liu (1): ASoC: soc-core.c: Skip dummy codec when adding platforms Cristian Ciocaltea (2): ASoC: SOF: amd: Move signed_fw_image to struct acp_quirk_entry ASoC: SOF: amd: Skip IRAM/DRAM size modification for Steam Deck OLED Jiawei Wang (2): ASoC: amd: yc: Revert "Fix non-functional mic on Lenovo 21J2" ASoC: amd: yc: Revert "add new YC platform variant (0x63) support" Luca Ceresoli (1): ASoC: rockchip: i2s-tdm: Fix inaccurate sampling rates M Cooley (1): ASoC: amd: yc: Fix non-functional mic on ASUS M7600RE Mark Brown (3): ASoC: Merge up release Add support for the internal RK3308 audio codec ASoC: SOF: amd: Skip IRAM/DRAM size modification Rob Herring (1): ASoC: dt-bindings: cirrus,cs42l43: Fix 'gpio-ranges' schema Shalini Manjunatha (1): ASoC: soc-compress: Fix and add DPCM locking Uwe Kleine-König (1): ASoC: tlv320adc3xxx: Don't strip remove function when driver is builtin .../ABI/testing/sysfs-bus-pci-devices-avs | 8 + .../bindings/sound/atmel,asoc-wm8904.yaml | 84 + .../bindings/sound/atmel,sam9x5-wm8731-audio.yaml | 76 + .../bindings/sound/atmel,sama5d2-classd.yaml | 7 +- .../bindings/sound/atmel-sam9x5-wm8731-audio.txt | 35 - .../devicetree/bindings/sound/atmel-wm8904.txt | 55 - .../bindings/sound/audio-graph-port.yaml | 2 +- .../devicetree/bindings/sound/cirrus,cs35l45.yaml | 3 + .../devicetree/bindings/sound/cirrus,cs42l43.yaml | 11 +- .../devicetree/bindings/sound/everest,es8326.yaml | 8 +- .../devicetree/bindings/sound/fsl,asrc.txt | 80 - .../devicetree/bindings/sound/fsl,easrc.yaml | 4 +- .../devicetree/bindings/sound/fsl,imx-asrc.yaml | 162 + .../devicetree/bindings/sound/fsl,micfil.yaml | 14 +- .../devicetree/bindings/sound/fsl,sai.yaml | 6 + .../bindings/sound/infineon,peb2466.yaml | 2 +- .../bindings/sound/microchip,sama7g5-i2smcc.yaml | 11 +- .../devicetree/bindings/sound/qcom,sm8250.yaml | 2 +- .../devicetree/bindings/sound/qcom,wcd938x.yaml | 81 +- .../bindings/sound/qcom,wcd939x-sdw.yaml | 69 + .../devicetree/bindings/sound/qcom,wcd939x.yaml | 96 + .../bindings/sound/qcom,wcd93xx-common.yaml | 95 + .../devicetree/bindings/sound/qcom,wsa8840.yaml | 11 +- .../devicetree/bindings/sound/realtek,rt1015.yaml | 41 + Documentation/devicetree/bindings/sound/rt1015.txt | 23 - .../devicetree/bindings/sound/samsung,tm2.yaml | 7 +- drivers/firmware/cirrus/cs_dsp.c | 7 +- drivers/soundwire/Makefile | 2 +- drivers/soundwire/amd_init.c | 235 ++ drivers/soundwire/amd_init.h | 13 + drivers/soundwire/amd_manager.c | 47 +- drivers/soundwire/amd_manager.h | 16 +- drivers/soundwire/dmi-quirks.c | 8 + include/linux/firmware/cirrus/cs_dsp.h | 1 - include/linux/soundwire/sdw_amd.h | 83 +- include/linux/spi/spi.h | 2 - include/sound/cs-amp-lib.h | 66 + include/sound/cs35l56.h | 10 + include/sound/cs42l42.h | 5 +- include/sound/hda-mlink.h | 2 +- include/sound/hda_register.h | 2 + include/sound/soc.h | 4 +- include/sound/sof/dai-amd.h | 7 + include/sound/sof/dai.h | 2 + include/sound/tas2781.h | 1 - include/trace/events/asoc.h | 45 +- include/uapi/sound/intel/avs/tokens.h | 9 + include/uapi/sound/sof/tokens.h | 4 + sound/pci/hda/Kconfig | 2 + sound/pci/hda/cs35l56_hda.c | 39 +- sound/pci/hda/tas2781_hda_i2c.c | 12 - sound/soc/amd/Kconfig | 17 + sound/soc/amd/Makefile | 2 +- sound/soc/amd/acp/Kconfig | 7 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-mach-common.c | 6 +- sound/soc/amd/acp/acp-sof-mach.c | 26 +- sound/soc/amd/acp/amd-sdw-acpi.c | 62 + sound/soc/amd/ps/Makefile | 2 +- sound/soc/amd/ps/acp63.h | 91 +- sound/soc/amd/ps/pci-ps.c | 565 +-- sound/soc/amd/ps/ps-mach.c | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/amd/yc/acp6x-mach.c | 14 +- sound/soc/amd/yc/pci-acp6x.c | 1 - sound/soc/atmel/mikroe-proto.c | 8 +- sound/soc/codecs/Kconfig | 55 +- sound/soc/codecs/Makefile | 13 + sound/soc/codecs/cs-amp-lib-test.c | 709 ++++ sound/soc/codecs/cs-amp-lib.c | 277 ++ sound/soc/codecs/cs35l56-sdw.c | 20 + sound/soc/codecs/cs35l56-shared.c | 84 + sound/soc/codecs/cs35l56.c | 44 +- sound/soc/codecs/cs42l42.c | 1 - sound/soc/codecs/cs42l43-jack.c | 27 +- sound/soc/codecs/cs42l43-sdw.c | 1 + sound/soc/codecs/cs42l43.c | 82 +- sound/soc/codecs/cs42l43.h | 21 +- sound/soc/codecs/da7213.c | 12 +- sound/soc/codecs/es8326.c | 110 +- sound/soc/codecs/es8326.h | 5 +- sound/soc/codecs/framer-codec.c | 413 +++ sound/soc/codecs/lpass-macro-common.h | 7 + sound/soc/codecs/lpass-tx-macro.c | 693 +++- sound/soc/codecs/lpass-va-macro.c | 57 + sound/soc/codecs/max98363.c | 2 +- sound/soc/codecs/max98373-sdw.c | 2 +- sound/soc/codecs/nau8540.c | 116 +- sound/soc/codecs/nau8540.h | 13 +- sound/soc/codecs/nau8825.c | 12 +- sound/soc/codecs/rt1017-sdca-sdw.c | 2 +- sound/soc/codecs/rt274.c | 2 +- sound/soc/codecs/rt286.c | 2 +- sound/soc/codecs/rt298.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/codecs/rt5514.c | 9 +- sound/soc/codecs/rt5616.c | 9 +- sound/soc/codecs/rt5640.c | 9 +- sound/soc/codecs/rt5645.c | 2 +- sound/soc/codecs/rt5651.c | 2 +- sound/soc/codecs/rt5659.c | 2 +- sound/soc/codecs/rt5660.c | 9 +- sound/soc/codecs/rt5663.c | 2 +- sound/soc/codecs/rt5665.c | 2 +- sound/soc/codecs/rt5668.c | 2 +- sound/soc/codecs/rt5682-i2c.c | 2 +- sound/soc/codecs/rt5682s.c | 2 +- sound/soc/codecs/rt712-sdca-dmic.c | 2 +- sound/soc/codecs/rt712-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/codecs/tas2781-comlib.c | 1 - sound/soc/codecs/tas2781-fmwlib.c | 2 +- sound/soc/codecs/tlv320adc3xxx.c | 4 +- sound/soc/codecs/wcd-clsh-v2.h | 1 + sound/soc/codecs/wcd-mbhc-v2.c | 95 +- sound/soc/codecs/wcd-mbhc-v2.h | 3 + sound/soc/codecs/wcd939x-sdw.c | 1551 ++++++++ sound/soc/codecs/wcd939x.c | 3686 ++++++++++++++++++++ sound/soc/codecs/wcd939x.h | 989 ++++++ sound/soc/codecs/wm_adsp.c | 27 +- sound/soc/codecs/wm_adsp.h | 2 + sound/soc/codecs/wsa884x.c | 53 +- sound/soc/fsl/eukrea-tlv320.c | 8 +- sound/soc/fsl/fsl_sai.c | 13 + sound/soc/fsl/p1022_rdk.c | 33 +- sound/soc/intel/atom/sst/sst_ipc.c | 3 +- sound/soc/intel/avs/Makefile | 5 +- sound/soc/intel/avs/apl.c | 58 +- sound/soc/intel/avs/avs.h | 70 +- sound/soc/intel/avs/board_selection.c | 85 + sound/soc/intel/avs/cnl.c | 61 + sound/soc/intel/avs/core.c | 161 +- sound/soc/intel/avs/icl.c | 197 ++ sound/soc/intel/avs/ipc.c | 66 +- sound/soc/intel/avs/loader.c | 2 +- sound/soc/intel/avs/messages.c | 1 + sound/soc/intel/avs/messages.h | 38 +- sound/soc/intel/avs/path.c | 33 + sound/soc/intel/avs/pcm.c | 77 +- sound/soc/intel/avs/registers.h | 21 +- sound/soc/intel/avs/skl.c | 59 +- sound/soc/intel/avs/sysfs.c | 35 + sound/soc/intel/avs/tgl.c | 54 + sound/soc/intel/avs/topology.c | 164 +- sound/soc/intel/avs/topology.h | 13 + sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/sof_board_helpers.c | 249 +- sound/soc/intel/boards/sof_board_helpers.h | 29 + sound/soc/intel/boards/sof_cs42l42.c | 236 +- sound/soc/intel/boards/sof_rt5682.c | 151 +- sound/soc/intel/boards/sof_sdw.c | 106 +- sound/soc/intel/boards/sof_sdw_common.h | 82 +- sound/soc/intel/boards/sof_sdw_cs42l42.c | 35 +- sound/soc/intel/boards/sof_sdw_cs42l43.c | 25 +- sound/soc/intel/boards/sof_sdw_cs_amp.c | 3 +- sound/soc/intel/boards/sof_sdw_maxim.c | 4 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 35 +- sound/soc/intel/boards/sof_sdw_rt700.c | 35 +- sound/soc/intel/boards/sof_sdw_rt711.c | 19 +- sound/soc/intel/boards/sof_sdw_rt712_sdca.c | 39 +- sound/soc/intel/boards/sof_sdw_rt715.c | 12 +- sound/soc/intel/boards/sof_sdw_rt715_sdca.c | 12 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 53 +- .../soc/intel/boards/sof_sdw_rt_sdca_jack_common.c | 19 +- sound/soc/intel/boards/sof_ssp_common.h | 8 + sound/soc/intel/catpt/dsp.c | 4 +- sound/soc/intel/common/soc-acpi-intel-adl-match.c | 12 +- sound/soc/intel/common/soc-acpi-intel-jsl-match.c | 10 +- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 102 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 61 +- sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 4 +- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 6 +- sound/soc/meson/aiu.c | 19 +- sound/soc/meson/aiu.h | 1 - sound/soc/meson/axg-fifo.c | 25 +- sound/soc/meson/axg-fifo.h | 14 +- sound/soc/meson/axg-frddr.c | 13 +- sound/soc/meson/axg-spdifin.c | 6 +- sound/soc/meson/axg-tdm-interface.c | 31 +- sound/soc/meson/axg-toddr.c | 30 +- sound/soc/meson/t9015.c | 20 +- sound/soc/pxa/Kconfig | 3 - sound/soc/qcom/common.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 352 +- sound/soc/sh/rz-ssi.c | 2 +- sound/soc/soc-compress.c | 4 + sound/soc/soc-core.c | 24 +- sound/soc/soc-dapm.c | 8 +- sound/soc/sof/amd/Kconfig | 18 + sound/soc/sof/amd/acp-common.c | 65 +- sound/soc/sof/amd/acp-dsp-offset.h | 10 + sound/soc/sof/amd/acp-loader.c | 34 +- sound/soc/sof/amd/acp.c | 252 +- sound/soc/sof/amd/acp.h | 33 +- sound/soc/sof/amd/pci-acp63.c | 7 + sound/soc/sof/amd/vangogh.c | 9 +- sound/soc/sof/core.c | 10 + sound/soc/sof/debug.c | 8 +- sound/soc/sof/fw-file-profile.c | 18 +- sound/soc/sof/imx/imx8.c | 16 + sound/soc/sof/imx/imx8m.c | 10 + sound/soc/sof/imx/imx8ulp.c | 10 + sound/soc/sof/intel/hda-common-ops.c | 1 + sound/soc/sof/intel/hda-dai-ops.c | 51 +- sound/soc/sof/intel/hda-dai.c | 17 +- sound/soc/sof/intel/hda-dsp.c | 5 + sound/soc/sof/intel/hda-stream.c | 9 + sound/soc/sof/intel/hda.c | 80 +- sound/soc/sof/intel/hda.h | 5 + sound/soc/sof/intel/lnl.c | 24 +- sound/soc/sof/ipc3-loader.c | 2 + sound/soc/sof/ipc3-pcm.c | 25 + sound/soc/sof/ipc3-topology.c | 40 + sound/soc/sof/ipc4-pcm.c | 19 +- sound/soc/sof/ipc4-priv.h | 4 + sound/soc/sof/ipc4-topology.c | 28 +- sound/soc/sof/ops.h | 9 + sound/soc/sof/sof-audio.c | 8 +- sound/soc/sof/sof-audio.h | 2 + sound/soc/sof/sof-priv.h | 10 + sound/soc/sof/topology.c | 30 +- sound/soc/ti/j721e-evm.c | 4 +- sound/soc/ti/omap-hdmi.c | 10 +- 224 files changed, 13139 insertions(+), 2591 deletions(-) create mode 100644 Documentation/ABI/testing/sysfs-bus-pci-devices-avs create mode 100644 Documentation/devicetree/bindings/sound/atmel,asoc-wm8904.yaml create mode 100644 Documentation/devicetree/bindings/sound/atmel,sam9x5-wm8731-audio.yaml delete mode 100644 Documentation/devicetree/bindings/sound/atmel-sam9x5-wm8731-audio.txt delete mode 100644 Documentation/devicetree/bindings/sound/atmel-wm8904.txt delete mode 100644 Documentation/devicetree/bindings/sound/fsl,asrc.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-asrc.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd939x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd939x.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd93xx-common.yaml create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt1015.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt1015.txt create mode 100644 drivers/soundwire/amd_init.c create mode 100644 drivers/soundwire/amd_init.h create mode 100644 include/sound/cs-amp-lib.h create mode 100644 sound/soc/amd/acp/amd-sdw-acpi.c create mode 100644 sound/soc/codecs/cs-amp-lib-test.c create mode 100644 sound/soc/codecs/cs-amp-lib.c create mode 100644 sound/soc/codecs/framer-codec.c create mode 100644 sound/soc/codecs/wcd939x-sdw.c create mode 100644 sound/soc/codecs/wcd939x.c create mode 100644 sound/soc/codecs/wcd939x.h create mode 100644 sound/soc/intel/avs/cnl.c create mode 100644 sound/soc/intel/avs/icl.c create mode 100644 sound/soc/intel/avs/sysfs.c create mode 100644 sound/soc/intel/avs/tgl.c

2 1

synaptics hi-res audio usb device duplex, usb bandwidth issues
by Ian Malone 21 Mar '24

21 Mar '24

Hi, In short I have a USB to 3.5mm adapter which seems not to work in duplex mode on USB2.0 systems, possibly due to a bandwidth calculation bug. This is an evolution of an issue I previously posted on the users list with no luck. I have an Anker USB-C to 3.5mm audio dongle (lsusb: Conexant Systems (Rockwell), Inc. Hi-Res Audio) which I've used for some time on my phone (Android with USB-3.2). On trying to use it with an older T420 laptop recently with only USB-2.0 ports I discovered it will not work in duplex mode. Input-only and output-only profiles work (tested recording and playback with audacity), but with duplex no sound is recorded (Fedora 39, pipewire). This is easily reproduced by looking at the pavucontrol volume monitor for Output Devices, if I switch the device to Analog or Digital Input in configuration then the Input Devices level monitor for it shows activity if I speak into or tap the microphone. With duplex selected there is no activity, the level monitor bar may or may not display. I can switch between the two behaviours by changing the profile. Various applications such as Audacity and Zoom appear to hang when accessing this microphone with the duplex profile set. I've used pipewire configuration to force the format to 16LE only (playback and recording), but this has not helped. In dmesg this error appears when this happens (microphone opened, for example by pavucontrol): [ 294.825544] usb 1-1.1: cannot submit urb 0, error -28: not enough bandwidth (T420, Fedora 39, kernel 6.7.5) The T420 has USB 2 Type A ports, so a Type C to Type A adapter is needed, but the problem is not limited to the T420 laptop. On an older Samsung Tablet (Tab-A 2019, USB-2.0 Type C, kernel 4.4.177) I've been able to test using Google Meet. Although there are no errors or freezing observed, in headphone mode over USB the headphone microphone is not being used and the tablet uses its onboard microphone instead (although headphones are selected and playback is over the headphones). Again, I can record (via voice recorder) from the headset microphone, just no duplex. Trying google meet on my phone (USB3.2) I do get duplex over the USB-3.5 adapter. On a newer laptop with USB-3.2 and a mix of type A and C ports I also get duplex (F37, kernel 6.5.12), however dmesg shows even here not all is well, using the type A and C ports on the side: [ 9172.326602] usb 3-1: new full-speed USB device number 6 using xhci_hcd [ 9172.490169] usb 3-1: New USB device found, idVendor=0572, idProduct=1b08, bcdDevice= 0.10 [ 9172.490174] usb 3-1: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 9172.490177] usb 3-1: Product: Hi-Res Audio [ 9172.490178] usb 3-1: Manufacturer: Synaptics [ 9172.490180] usb 3-1: SerialNumber: 000000000000000000000000 [ 9172.683435] input: Synaptics Hi-Res Audio as /devices/pci0000:00/0000:00:08.1/0000:05:00.4/usb3/3-1/3-1:1.3/0003:0572:1B08.000A/input/input36 [ 9172.736025] hid-generic 0003:0572:1B08.000A: input,hidraw5: USB HID v1.11 Device [Synaptics Hi-Res Audio] on usb-0000:05:00.4-1/input3 [ 9173.386988] usb 3-1: Not enough bandwidth for new device state. [ 9173.386994] usb 3-1: Not enough bandwidth for altsetting 1 [ 9173.386996] endpoint_set_interface: 70 callbacks suppressed [ 9173.386998] usb 3-1: 1:1: usb_set_interface failed (-28) [ 9173.387110] usb 3-1: Not enough bandwidth for new device state. [ 9173.387113] usb 3-1: Not enough bandwidth for altsetting 1 (...these 3 lines repeat 9 more times, followed by...) [ 9173.388785] usb 3-1: Not enough bandwidth for new device state. [ 9173.388789] usb 3-1: Not enough bandwidth for altsetting 1 (...these 2 lines repeat 5 more times, then stop) This is a USB3.2 system, the onboard camera has a hardware switch, so I can disable it, and the only other peripheral that appears is the AX210 bluetooth (I'd thought that would have its own bus as it's part of a separate wifi chip, but apparently not, lsusb -t below), bandwidth for even a 96kHz 24bit duplex audio device shouldn't really be an issue. $ lsusb Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 003 Device 003: ID 8087:0032 Intel Corp. AX210 Bluetooth Bus 003 Device 002: ID 048d:c965 Integrated Technology Express, Inc. ITE Device(8295) Bus 003 Device 007: ID 0572:1b08 Conexant Systems (Rockwell), Inc. Hi-Res Audio Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 002: ID 0bda:0411 Realtek Semiconductor Corp. Hub Bus 002 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 001 Device 004: ID 048d:c101 Integrated Technology Express, Inc. ITE Device(8910) Bus 001 Device 002: ID 0bda:5411 Realtek Semiconductor Corp. RTS5411 Hub Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub $ lsusb -t /: Bus 04.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 10000M /: Bus 03.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 1: Dev 7, If 0, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 7, If 1, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 7, If 2, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 1: Dev 7, If 3, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 3: Dev 2, If 0, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 3: Dev 2, If 1, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 3, If 0, Class=Wireless, Driver=btusb, 12M |__ Port 4: Dev 3, If 1, Class=Wireless, Driver=btusb, 12M /: Bus 02.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 10000M |__ Port 2: Dev 2, If 0, Class=Hub, Driver=hub/4p, 5000M /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 2: Dev 2, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 4: Dev 4, If 0, Class=Human Interface Device, Driver=usbhid, 12M The only situation in which I *don't* get these dmesg errors is if I use one of the rear USB 3.2 ports that seem to lead to it being on its own bus: $ lsusb -t /: Bus 04.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 10000M /: Bus 03.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 3: Dev 2, If 0, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 3: Dev 2, If 1, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 3, If 0, Class=Wireless, Driver=btusb, 12M |__ Port 4: Dev 3, If 1, Class=Wireless, Driver=btusb, 12M /: Bus 02.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/2p, 10000M |__ Port 2: Dev 2, If 0, Class=Hub, Driver=hub/4p, 5000M /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=xhci_hcd/4p, 480M |__ Port 2: Dev 2, If 0, Class=Hub, Driver=hub/4p, 480M |__ Port 2: Dev 7, If 0, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 2: Dev 7, If 1, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 2: Dev 7, If 2, Class=Audio, Driver=snd-usb-audio, 12M |__ Port 2: Dev 7, If 3, Class=Human Interface Device, Driver=usbhid, 12M |__ Port 4: Dev 4, If 0, Class=Human Interface Device, Driver=usbhid, 12M In this case the dmesg output is clear after device connection. This seems like some kind of bandwidth calculation problem in snd_usb_audio. I've tried building a kernel and modifying various of the defines in sound/usb/card.h (currently MAX_PACKS 4 and MAX_PACKS_HS (MAX_PACKS * 4), compared to 6 and *8) but not hit on a winning formula yet. Is there any information I could collect to try to fix this? -- imalone

2 6

Internal speakers and headphone not working on Huawei notebook
by Jacek Blocki 20 Mar '24

20 Mar '24

Hello, I have Huawei notebook (KLVL-WXXW-PCB) with Ubuntu 22.04.3 LTS installed alongside Windows. In Ubuntu, neither internal speakers nor headphone jack work. I only see Dummy Output while trying to adjust volume, but sound works with external HDMI monitor. Internal speakers work with Windows, so I believe hardware is fine and kernel driver (snd_hda_intel?) needs to be modified to fix the problem. I have submitted Ubuntu bug report back in January, but nobody touched it. So I try this list hoping for better result. $ cat /proc/asound/cards 0 [Generic ]: HDA-Intel - HD-Audio Generic HD-Audio Generic at 0xd03c0000 irq 64 1 [acp ]: acp - acp HUAWEI-KLVL_WXXW-M1010-KLVL_WXXW_PCB I run 6.5.0-26-generic kernel. Regards, Jacek Błocki

1 0

[sound/soc/qcom] Question about memory allocation in asoc_qcom_lpass_cpu_platform_probe
by Zijie Zhao 20 Mar '24

20 Mar '24

Dear QCOM AUDIO maintainers, We encountered an usual usage of devm_kzalloc while performing a static analysis for kernel code. https://elixir.bootlin.com/linux/v6.8/source/sound/soc/qcom/lpass-cpu.c#L12… ``` int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) { ... drvdata = devm_kzalloc(dev, sizeof(struct lpass_data), GFP_KERNEL); if (!drvdata) return -ENOMEM; ... /* Allocation for i2sctl regmap fields */ drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl), GFP_KERNEL); /* Initialize bitfields for dai I2SCTL register */ ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl, drvdata->lpaif_map); if (ret) { dev_err(dev, "error init i2sctl field: %d\n", ret); return ret; } ... } ``` ``` static int lpass_cpu_init_i2sctl_bitfields(struct device *dev, struct lpaif_i2sctl *i2sctl, struct regmap *map) { struct lpass_data *drvdata = dev_get_drvdata(dev); const struct lpass_variant *v = drvdata->variant; i2sctl->loopback = devm_regmap_field_alloc(dev, map, v->loopback); ... } ``` Here devm_kzalloc might return NULL for drvdata->i2sctl if out-of-memory. Then i2sctl->loopback will cause segfault since i2sctl is never checked against NULL. Would a NULL check needed here? Like the drvdata above. Please let us know if we missed any key information or assumption! We appreciate your time! Best, Zijie

1 0

[PATCH] ASoC: SOF: Remove libraries from topology lookups
by cujomalainey＠chromium.org 20 Mar '24

20 Mar '24

From: Curtis Malainey <cujomalainey(a)chromium.org> Default firmware shipped in open source are not licensed for 3P libraries, therefore topologies should not reference them. If a OS wants to use 3P (that they have licensed) then they should use the appropriate topology override mechanisms. Fixes: 8a7d5d85ed2161 ("ASoC: SOF: mediatek: mt8195: Add devicetree support to select topologies") Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> Cc: Wojciech Macek <wmacek(a)google.com> --- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 8ee7ee246344c..f143baf75af60 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -573,7 +573,7 @@ static struct snd_sof_dsp_ops sof_mt8195_ops = { static struct snd_sof_of_mach sof_mt8195_machs[] = { { .compatible = "google,tomato", - .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682-dts.tplg" + .sof_tplg_filename = "sof-mt8195-mt6359-rt1019-rt5682.tplg" }, { .compatible = "mediatek,mt8195", .sof_tplg_filename = "sof-mt8195.tplg" -- 2.44.0.291.gc1ea87d7ee-goog

4 6

[PATCH 1/2] ASoC: codecs: ES8326: Reducing power consumption
by Zhang Yi 20 Mar '24

20 Mar '24

> > For lower power consumption during hibernation, the configuration of > > es8326_suspend and es8326_remove will be adjusted. > > Adding es8326_i2c_shutdown and es8326_i2c_remove to cover different > > situations > > > > Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> > > --- > > sound/soc/codecs/es8326.c | 34 +++++++++++++++++++++++++++++++++- > > 1 file changed, 33 insertions(+), 1 deletion(-) > > > > diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c > > index 15289dadafea..5c89323035a2 100755 > > --- a/sound/soc/codecs/es8326.c > > +++ b/sound/soc/codecs/es8326.c > > @@ -1072,12 +1072,13 @@ static int es8326_suspend(struct snd_soc_component *component) > > es8326->calibrated = false; > > regmap_write(es8326->regmap, ES8326_CLK_CTL, ES8326_CLK_OFF); > > regcache_cache_only(es8326->regmap, true); > > - regcache_mark_dirty(es8326->regmap); > > > > /* reset register value to default */ > > regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); > > usleep_range(1000, 3000); > > regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); > > + > > + regcache_mark_dirty(es8326->regmap); > > return 0; > > } > > > > @@ -1163,8 +1164,13 @@ static int es8326_set_jack(struct snd_soc_component *component, > > > > static void es8326_remove(struct snd_soc_component *component) > > { > > + struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); > > + > > es8326_disable_jack_detect(component); > > es8326_set_bias_level(component, SND_SOC_BIAS_OFF); > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); > > + usleep_range(1000, 3000); > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); > > } > > > > static const struct snd_soc_component_driver soc_component_dev_es8326 = { > > @@ -1236,6 +1242,30 @@ static int es8326_i2c_probe(struct i2c_client *i2c) > > &es8326_dai, 1); > > } > > > > + > > +static void es8326_i2c_shutdown(struct i2c_client *i2c) > > +{ > > + struct snd_soc_component *component; > > + struct es8326_priv *es8326; > > + > > + dev_dbg(component->dev, "Enter into %s\n", __func__); > Above line works on unset pointer, you need to set component before > dereferencing it. > > + es8326 = i2c_get_clientdata(i2c); > > + component = es8326->component; > > + cancel_delayed_work_sync(&es8326->jack_detect_work); > > + cancel_delayed_work_sync(&es8326->button_press_work); > > + > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x01); > > + usleep_range(1000, 3000); > > + regmap_write(es8326->regmap, ES8326_CSM_I2C_STA, 0x00); > > + > > + return; > > +} > > + > > +static void es8326_i2c_remove(struct i2c_client *i2c) > > +{ > > + es8326_i2c_shutdown(i2c); > > +} > > + > > static const struct i2c_device_id es8326_i2c_id[] = { > > {"es8326", 0 }, > > {} > > @@ -1265,6 +1295,8 @@ static struct i2c_driver es8326_i2c_driver = { > > .of_match_table = of_match_ptr(es8326_of_match), > > }, > > .probe = es8326_i2c_probe, > > + .shutdown = es8326_i2c_shutdown, > > + .remove = es8326_i2c_remove, > > .id_table = es8326_i2c_id, > > }; > > module_i2c_driver(es8326_i2c_driver); I'm sorry. I'll fix my mistake.

1 0

[PATCH 0/2] ASoC: codecs: ES8326: Reducin powerconsumption and
by Zhang Yi 20 Mar '24

20 Mar '24

We changed the configuration related to hibernation. and delete the REG_SUPPLY to cover mute issue. Zhang Yi (2): ASoC: codecs: ES8326: Reducing power consumption ASoC: codecs: ES8326: Delete unused REG_SUPPLY sound/soc/codecs/es8326.c | 42 ++++++++++++++++++++++++++++++--------- 1 file changed, 33 insertions(+), 9 deletions(-) -- 2.17.1

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[PATCH] ALSA: hda/realtek: Add quirks for some Clevo laptops
by Tim Crawford 20 Mar '24

20 Mar '24

Add audio quirks to fix speaker output and headset detection on some new Clevo models: - L240TU (ALC245) - PE60SNE-G (ALC1220) - V350SNEQ (ALC245) Co-authored-by: Jeremy Soller <jeremy(a)system76.com> Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d463d416fc23..25183996756c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2645,6 +2645,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x65f1, "Clevo PC50HS", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x65f5, "Clevo PD50PN[NRT]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x66a2, "Clevo PE60RNE", ALC1220_FIXUP_CLEVO_PB51ED_PINS), + SND_PCI_QUIRK(0x1558, 0x66a6, "Clevo PE60SN[CDE]-[GS]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67d1, "Clevo PB71[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS), SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS), @@ -10139,12 +10140,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x152d, 0x1082, "Quanta NL3", ALC269_FIXUP_LIFEBOOK), + SND_PCI_QUIRK(0x1558, 0x0353, "Clevo V35[05]SN[CDE]Q", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1323, "Clevo N130ZU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1325, "Clevo N15[01][CW]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1401, "Clevo L140[CZ]U", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1403, "Clevo N140CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x1404, "Clevo N150CU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x14a1, "Clevo L141MU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x2624, "Clevo L240TU", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4018, "Clevo NV40M[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4019, "Clevo NV40MZ", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x4020, "Clevo NV40MB", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.44.0

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Re: [PATCH] Add Asus VivoBook S 16X to quirks table for acp6x
by Geraldo Nascimento 20 Mar '24

20 Mar '24

On Tue, Mar 19, 2024 at 05:16:10PM +0800, marcel wrote: > Hi again, > > it's been a while since i sent my patch and I still can't see it in > kernel on github, so there's my question because I'm not sure if I > understand correctly. Should I just wait a little bit more and it just > takes time or send patch again and change things you pointed? Also which > version should I type after [PATCH v...], "1" ? Because it's my first > attempt? Sorry for taking your time and if the questions are obvious, > but it's my first time doing something like this. Hi Marcel, please address my past concerns regarding your patch and resubmit as v2. Remember to Cc: Takashi Iwai and Jaroslav Kysela besides linux-sound mailing list or your patch will fall into limbo again, overlooked by the maintainers. There's also a get_maintainers.pl script in the linux tree. You might wanna use that as it adds some relevant people who might take a interest in your patch. Thanks, Geraldo Nascimento > > Greetings

1 0

[PATCH v14 00/16] Add audio support in v4l2 framework
by Shengjiu Wang 19 Mar '24

19 Mar '24

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v14: - document the reservation of 'AUXX' fourcc format. - add v4l2_audfmt_to_fourcc() definition. changes in v13 - change 'pixelformat' to 'audioformat' in dev-audio-mem2mem.rst - add more description for clock drift in ext-ctrls-audio-m2m.rst - Add "media: v4l2-ctrls: add support for fraction_bits" from Hans to avoid build issue for kernel test robot changes in v12 - minor changes according to comments - drop min_buffers_needed = 1 and V4L2_CTRL_FLAG_UPDATE flag - drop bus_info changes in v11 - add add-fixed-point-test-controls in vivid. - add v4l2_ctrl_fp_compose() helper function for min and max changes in v10 - remove FIXED_POINT type - change code base on media: v4l2-ctrls: add support for fraction_bits - fix issue reported by kernel test robot - remove module_alias changes in v9: - add MEDIA_ENT_F_PROC_AUDIO_RESAMPLER. - add MEDIA_INTF_T_V4L_AUDIO - add media controller support - refine the vim2m-audio to support 8k<->16k conversion. changes in v8: - refine V4L2_CAP_AUDIO_M2M to be 0x00000008 - update doc for FIXED_POINT - address comments for imx-asrc changes in v7: - add acked-by from Mark - separate commit for fixed point, m2m audio class, audio rate controls - use INTEGER_MENU for rate, FIXED_POINT for rate offset - remove used fmts - address other comments for Hans changes in v6: - use m2m_prepare/m2m_unprepare/m2m_start/m2m_stop to replace m2m_start_part_one/m2m_stop_part_one, m2m_start_part_two/m2m_stop_part_two. - change V4L2_CTRL_TYPE_ASRC_RATE to V4L2_CTRL_TYPE_FIXED_POINT - fix warning by kernel test rebot - remove some unused format V4L2_AUDIO_FMT_XX - Get SNDRV_PCM_FORMAT from V4L2_AUDIO_FMT in driver. - rename audm2m to viaudm2m. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Hans Verkuil (1): media: v4l2-ctrls: add support for fraction_bits Shengjiu Wang (15): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Define audio sample format fourcc type media: uapi: Add V4L2_CTRL_CLASS_M2M_AUDIO media: uapi: Add audio rate controls support media: uapi: Declare interface types for Audio media: uapi: Add an entity type for audio resampler media: vivid: add fixed point test controls media: imx-asrc: Add memory to memory driver media: vim2m-audio: add virtual driver for audio memory to memory .../media/mediactl/media-types.rst | 11 + .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-audio-m2m.rst | 59 + .../userspace-api/media/v4l/pixfmt-audio.rst | 100 ++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/v4l/vidioc-queryctrl.rst | 11 +- .../media/videodev2.h.rst.exceptions | 3 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 13 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1256 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 10 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 793 +++++++++++ drivers/media/test-drivers/vivid/vivid-core.h | 2 + .../media/test-drivers/vivid/vivid-ctrls.c | 26 + drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 9 + drivers/media/v4l2-core/v4l2-ctrls-api.c | 1 + drivers/media/v4l2-core/v4l2-ctrls-core.c | 93 +- drivers/media/v4l2-core/v4l2-ctrls-defs.c | 10 + drivers/media/v4l2-core/v4l2-dev.c | 21 + drivers/media/v4l2-core/v4l2-ioctl.c | 66 + drivers/media/v4l2-core/v4l2-mem2mem.c | 13 +- include/media/v4l2-ctrls.h | 13 +- include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 60 + include/uapi/linux/media.h | 2 + include/uapi/linux/v4l2-controls.h | 9 + include/uapi/linux/videodev2.h | 50 +- sound/soc/fsl/fsl_asrc.c | 144 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 233 +++ sound/soc/fsl/fsl_easrc.h | 6 +- 42 files changed, 3128 insertions(+), 27 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-audio-m2m.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/vim2m-audio.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

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[PATCH v1 1/1] ASoC: fsl: imx-es8328: Switch to using gpiod API
by Andy Shevchenko 18 Mar '24

18 Mar '24

This updates the driver to gpiod API, and removes yet another use of of_get_named_gpio(). Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/fsl/imx-es8328.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/imx-es8328.c b/sound/soc/fsl/imx-es8328.c index 6f0d031c1d5f..ad915c55cc35 100644 --- a/sound/soc/fsl/imx-es8328.c +++ b/sound/soc/fsl/imx-es8328.c @@ -3,7 +3,7 @@ // Copyright 2012 Freescale Semiconductor, Inc. // Copyright 2012 Linaro Ltd. -#include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/of.h> #include <linux/of_platform.h> @@ -23,7 +23,7 @@ struct imx_es8328_data { struct snd_soc_card card; char codec_dai_name[DAI_NAME_SIZE]; char platform_name[DAI_NAME_SIZE]; - int jack_gpio; + struct gpio_desc *jack_gpiod; }; static struct snd_soc_jack_gpio headset_jack_gpios[] = { @@ -54,8 +54,8 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) struct imx_es8328_data, card); int ret = 0; - /* Headphone jack detection */ - if (gpio_is_valid(data->jack_gpio)) { + if (data->jack_gpiod) { + /* Headphone jack detection */ ret = snd_soc_card_jack_new_pins(rtd->card, "Headphone", SND_JACK_HEADSET | SND_JACK_BTN_0, &headset_jack, @@ -64,7 +64,7 @@ static int imx_es8328_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret) return ret; - headset_jack_gpios[0].gpio = data->jack_gpio; + headset_jack_gpios[0].desc = data->jack_gpiod; ret = snd_soc_jack_add_gpios(&headset_jack, ARRAY_SIZE(headset_jack_gpios), headset_jack_gpios); @@ -174,7 +174,11 @@ static int imx_es8328_probe(struct platform_device *pdev) data->dev = dev; - data->jack_gpio = of_get_named_gpio(pdev->dev.of_node, "jack-gpio", 0); + data->jack_gpiod = devm_gpiod_get_optional(dev, "jack"); + if (IS_ERR(data->jack_gpiod)) { + ret = PTR_ERR(data->jack_gpiod); + goto put_device; + } /* * CPU == Platform -- 2.43.0.rc1.1.gbec44491f096

1 1

[PATCH V0 0/1] ASoC: soc-compress: Fix and add missing DPCM locking
by Shalini Manjunatha 18 Mar '24

18 Mar '24

We find mising DPCM locking inside soc_compr_set_params_fe() before calling dpcm_be_dai_hw_params() and dpcm_be_dai_prepare() which cause lockdep assert for DPCM lock not held in __soc_pcm_hw_params() and __soc_pcm_prepare() in above flow when ever we play compress offload audio playback use case. To fix this issue added DPCM lock and unlock in both places of above code flow mentioned. Shalini Manjunatha (1): ASoC: soc-compress: Fix and add DPCM locking sound/soc/soc-compress.c | 4 ++++ 1 file changed, 4 insertions(+) -- 2.7.4

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[PATCH] ALSA: hda/realtek: fix the hp playback volume issue for LG machines
by Hui Wang 18 Mar '24

18 Mar '24

Recently we tested the headphone playback on 2 LG machines, if we set the volume to the max value or near to the max value, the sound is too loud, it could even bring harm to listeners. A workaround is to decrease the max volume to a reasonable value for the headphone's amplifier, then the users couldn't set the volume bigger than that value from the userspace. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e904f62e1952..d463d416fc23 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6964,6 +6964,25 @@ static void alc256_fixup_mic_no_presence_and_resume(struct hda_codec *codec, } } +static void alc256_decrease_headphone_amp_val(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + u32 caps; + u8 nsteps, offs; + + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + caps = query_amp_caps(codec, 0x3, HDA_OUTPUT); + nsteps = ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) - 10; + offs = ((caps & AC_AMPCAP_OFFSET) >> AC_AMPCAP_OFFSET_SHIFT) - 10; + caps &= ~AC_AMPCAP_NUM_STEPS & ~AC_AMPCAP_OFFSET; + caps |= (nsteps << AC_AMPCAP_NUM_STEPS_SHIFT) | (offs << AC_AMPCAP_OFFSET_SHIFT); + + if (snd_hda_override_amp_caps(codec, 0x3, HDA_OUTPUT, caps)) + codec_warn(codec, "failed to override amp caps for NID 0x3\n"); +} + static void alc_fixup_dell4_mic_no_presence_quiet(struct hda_codec *codec, const struct hda_fixup *fix, int action) @@ -7382,6 +7401,7 @@ enum { ALC294_FIXUP_CS35L41_I2C_2, ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED, ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, + ALC256_FIXUP_HEADPHONE_AMP_VOL, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9581,6 +9601,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc256_fixup_acer_sfg16_micmute_led, }, + [ALC256_FIXUP_HEADPHONE_AMP_VOL] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_decrease_headphone_amp_val, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10319,6 +10343,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x9e56, "Lenovo ZhaoYang CF4620Z", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1849, 0x1233, "ASRock NUC Box 1100", ALC233_FIXUP_NO_AUDIO_JACK), SND_PCI_QUIRK(0x1849, 0xa233, "Positivo Master C6300", ALC269_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1854, 0x0440, "LG CQ6", ALC256_FIXUP_HEADPHONE_AMP_VOL), + SND_PCI_QUIRK(0x1854, 0x0441, "LG CQ6 AIO", ALC256_FIXUP_HEADPHONE_AMP_VOL), SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x19e5, 0x320f, "Huawei WRT-WX9 ", ALC256_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1b35, 0x1235, "CZC B20", ALC269_FIXUP_CZC_B20), -- 2.34.1

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[PATCH] ALSA: hda: cs35l41: Support HP Spectre x360 14 eu0000
by Anthony I Gilea 18 Mar '24

18 Mar '24

The new HP Spectre x360 has _DSD for CS35L41 amps in ACPI but reset-gpios and spk-id-gpios are merged into single Package of size 4 so _DSD parser fails to parse it. Overwrite broken _DSD with the correct configuration. Signed-off-by: Anthony I Gilea <i(a)cpp.in> --- sound/pci/hda/cs35l41_hda_property.c | 2 ++ 1 file changed, 2 insertions(+) diff --ruNp a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c --- a/sound/pci/hda/cs35l41_hda_property.c 2024-03-10 23:38:09.000000000 +0300 +++ b/sound/pci/hda/cs35l41_hda_property.c 2024-03-14 18:22:01.887566917 +0300 @@ -64,6 +64,7 @@ static const struct cs35l41_config cs35l { "103C8BE5", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8BE6", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "103C8B3A", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, + { "103C8C15", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4100, 24 }, { "104312AF", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 1, 2, 0, 1000, 4500, 24 }, { "10431433", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, { "10431463", 2, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 1000, 4500, 24 }, @@ -405,6 +406,7 @@ static const struct cs35l41_prop_model c { "CSC3551", "103C8BE5", generic_dsd_config }, { "CSC3551", "103C8BE6", generic_dsd_config }, { "CSC3551", "103C8B3A", generic_dsd_config }, + { "CSC3551", "103C8C15", generic_dsd_config }, { "CSC3551", "104312AF", generic_dsd_config }, { "CSC3551", "10431433", generic_dsd_config }, { "CSC3551", "10431463", generic_dsd_config },

3 2

[PATCH v11] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 18 Mar '24

18 Mar '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of loudspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v11: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log - Add code to check the size of calibrated data in UEFI - Add description of Layout of calibrated Data in UEFI - switch back link_id and unique_id to identify the devcie - unique_id offset as the device index of the calibrated data - update calibrated data layout stored in UEFI, add link_id and unique_id into the calibrated data in case of tas2783 sitting in different SDQW buses. - correct the wrong unique_id - set the unique_id as hex style in dsp bin file name --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 932 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 106 ++++ 4 files changed, 1056 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 59f9742e9ff4..6ba3fd0c0811 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -247,6 +247,7 @@ 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 @@ -1847,6 +1848,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f53baa2b9565..6afceb126ae3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -284,6 +284,7 @@ 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 @@ -664,6 +665,7 @@ 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..71c218792bc9 --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,932 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static const struct reg_default tas2783_reg_defaults[] = { + /* Default values for ROM mode. Activated. */ + { 0x8002, 0x1a }, /* AMP inactive. */ + { 0x8097, 0xc8 }, + { 0x80b5, 0x74 }, + { 0x8099, 0x20 }, + { 0xfe8d, 0x0d }, + { 0xfebe, 0x4a }, + { 0x8230, 0x00 }, + { 0x8231, 0x00 }, + { 0x8232, 0x00 }, + { 0x8233, 0x01 }, + { 0x8418, 0x00 }, /* Set volume to 0 dB. */ + { 0x8419, 0x00 }, + { 0x841a, 0x00 }, + { 0x841b, 0x00 }, + { 0x8428, 0x40 }, /* Unmute channel */ + { 0x8429, 0x00 }, + { 0x842a, 0x00 }, + { 0x842b, 0x00 }, + { 0x8548, 0x00 }, /* Set volume to 0 dB. */ + { 0x8549, 0x00 }, + { 0x854a, 0x00 }, + { 0x854b, 0x00 }, + { 0x8558, 0x40 }, /* Unmute channel */ + { 0x8559, 0x00 }, + { 0x855a, 0x00 }, + { 0x855b, 0x00 }, + { 0x800a, 0x3a }, /* Enable both channel */ + { 0x800e, 0x44 }, + { 0x800f, 0x40 }, + { 0x805c, 0x99 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0), 0 }, + { SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0 }, +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0x807F: /* Page 0 */ + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + 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 = 0x44ffffff, + .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 const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +static void tas2783_apply_calibv1(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 offset; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + offset = tas_dev->sdw_peripheral->id.unique_id - + TAS2783_UNIQUE_ID_BASE; + if (offset >= TAS2783_MAX_DEV_NUM) { + dev_err(tas_dev->dev, "%s: offset(%u) is out of range.\n", + __func__, offset); + return; + } + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)(cali_data + offset * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "Cali failed %x:%d\n", + tas2783_cali_reg[i], ret); + break; + } + } +} + +static void tas2783_apply_calibv2(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + const unsigned int arr_size = ARRAY_SIZE(tas2783_cali_reg); + struct regmap *map = tas_dev->regmap; + unsigned int dev_sum = cali_data[1], i, j, k; + u8 *cali_start; + u16 dev_info; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + dev_info = tas_dev->sdw_peripheral->bus->link_id | + tas_dev->sdw_peripheral->id.unique_id << 16; + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)&cali_data[3]; + for (i = 0; i < dev_sum; i++) { + k = i * (arr_size + 1) + 3; + if (dev_info != cali_data[k]) { + for (j = 0; j < arr_size; j++) { + k = 4 * (k + 1 + j); + ret = regmap_bulk_write(map, + tas2783_cali_reg[j], + &cali_start[k], 4); + if (ret) { + dev_err(tas_dev->dev, + "Cali failed %x:%d\n", + tas2783_cali_reg[j], ret); + break; + } + } + break; + } + } +} + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data in UEFI(total 256 bytes): + * V2: + * ChipID (2783, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + * Or + * V1: + * Calibrated Data of Dev 0(unique_id offset 0) (20 bytes) + * Calibrated Data of Dev 1(unique_id offset 1) (20 bytes) + * Calibrated Data of Dev 2(unique_id offset 2) (20 bytes) + * Calibrated Data of Dev 3(unique_id offset 3) (20 bytes) + * Calibrated Data of Dev 4(unique_id offset 4) (20 bytes) + * Calibrated Data of Dev 5(unique_id offset 5) (20 bytes) + * Calibrated Data of Dev 6(unique_id offset 6) (20 bytes) + * Calibrated Data of Dev 7(unique_id offset 7) (20 bytes) + * TimeStamp of Calibration (4 bytes) + * CRC (4 bytes) + * Reserved (88 bytes) + */ +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + 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 calibration_data cali_data; + unsigned int *tmp_val; + unsigned int crc; + efi_status_t status; + + cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, NULL); + if (status == EFI_BUFFER_TOO_SMALL + && cali_data.total_sz < TAS2783_MAX_CALIDATA_SIZE) { + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, + cali_data.data); + dev_dbg(tas_dev->dev, "%s: cali get %lx bytes result:%ld\n", + __func__, cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)cali_data.data; + + if (tmp_val[0] == 2783) { + /* Calibrated Data V2 */ + unsigned int dev_sum = tmp_val[1]; + + if (dev_sum > TAS2783_MAX_DEV_NUM || + dev_sum == 0) { + dev_dbg(tas_dev->dev, "No dev in calibrated data V2."); + return; + } + crc = crc32(~0, cali_data.data, 12 + dev_sum * 24) ^ ~0; + if (crc == tmp_val[3 + dev_sum * 6]) { + tas2783_apply_calibv2(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V2: %ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, + "V2: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } + } else { + /* Calibrated Data V1 */ + /* 8 devs * 20 bytes calibrated data/dev + 4 bytes Timestamp */ + crc = crc32(~0, cali_data.data, 164) ^ ~0; + if (crc == tmp_val[41]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calibv1(tas_dev, tmp_val); + dev_dbg(tas_dev->dev, "V1: %ptTs", &tmp_val[40]); + } else { + dev_dbg(tas_dev->dev, + "V1: CRC 0x%08x not match 0x%08x\n", + crc, tmp_val[41]); + } + } +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = + (struct tasdevice_priv *) context; + struct tas2783_firmware_node *p; + struct regmap *map = tas_dev->regmap; + unsigned char *buf = NULL; + int offset = 0, img_sz; + int ret; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read %s: work in bypass-dsp mode.\n", + __func__, tas_dev->dspfw_binaryname); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) { + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + sizeof(unsigned int) * 5, p->length); + } else { + ret = regmap_write(map, p->download_addr, + *(buf + offset + sizeof(unsigned int) * 5)); + } + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += sizeof(unsigned int) * 5 + p->length; + } + + tas2783_calibration(tas_dev); +} + +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("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_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 tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +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 *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + 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; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + 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 *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, 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 *tas_dev = + snd_soc_component_get_drvdata(component); + struct regmap *map = tas_dev->regmap; + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (mute) { + if (direction == SNDRV_PCM_STREAM_CAPTURE) { + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, + "%s: Disable AEF failed.\n", __func__); + } else { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 mute failed.\n", __func__); + goto out; + } + /* + * Both playback and echo data will be shutdown in + * playback stream. + */ + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR&AEF shutdown failed.\n", + __func__); + goto out; + } + tas_dev->pstream = false; + } + } else { + /* FU23 Unmute, 0x40400108. */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + goto out; + } + tas_dev->pstream = true; + } else { + /* Capture stream is the echo ref data for voice. + * Without playback, it can't be active. + */ + if (tas_dev->pstream == true) { + ret = regmap_update_bits(map, + TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, + TAS2783_REG_AEF_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: AEF enable failed.\n", + __func__); + goto out; + } + } else { + dev_err(tas_dev->dev, + "%s: No playback, no AEF!", __func__); + ret = -EINVAL; + } + } + } +out: + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .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 = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + tas_dev->component = comp; + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, "%s-tas2783.bin", + comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsys_id-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-0x%x.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } else { + /* + * tas2783-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(tas_dev->dspfw_binaryname, + TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-0x%x.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } + } + + ret = request_firmware(&fw_entry, tas_dev->dspfw_binaryname, + tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .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, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + 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:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i, j; + u32 bit; + + 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 = 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 = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children. + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + /* Update the status */ + tas_priv->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || tas_priv->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 int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->chip_id = id->driver_data; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .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_AUTHOR("Shenghao Ding <shenghao-ding(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..631776dcec4c --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,106 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +/* Unique id start */ +#define TAS2783_UNIQUE_ID_BASE 0x08 +#define TAS2783_MAX_DEV_NUM 8 + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_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, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* 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, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 256 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct snd_soc_component *component; + struct sdw_slave *sdw_peripheral; + enum sdw_slave_status status; + struct sdw_bus_params params; + struct regmap *regmap; + struct device *dev; + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + unsigned char dev_name[32]; + unsigned int chip_id; + bool pstream; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

4 6

[PATCH] ALSA: core: add kunitconfig
by Takashi Sakamoto 17 Mar '24

17 Mar '24

It is helpful to add .kunitconfig if we work with the tools provided by KUnit project. The file describes the series of kernel configurations to satisfy the dependency to build the target test. For example: $ ./tools/testing/kunit/kunit.py run --arch=arm64 --cross_compile=aarch64-linux-gnu- --kunitconfig=sound/core/ [11:35:13] Configuring KUnit Kernel ... Regenerating .config ... Populating config with: $ make ARCH=arm64 O=.kunit olddefconfig CROSS_COMPILE=aarch64-linux-gnu- [11:35:19] Building KUnit Kernel ... Populating config with: $ make ARCH=arm64 O=.kunit olddefconfig CROSS_COMPILE=aarch64-linux-gnu- Building with: $ make ARCH=arm64 O=.kunit --jobs=8 CROSS_COMPILE=aarch64-linux-gnu- [11:37:35] Starting KUnit Kernel (1/1)... [11:37:35] ============================================================ Running tests with: $ qemu-system-aarch64 -nodefaults -m 1024 -kernel .kunit/arch/arm64/boot/Image.gz -append 'kunit.enable=1 console=ttyAMA0 kunit_shutdown=reboot' -no-reboot -nographic -serial stdio -machine virt -cpu max,pauth-impdef=on [11:37:35] ============== sound-core-test (10 subtests) =============== [11:37:35] [PASSED] test_phys_format_size [11:37:35] [PASSED] test_format_width [11:37:35] [PASSED] test_format_endianness [11:37:35] [PASSED] test_format_signed [11:37:35] [PASSED] test_format_fill_silence [11:37:35] [PASSED] test_playback_avail [11:37:35] [PASSED] test_capture_avail [11:37:35] [PASSED] test_card_set_id [11:37:35] [PASSED] test_pcm_format_name [11:37:35] [PASSED] test_card_add_component [11:37:35] ================= [PASSED] sound-core-test ================= [11:37:35] ============================================================ [11:37:35] Testing complete. Ran 10 tests: passed: 10 [11:37:35] Elapsed time: 142.333s total, 5.617s configuring, 136.047s building, 0.630s running Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/.kunitconfig | 5 +++++ 1 file changed, 5 insertions(+) create mode 100644 sound/core/.kunitconfig diff --git a/sound/core/.kunitconfig b/sound/core/.kunitconfig new file mode 100644 index 000000000000..440f974ba0b7 --- /dev/null +++ b/sound/core/.kunitconfig @@ -0,0 +1,5 @@ +CONFIG_KUNIT=y +CONFIG_SOUND=y +CONFIG_SND=y +CONFIG_SND_PCM=y +CONFIG_SND_CORE_TEST=y -- 2.43.0

2 1

[PATCH] ALSA: hda/realtek: add in quirk for Acer Swift Go 16 - SFG16-71
by Ian Murphy 17 Mar '24

17 Mar '24

Keyboard has an LED that is ON/OFF when mic is muted/active - LED is controlled by GPIO pin - Patch enables led to appear in /sys/class/leds/ as hda::micmute - Enables LED when mic is MUTED - Disables LED when mic is active Signed-off-by: Ian Murphy <iano200(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a91d5ec29bfb..05c685485606 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6891,6 +6891,13 @@ static void yoga7_14arb7_fixup_i2c(struct hda_codec *cdc, comp_generic_fixup(cdc, action, "i2c", "INT8866", "-%s:00", 1); } +static void alc256_fixup_acer_sfg16_micmute_led(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + alc_fixup_hp_gpio_led(codec, action, 0, 0x04); +} + + /* for alc295_fixup_hp_top_speakers */ #include "hp_x360_helper.c" @@ -7374,6 +7381,7 @@ enum { ALC289_FIXUP_DELL_CS35L41_SPI_2, ALC294_FIXUP_CS35L41_I2C_2, ALC245_FIXUP_CS35L56_SPI_4_HP_GPIO_LED, + ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9569,6 +9577,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_GPIO_LED, }, + [ALC256_FIXUP_ACER_SFG16_MICMUTE_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc256_fixup_acer_sfg16_micmute_led, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9612,6 +9624,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x1430, "Acer TravelMate B311R-31", ALC256_FIXUP_ACER_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1025, 0x1466, "Acer Aspire A515-56", ALC255_FIXUP_ACER_HEADPHONE_AND_MIC), SND_PCI_QUIRK(0x1025, 0x1534, "Acer Predator PH315-54", ALC255_FIXUP_ACER_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1025, 0x169a, "Acer Swift SFG16", ALC256_FIXUP_ACER_SFG16_MICMUTE_LED), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x053c, "Dell Latitude E5430", ALC292_FIXUP_DELL_E7X), SND_PCI_QUIRK(0x1028, 0x054b, "Dell XPS one 2710", ALC275_FIXUP_DELL_XPS), -- 2.44.0

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chore: Use actions/checkout@v4
by GitHub pull_request - opened 16 Mar '24

16 Mar '24

alsa-project/alsa-utils pull request #259 was opened from Korilakkuma: Refer to https://github.blog/changelog/2023-09-22-github-actions-transitioning-from-…. > Node.js 16 actions are deprecated. Please update the following actions to use Node.js 20: actions/checkout@v3. Request URL : https://github.com/alsa-project/alsa-utils/pull/259 Patch URL : https://github.com/alsa-project/alsa-utils/pull/259.patch Repository URL: https://github.com/alsa-project/alsa-utils

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GitHub Actions: Use actions/checkout@v4
by GitHub pull_request - opened 16 Mar '24

16 Mar '24

alsa-project/alsa-lib pull request #390 was opened from ryandesign: Fixes: #389 Request URL : https://github.com/alsa-project/alsa-lib/pull/390 Patch URL : https://github.com/alsa-project/alsa-lib/pull/390.patch Repository URL: https://github.com/alsa-project/alsa-lib

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Linux 6.9 pre-rc1 might have broken UCM matching for USB devices
by GitHub issues - edited 15 Mar '24

15 Mar '24

alsa-project/alsa-ucm-conf issue #404 was edited from pinkflames: While testing Torvalds' master branch of Linux which will become 6.9-rc1 eventually, I have stumbled upon PipeWire/WirePlumber apparently no longer applying the UCM profile for my USB connected on-board audio controller. Git bisection leads to the following commit: torvalds/linux@1601cd53c7e3197181277326dbfc131d20a74e46 . It's possible that I'm wrong in concluding that the issue is UCM no longer applying/matching but that's my best guess looking at seemingly ACP named devices and the lack of any `ucm` strings in the output of pw-dump, when the regression is present. Since this is likely a kernel regression, I'll be reporting that to the [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218605), too, but I thought alsa-ucm-conf might be interested in this change as well. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/404 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Linux 6.9 pre-rc1 might have broken UCM matching
by GitHub issues - edited 15 Mar '24

15 Mar '24

alsa-project/alsa-ucm-conf issue #404 was edited from pinkflames: While testing Torvalds' master branch of Linux which will become 6.9-rc1 eventually, I have stumbled upon PipeWire/WirePlumber apparently no longer applying the UCM profile for my USB connected on-board audio controller. Git bisection leads to the following commit: torvalds/linux@1601cd53c7e3197181277326dbfc131d20a74e46 . It's possible that I'm wrong in concluding that the issue is UCM no longer applying/matching but that's my best guess looking at seemingly ACP named devices and the lack of any `ucm` strings in the output of pw-dump, when the regression is present. Since this is likely a kernel regression, I'll be reporting that to the [kernel bugzilla](https://bugzilla.kernel.org/show_bug.cgi?id=218605), too, but I thought alsa-ucm-conf might be interested in this change as well. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/404 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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