January 2025 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: codecs: ES8326: Improved PSRR
by Zhang Yi 20 Jan '25

20 Jan '25

Modified configuration to improve PSSR when ES8326 is working Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b06eead7e0f6..066d92b54312 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -911,7 +911,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x1f); - regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x08); + regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x0d); queue_delayed_work(system_wq, &es8326->jack_detect_work, msecs_to_jiffies(400)); es8326->hp = 1; @@ -1023,7 +1023,7 @@ static void es8326_init(struct snd_soc_component *component) struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); regmap_write(es8326->regmap, ES8326_RESET, 0x1f); - regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x3E); regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0); usleep_range(10000, 15000); regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9); -- 2.17.1

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[GIT PULL] ASoC updates for v6.14
by Mark Brown 20 Jan '25

20 Jan '25

The following changes since commit 9d89551994a430b50c4fffcb1e617a057fa76e20: Linux 6.13-rc6 (2025-01-05 14:13:40 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.14 for you to fetch changes up to fee89ddd76e45841a2b01d87b481bc02483f4572: ASoC: xilinx: xlnx_spdif: Simpify using devm_clk_get_enabled() (2025-01-16 15:20:41 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.14 This was quite a quiet release for what I imagine are holiday related reasons, the diffstat is dominated by some Cirrus Logic Kunit tests. There's the usual mix of small improvements and fixes, plus a few new drivers and features. The diffstat includes some DRM changes due to work on HDMI audio. - Allow clocking on each DAI in an audio graph card to be configured separately. - Improved power management for Renesas RZ-SSI. - KUnit testing for the Cirrus DSP framework. - Memory to meory operation support for Freescale/NXP platforms. - Support for pause operations in SOF. - Support for Allwinner suinv F1C100s, Awinc AW88083, Realtek ALC5682I-VE ---------------------------------------------------------------- Alexey Klimov (1): ASoC: qcom: sdm845: add handling of secondary MI2S clock Amadeusz Sławiński (4): ASoC: Intel: avs: Fix init-config parsing ASoC: Intel: avs: Print IPC error messages in lower layer ASoC: Intel: avs: Add MODULE_FIRMWARE to inform about FW ASoC: Intel: avs: Update ASRC definition Andrew Davis (21): ASoC: ad193x: Remove use of i2c_match_id() ASoC: adau1761: Remove use of i2c_match_id() ASoC: adau1781: Remove use of i2c_match_id() ASoC: adau1977: Remove use of i2c_match_id() ASoC: alc5623: Remove use of i2c_match_id() ASoC: alc5632: Remove use of i2c_match_id() ASoC: max98088: Remove use of i2c_match_id() ASoC: max98090: Remove use of i2c_match_id() ASoC: max98095: Remove use of i2c_match_id() ASoC: pcm186x: Remove use of i2c_match_id() ASoc: pcm6240: Remove use of i2c_match_id() ASoC: ssm2602: Remove use of i2c_match_id() ASoC: tas2562: Remove use of i2c_match_id() ASoC: tas2781: Remove use of i2c_match_id() ASoC: tas5720: Remove use of i2c_match_id() ASoC: tlv320adc3xxx: Remove use of i2c_match_id() ASoC: tlv320aic31xx: Remove use of i2c_match_id() ASoC: tlv320aic3x: Remove use of i2c_match_id() ASoC: tpa6130a2: Remove use of i2c_match_id() ASoC: wm8904: Remove use of i2c_match_id() ASoC: wm8985: Remove use of i2c_match_id() Andy Shevchenko (1): ASoc: mediatek: mt8365: Don't use "proxy" headers Arnd Bergmann (1): firmware: cs_dsp: avoid large local variables Bard Liao (7): ASoC: SOF: Intel: hda-dai: Ensure DAI widget is valid during params ASoC: Intel: sof_sdw: correct mach_params->dmic_num ASoC: Intel: sof_sdw: reduce log level for not using internal dmic ASoC: Intel: sof_sdw: improve the log of DAI link numbers ASoC: Intel: soc-acpi-intel-ptl-match: add rt712_vb + rt1320 support ASoC: Intel: soc-acpi-intel-lnl-match: add rt713_vb_l2_rt1320_l13 support ASoC: Intel: soc-acpi-intel-ptl-match: add rt713_vb_l2_rt1320_l13 support Bence Csókás (1): ASoC: sun4i-codec: Use new devm clk and reset APIs Cezary Rojewski (9): ASoC: Intel: avs: Do not readq() u32 registers ASoC: Intel: avs: Fix the minimum firmware version numbers ASoC: Intel: avs: Fix theoretical infinite loop ASoC: Intel: avs: Update hda component teardown sequences ASoC: Intel: avs: Clearly state assumptions of hw_params() ASoC: Intel: avs: Improve logging of firmware loading ASoC: Intel: avs: Adjust DSP status register names ASoC: Intel: avs: Adjust IPC traces ASoC: Intel: avs: Add missing includes Chancel Liu (4): ASoC: fsl_utils: Add function to constrain rates ASoC: fsl_micfil: Switch to common sample rate constraint function ASoC: fsl_xcvr: Add sample rate constraint ASoC: fsl_sai: Add sample rate constraint Charles Keepax (5): ASoC: SDCA: Add missing header includes ASoC: SDCA: Clean up error messages ASoC: SDCA: Add bounds check for function address ASoC: SDCA: Add missing function type names ASoC: SDCA: Split function type patching and function naming Christophe JAILLET (1): ASoC: cs42l51: Constify struct i2c_device_id Claudiu Beznea (16): ASoC: renesas: rz-ssi: Terminate all the DMA transactions ASoC: renesas: rz-ssi: Use only the proper amount of dividers ASoC: renesas: rz-ssi: Fix typo on SSI_RATES macro comment ASoC: renesas: rz-ssi: Remove pdev member of struct rz_ssi_priv ASoC: renesas: rz-ssi: Remove the rz_ssi_get_dai() function ASoC: renesas: rz-ssi: Remove the first argument of rz_ssi_stream_is_play() ASoC: renesas: rz-ssi: Use readl_poll_timeout_atomic() ASoC: renesas: rz-ssi: Use temporary variable for struct device ASoC: renesas: rz-ssi: Use goto label names that specify their actions ASoC: renesas: rz-ssi: Rely on the ASoC subsystem to runtime resume/suspend the SSI ASoC: renesas: rz-ssi: Enable runtime PM autosuspend support ASoC: renesas: rz-ssi: Add runtime PM support ASoC: renesas: rz-ssi: Issue software reset in hw_params API ASoC: renesas: rz-ssi: Add suspend to RAM support ASoC: dt-bindings: renesas,rz-ssi: Remove DMA description ASoC: dt-bindings: renesas,rz-ssi: Document the Renesas RZ/G3S SoC Colin Ian King (1): ASoC: soc-core: remove redundant assignment to variable ret Csókás Bence (1): ASoC: dt-bindings: Add Allwinner suniv F1C100s Audio Codec Dan Carpenter (1): ASoC: renesas: rz-ssi: Add a check for negative sample_space Derek Fang (1): ASoC: rt5682: Support the ALC5682I-VE codec Dheeraj Reddy Jonnalagadda (1): ASoC: rockchip: i2s-tdm: Fix a useless call issue Dr. David Alan Gilbert (1): ASoC: rt715: Remove unused hda_to_sdw George Lander (1): ASoC: sun4i-spdif: Add clock multiplier settings Krzysztof Kozlowski (3): ASoC: codecs: wcd9335: Add define for number of DAIs ASoC: dt-bindings: qcom,wcd9335: Drop number of DAIs from the header ASoC: dt-bindings: Correct indentation and style in DTS example Kuninori Morimoto (18): ASoC: simple-card-utils: use __free(device_node) for device node ASoC: simple-card-utils: check port reg first on graph_get_dai_id() ASoC: simple-card-utils: use for_each_of_graph_port() on graph_get_dai_id() ASoC: soc-core: tidyup ret handling for card->disable_route_checks ASoC: soc-pcm: don't use soc_pcm_ret() on .prepare callback ASoC: simple-card-utils: Unify clock direction by clk_direction ASoC: dt-bindings: renesas,rsnd: remove post-init-providers property ASoC: soc-card: remove card check ASoC: intel: remove disable_route_checks ASoC: remove disable_route_checks ASoC: audio-graph-card2: use __free(device_node) for device node ASoC: audio-graph-card: use __free(device_node) for device node ASoC: simple-card: use __free(device_node) for device node ASoC: soc-core: return 0 if np was NULL on snd_soc_daifmt_parse_clock_provider_raw() ASoC: audio-graph-card2: use of_graph_get_port_by_id() at graph_get_next_multi_ep() ASoC: soc-core: Enable to use extra format on each DAI ASoC: audio-graph-card2: Use extra format on each DAI ASoC: soc-dapm: remove !card check from snd_soc_dapm_set_bias_level() Marcus Cooper (2): ASoC: sun4i-spdif: Always set the valid data to be the MSB ASoC: sun4i-spdif: Add working 24bit audio support Mark Brown (23): ASoC: fsl_xcvr: Add suspend and resume support ASoC: Correct *-objs usages Add support for codec of F1C100s ASoC: Splitting cs35l56 SoundWire DAI into separate ASoC: Merge up origin to resolve interaction with manline symbol changes ASoC: Merge up v6.12-rc2 ASoC: Intel: boards: updates for 6.14 ASoC: sun4i-spdif: Add 24bit support Add function to constrain rates ASoC: simple-card-utils: tidyup for Multi connection Add audio support for the Renesas RZ/G3S SoC ASoC: SOF: core/Intel: Handle pause supported token firmware: cirrus: Add KUnit tests for cs_dsp ASoC: fsl: add memory to memory function for ASRC ASoC: Intel: Add matches for Cirrus Logic CDB35L56 Minor debugging improvements ASoC: codecs: Add aw88083 amplifier driver ASoC: Merge up v6.13-rc6 ASoC: Intel: avs: Fixes and cleanups ASoC: remove disable_route_checks ASoC: fsl: Support MQS on i.MX943 ASoC: extra format on each DAI ASoC: fsl: Support micfil on i.MX943 Martin Blumenstingl (2): ASoC: soc-dai: add snd_soc_dai_prepare() and use it internally ASoC: dapm: add support for preparing streams Mesih Kilinc (2): ASoC: sun4i-codec: Add DMA Max Burst field ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s Michal Simek (1): ASoC: xilinx: xlnx_spdif: Simpify using devm_clk_get_enabled() Neil Armstrong (1): ASoC: dt-bindings: convert rt5682.txt to dt-schema Nícolas F. R. A. Prado (2): ASoC: dt-bindings: mediatek,mt8188-mt6359: Add compatible for mt8390 evk ASoC: dt-bindings: mediatek,mt8188-mt6359: Allow DL_SRC/UL_SRC dai-links Peter Ujfalusi (4): ASoC: SOF: Add support for pause supported tokens from topology ASoC: SOF: Intel: hda-pcm: Follow the pause_supported flag to drop PAUSE support ASoC: SOF: ipc4-topology: Use macro to set the EXT_PARAM_SIZE in widget setup ASoC: SOF: sof-priv: Remove unused SOF_DAI_STREAM() and SOF_FORMATS Richard Fitzgerald (21): ASoC: cs35l56: Split SoundWire DAI into separate playback and capture ASoC: sdw_utils: cs_amp: Assign non-overlapping TDM masks for each codec on a bus firmware: cs_dsp: Add mock regmap for KUnit testing firmware: cs_dsp: Add mock DSP memory map for KUnit testing firmware: cs_dsp: Add mock wmfw file generator for KUnit testing firmware: cs_dsp: Add mock bin file generator for KUnit testing firmware: cs_dsp: Add KUnit testing of bin file download firmware: cs_dsp: Add KUnit testing of wmfw download firmware: cs_dsp: Add KUnit testing of control parsing firmware: cs_dsp: Add KUnit testing of control cache firmware: cs_dsp: Add KUnit testing of control read/write firmware: cs_dsp: Add KUnit testing of bin error cases firmware: cs_dsp: Add KUnit testing of wmfw error cases firmware: cs_dsp: Add KUnit testing of client callbacks ASoC: Intel: tgl-match: Add entries for CS35L56 on CDB35L56-EIGHT-C ASoC: Intel: mtl-match: Add CDB35L56-EIGHT-C with aggregated speakers ASoC: Intel: mtl-match: Add CDB35L56-EIGHT-C 8x CS35L56 without CS42L43 firmware: cs_dsp: Fix kerneldoc typos in cs_dsp_mock_bin.c firmware: cs_dsp: Avoid using a u32 as a __be32 in cs_dsp_mock_mem_maps.c firmware: cs_dsp: Fix endianness conversion in cs_dsp_mock_wmfw.c firmware: cs_dsp: Delete redundant assignments in cs_dsp_test_bin.c Rob Herring (Arm) (2): ASoC: Use of_property_present() for non-boolean properties ASoC: simple-card: Drop unnecessary "dai-tdm-slot-width-map" property presence check Shengjiu Wang (13): ASoC: fsl_xcvr: Use regmap for PHY and PLL registers ASoC: fsl_xcvr: Add suspend and resume support ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device ASoC: fsl_asrc_m2m: force cast for snd_pcm_format_t type ASoC: fsl_mqs: Add i.MX943 platform support ASoC: dt-bindings: fsl,mqs: Add compatible string for i.MX943 platform ASoC: fsl_micfil: Add i.MX943 platform support ASoC: dt-bindings: fsl,micfil: Add compatible string for i.MX943 platform Simon Trimmer (4): ASoC: Intel: sof_sdw: Correct quirk for Lenovo Yoga Slim 7 ASoC: Intel: sof_sdw: Add a dev_dbg message for the SOC_SDW_CODEC_MIC quirk ASoC: Intel: soc-acpi: arl: Correct naming of a cs35l56 address struct ASoC: Intel: soc-acpi: arl: Add match entries for new cs42l43 laptops Stephen Gordon (2): ASoC: simple_card: Improve debugging messages ASoC: simple_card: Show if link is unidirectional Takashi Iwai (5): ASoC: cs40l50: Use *-y for Makefile ASoC: mediatek: mt8365: Use *-y for Makefile ASoC: SDCA: Use *-y for Makefile ASoC: cs42l84: Use *-y for Makefile ASoC: wcd937x: Use *-y for Makefile Tang Bin (2): ASoC: mediatek: mt8192-afe-pcm: Simplify probe() with local dev variable ASoC: tas2781: Fix redundant logical jump Thorsten Blum (4): ASoC: SOF: Intel: Use str_yes_no() helper in atom_dump() ASoC: madera: Use str_enabled_disabled() helper function ASoC: SOF: Intel: Use str_yes_no() to improve bdw_dump() ASoC: codecs: Use ARRAY_SIZE() to calculate PEB2466_TLV_SIZE Uwe Kleine-König (1): ASoC: Drop explicit initialization of struct i2c_device_id::driver_data to 0 Vijendar Mukunda (2): ASoC: amd: ps: update mach params subsystem_rev variable ASoC: amd: ps: add ZSC control register programming sequence Weidong Wang (2): ASoC: dt-bindings: Add schema for "awinic,aw88083" ASoC: codecs: Add aw88083 amplifier driver Wolfram Sang (1): ASoC: cs42l43: don't include '<linux/find.h>' directly Yu-Chun Lin (1): ASoC: amd: Add ACPI dependency to fix build error .../devicetree/bindings/sound/adi,ssm2518.yaml | 20 +- .../bindings/sound/allwinner,sun4i-a10-codec.yaml | 31 + .../devicetree/bindings/sound/awinic,aw88395.yaml | 2 + .../devicetree/bindings/sound/everest,es71x4.yaml | 10 +- .../devicetree/bindings/sound/everest,es7241.yaml | 19 +- .../devicetree/bindings/sound/fsl,easrc.yaml | 32 +- .../devicetree/bindings/sound/fsl,micfil.yaml | 1 + .../devicetree/bindings/sound/fsl,mqs.yaml | 2 + .../devicetree/bindings/sound/fsl,xcvr.yaml | 34 +- .../bindings/sound/intel,keembay-i2s.yaml | 32 +- .../bindings/sound/mediatek,mt8188-mt6359.yaml | 16 +- .../bindings/sound/neofidelity,ntp8918.yaml | 26 +- .../devicetree/bindings/sound/realtek,rt5682.yaml | 156 + .../devicetree/bindings/sound/renesas,rsnd.yaml | 6 - .../devicetree/bindings/sound/renesas,rz-ssi.yaml | 19 +- Documentation/devicetree/bindings/sound/rt5682.txt | 98 - .../devicetree/bindings/sound/ti,pcm6240.yaml | 32 +- .../devicetree/bindings/sound/ti,tas2562.yaml | 30 +- .../devicetree/bindings/sound/ti,tas2770.yaml | 34 +- .../devicetree/bindings/sound/ti,tas2781.yaml | 38 +- .../devicetree/bindings/sound/ti,tas27xx.yaml | 34 +- .../devicetree/bindings/sound/ti,tas57xx.yaml | 36 +- MAINTAINERS | 4 +- drivers/firmware/cirrus/Kconfig | 20 + drivers/firmware/cirrus/Makefile | 2 + drivers/firmware/cirrus/test/Makefile | 23 + drivers/firmware/cirrus/test/cs_dsp_mock_bin.c | 199 ++ .../firmware/cirrus/test/cs_dsp_mock_mem_maps.c | 752 +++++ drivers/firmware/cirrus/test/cs_dsp_mock_regmap.c | 367 +++ drivers/firmware/cirrus/test/cs_dsp_mock_utils.c | 13 + drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c | 473 +++ drivers/firmware/cirrus/test/cs_dsp_test_bin.c | 2556 +++++++++++++++ .../firmware/cirrus/test/cs_dsp_test_bin_error.c | 600 ++++ .../firmware/cirrus/test/cs_dsp_test_callbacks.c | 688 ++++ .../cirrus/test/cs_dsp_test_control_cache.c | 3282 ++++++++++++++++++++ .../cirrus/test/cs_dsp_test_control_parse.c | 1851 +++++++++++ .../firmware/cirrus/test/cs_dsp_test_control_rw.c | 2669 ++++++++++++++++ drivers/firmware/cirrus/test/cs_dsp_test_wmfw.c | 2211 +++++++++++++ .../firmware/cirrus/test/cs_dsp_test_wmfw_error.c | 1347 ++++++++ drivers/firmware/cirrus/test/cs_dsp_tests.c | 14 + include/dt-bindings/sound/qcom,wcd9335.h | 1 - include/linux/firmware/cirrus/cs_dsp_test_utils.h | 160 + include/sound/sdca.h | 7 +- include/sound/sdca_function.h | 3 + include/sound/simple_card_utils.h | 8 +- include/sound/soc-dai.h | 3 + include/sound/soc.h | 12 +- include/sound/soc_sdw_utils.h | 2 + include/uapi/sound/compress_params.h | 23 +- include/uapi/sound/sof/tokens.h | 2 + sound/soc/amd/Kconfig | 2 +- sound/soc/amd/ps/pci-ps.c | 16 +- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/Makefile | 8 +- sound/soc/codecs/ad193x-i2c.c | 3 +- sound/soc/codecs/adau1761-i2c.c | 5 +- sound/soc/codecs/adau1781-i2c.c | 5 +- sound/soc/codecs/adau1977-i2c.c | 5 +- sound/soc/codecs/alc5623.c | 10 +- sound/soc/codecs/alc5632.c | 6 +- sound/soc/codecs/aw88081.c | 333 +- sound/soc/codecs/aw88081.h | 43 + sound/soc/codecs/cs35l56.c | 8 +- sound/soc/codecs/cs42l43.c | 2 +- sound/soc/codecs/cs42l51-i2c.c | 6 +- sound/soc/codecs/cs42l84.c | 2 +- sound/soc/codecs/es8323.c | 2 +- sound/soc/codecs/madera.c | 7 +- sound/soc/codecs/max98088.c | 4 +- sound/soc/codecs/max98090.c | 18 +- sound/soc/codecs/max98095.c | 4 +- sound/soc/codecs/ntp8835.c | 2 +- sound/soc/codecs/ntp8918.c | 2 +- sound/soc/codecs/pcm186x-i2c.c | 3 +- sound/soc/codecs/pcm6240.c | 3 +- sound/soc/codecs/peb2466.c | 3 +- sound/soc/codecs/rt5682-i2c.c | 6 + sound/soc/codecs/rt5682.c | 12 +- sound/soc/codecs/rt5682.h | 2 + sound/soc/codecs/rt715-sdw.c | 41 - sound/soc/codecs/rt715.h | 3 - sound/soc/codecs/sma1307.c | 4 +- sound/soc/codecs/ssm2602-i2c.c | 5 +- sound/soc/codecs/tas2562.c | 4 +- sound/soc/codecs/tas2781-i2c.c | 71 +- sound/soc/codecs/tas5720.c | 10 +- sound/soc/codecs/tlv320adc3xxx.c | 4 +- sound/soc/codecs/tlv320aic31xx.c | 6 +- sound/soc/codecs/tlv320aic3x-i2c.c | 3 +- sound/soc/codecs/tpa6130a2.c | 4 +- sound/soc/codecs/uda1342.c | 2 +- sound/soc/codecs/wcd9335.c | 2 + sound/soc/codecs/wm8904.c | 13 +- sound/soc/codecs/wm8985.c | 4 +- sound/soc/fsl/Kconfig | 2 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/fsl_asrc.c | 179 +- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 + sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++ sound/soc/fsl/fsl_easrc.c | 261 +- sound/soc/fsl/fsl_easrc.h | 4 + sound/soc/fsl/fsl_micfil.c | 131 +- sound/soc/fsl/fsl_micfil.h | 2 +- sound/soc/fsl/fsl_mqs.c | 28 + sound/soc/fsl/fsl_sai.c | 7 +- sound/soc/fsl/fsl_sai.h | 3 + sound/soc/fsl/fsl_utils.c | 45 + sound/soc/fsl/fsl_utils.h | 5 + sound/soc/fsl/fsl_xcvr.c | 404 ++- sound/soc/fsl/fsl_xcvr.h | 13 + sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-card.c | 2 +- sound/soc/fsl/imx-rpmsg.c | 2 +- sound/soc/generic/audio-graph-card.c | 48 +- sound/soc/generic/audio-graph-card2.c | 262 +- sound/soc/generic/simple-card-utils.c | 79 +- sound/soc/generic/simple-card.c | 58 +- sound/soc/intel/avs/apl.c | 2 +- sound/soc/intel/avs/core.c | 24 +- sound/soc/intel/avs/debugfs.c | 1 + sound/soc/intel/avs/ipc.c | 25 +- sound/soc/intel/avs/loader.c | 36 +- sound/soc/intel/avs/messages.c | 22 +- sound/soc/intel/avs/messages.h | 3 +- sound/soc/intel/avs/pcm.c | 5 +- sound/soc/intel/avs/registers.h | 2 +- sound/soc/intel/avs/topology.c | 4 +- sound/soc/intel/avs/trace.h | 38 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 - sound/soc/intel/boards/sof_sdw.c | 33 +- sound/soc/intel/common/soc-acpi-intel-arl-match.c | 45 +- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 70 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 289 +- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 148 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 194 ++ sound/soc/intel/keembay/kmb_platform.c | 2 +- sound/soc/mediatek/common/mtk-soundcard-driver.c | 4 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 19 +- sound/soc/mediatek/mt8365/Makefile | 2 +- sound/soc/mediatek/mt8365/mt8365-mt6357.c | 11 +- sound/soc/qcom/common.c | 6 +- sound/soc/qcom/sc7180.c | 2 +- sound/soc/qcom/sdm845.c | 5 +- sound/soc/renesas/rz-ssi.c | 228 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 28 - sound/soc/sdca/Makefile | 2 +- sound/soc/sdca/sdca_device.c | 2 + sound/soc/sdca/sdca_functions.c | 132 +- sound/soc/sdw_utils/soc_sdw_cs_amp.c | 46 + sound/soc/sdw_utils/soc_sdw_utils.c | 13 +- sound/soc/soc-card.c | 4 +- sound/soc/soc-core.c | 58 +- sound/soc/soc-dai.c | 27 +- sound/soc/soc-dapm.c | 14 +- sound/soc/soc-pcm.c | 32 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/atom.c | 16 +- sound/soc/sof/intel/bdw.c | 16 +- sound/soc/sof/intel/hda-dai.c | 12 + sound/soc/sof/intel/hda-pcm.c | 15 + sound/soc/sof/intel/hda.c | 5 + sound/soc/sof/ipc4-topology.c | 2 +- sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/sof-priv.h | 8 - sound/soc/sof/topology.c | 4 + sound/soc/sunxi/sun4i-codec.c | 409 ++- sound/soc/sunxi/sun4i-spdif.c | 24 +- sound/soc/xilinx/xlnx_spdif.c | 38 +- 170 files changed, 21739 insertions(+), 1441 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5682.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt5682.txt create mode 100644 drivers/firmware/cirrus/test/Makefile create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_bin.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_mem_maps.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_regmap.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_utils.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_bin.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_bin_error.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_callbacks.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_control_cache.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_control_parse.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_control_rw.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_wmfw.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_wmfw_error.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_tests.c create mode 100644 include/linux/firmware/cirrus/cs_dsp_test_utils.h create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c

2 1

[PATCH V2 7/7] soundwire: amd: add soundwire host wake interrupt enable/disable sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

For wake event, SoundWire host wake interrupt will be asserted based on below pre-conditions for ACP7.0 & ACP7.1 platforms. - ACP device should be in D0 state. - SoundWire manager instance should be in D3 state. - SoundWire manager device state should be set to D3. - ACP_PME_EN should be set to 1. Implement code changes to enable/disable SoundWire host wake interrupt mask during suspend and resume as per design flow for ACP7.0 & ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 58 +++++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 2 ++ 2 files changed, 60 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 03c9e15dce5b..161d963c80e6 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -166,6 +166,34 @@ static int amd_sdw_set_device_state(struct amd_sdw_manager *amd_manager, u32 tar return 0; } +static int amd_sdw_host_wake_enable(struct amd_sdw_manager *amd_manager, bool enable) +{ + u32 intr_cntl1; + u32 sdw_host_wake_irq_mask; + + if (!amd_manager->wake_en_mask) + return 0; + + switch (amd_manager->instance) { + case ACP_SDW0: + sdw_host_wake_irq_mask = AMD_SDW0_HOST_WAKE_INTR_MASK; + break; + case ACP_SDW1: + sdw_host_wake_irq_mask = AMD_SDW1_HOST_WAKE_INTR_MASK; + break; + default: + return -EINVAL; + } + + intr_cntl1 = readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(ACP_SDW1)); + if (enable) + intr_cntl1 |= sdw_host_wake_irq_mask; + else + intr_cntl1 &= ~sdw_host_wake_irq_mask; + writel(intr_cntl1, amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(ACP_SDW1)); + return 0; +} + static void amd_sdw_ctl_word_prep(u32 *lower_word, u32 *upper_word, struct sdw_msg *msg, int cmd_offset) { @@ -1186,11 +1214,21 @@ static int __maybe_unused amd_suspend(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, false); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, false); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } /* * As per hardware programming sequence on AMD platforms, * clock stop should be invoked first before powering-off @@ -1224,11 +1262,21 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) } if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, true); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, true); + if (ret) + return ret; + } ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, true); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, true); + if (ret) + return ret; + } ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; @@ -1269,8 +1317,18 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) ret = amd_sdw_clock_stop_exit(amd_manager); if (ret) return ret; + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 1d5e94371f81..6cc916b0c820 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -194,6 +194,8 @@ #define AMD_SDW_CLK_RESUME_DONE 3 #define AMD_SDW_WAKE_STAT_MASK BIT(16) #define AMD_SDW_WAKE_INTR_MASK BIT(16) +#define AMD_SDW0_HOST_WAKE_INTR_MASK BIT(22) +#define AMD_SDW1_HOST_WAKE_INTR_MASK BIT(23) #define AMD_SDW_DEVICE_STATE 0x1430 #define AMD_SDW0_DEVICE_STATE_MASK GENMASK(1, 0) #define AMD_SDW1_DEVICE_STATE_MASK GENMASK(3, 2) -- 2.34.1

1 0

[PATCH V2 6/7] soundwire: amd: set ACP_PME_EN during runtime suspend sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Set ACP_PME_EN to 1 during runtime suspend sequence as per design flow for ACP7.0 & ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 9 +++++++++ drivers/soundwire/amd_manager.h | 1 + 2 files changed, 10 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 42d553878e91..03c9e15dce5b 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1215,6 +1215,7 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) struct amd_sdw_manager *amd_manager = dev_get_drvdata(dev); struct sdw_bus *bus = &amd_manager->bus; int ret; + u32 val; if (bus->prop.hw_disabled) { dev_dbg(bus->dev, "SoundWire manager %d is disabled,\n", @@ -1239,6 +1240,14 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D3); if (ret) return ret; + if (amd_manager->wake_en_mask) { + val = readl(amd_manager->acp_mmio + ACP_PME_EN); + if (!val) { + writel(1, amd_manager->acp_mmio + ACP_PME_EN); + val = readl(amd_manager->acp_mmio + ACP_PME_EN); + dev_dbg(amd_manager->dev, "ACP_PME_EN:0x%x\n", val); + } + } } return 0; } diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 8430f279d88e..1d5e94371f81 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -199,6 +199,7 @@ #define AMD_SDW1_DEVICE_STATE_MASK GENMASK(3, 2) #define AMD_SDW_DEVICE_STATE_D0 0 #define AMD_SDW_DEVICE_STATE_D3 3 +#define ACP_PME_EN 0x0001400 static u32 amd_sdw_freq_tbl[AMD_SDW_MAX_FREQ_NUM] = { AMD_SDW_DEFAULT_CLK_FREQ, -- 2.34.1

1 0

[PATCH V2 5/7] soundwire: amd: set device power state during suspend/resume sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Set SoundWire manager device power state during suspend and resume sequence for ACP7.0 & ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 58 ++++++++++++++++++++++++++++++--- drivers/soundwire/amd_manager.h | 5 +++ 2 files changed, 58 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index d611498d5488..42d553878e91 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -143,6 +143,29 @@ static void amd_sdw_wake_enable(struct amd_sdw_manager *amd_manager, bool enable writel(wake_ctrl, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_MASK_8TO11); } +static int amd_sdw_set_device_state(struct amd_sdw_manager *amd_manager, u32 target_device_state) +{ + u32 sdw_dev_state; + + sdw_dev_state = readl(amd_manager->acp_mmio + AMD_SDW_DEVICE_STATE); + switch (amd_manager->instance) { + case ACP_SDW0: + u32p_replace_bits(&sdw_dev_state, target_device_state, + AMD_SDW0_DEVICE_STATE_MASK); + break; + case ACP_SDW1: + u32p_replace_bits(&sdw_dev_state, target_device_state, + AMD_SDW1_DEVICE_STATE_MASK); + break; + default: + return -EINVAL; + } + writel(sdw_dev_state, amd_manager->acp_mmio + AMD_SDW_DEVICE_STATE); + sdw_dev_state = readl(amd_manager->acp_mmio + AMD_SDW_DEVICE_STATE); + dev_dbg(amd_manager->dev, "AMD_SDW_DEVICE_STATE:0x%x\n", sdw_dev_state); + return 0; +} + static void amd_sdw_ctl_word_prep(u32 *lower_word, u32 *upper_word, struct sdw_msg *msg, int cmd_offset) { @@ -1163,7 +1186,9 @@ static int __maybe_unused amd_suspend(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, false); - return amd_sdw_clock_stop(amd_manager); + ret = amd_sdw_clock_stop(amd_manager); + if (ret) + return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, false); /* @@ -1173,7 +1198,14 @@ static int __maybe_unused amd_suspend(struct device *dev) ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; - return amd_deinit_sdw_manager(amd_manager); + ret = amd_deinit_sdw_manager(amd_manager); + if (ret) + return ret; + } + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D3); + if (ret) + return ret; } return 0; } @@ -1191,13 +1223,22 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) } if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, true); - return amd_sdw_clock_stop(amd_manager); + ret = amd_sdw_clock_stop(amd_manager); + if (ret) + return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, true); ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; - return amd_deinit_sdw_manager(amd_manager); + ret = amd_deinit_sdw_manager(amd_manager); + if (ret) + return ret; + } + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D3); + if (ret) + return ret; } return 0; } @@ -1216,7 +1257,9 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) } if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { - return amd_sdw_clock_stop_exit(amd_manager); + ret = amd_sdw_clock_stop_exit(amd_manager); + if (ret) + return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); @@ -1239,6 +1282,11 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) return ret; amd_sdw_set_frameshape(amd_manager); } + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D0); + if (ret) + return ret; + } return 0; } diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 30244a31c21c..8430f279d88e 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -194,6 +194,11 @@ #define AMD_SDW_CLK_RESUME_DONE 3 #define AMD_SDW_WAKE_STAT_MASK BIT(16) #define AMD_SDW_WAKE_INTR_MASK BIT(16) +#define AMD_SDW_DEVICE_STATE 0x1430 +#define AMD_SDW0_DEVICE_STATE_MASK GENMASK(1, 0) +#define AMD_SDW1_DEVICE_STATE_MASK GENMASK(3, 2) +#define AMD_SDW_DEVICE_STATE_D0 0 +#define AMD_SDW_DEVICE_STATE_D3 3 static u32 amd_sdw_freq_tbl[AMD_SDW_MAX_FREQ_NUM] = { AMD_SDW_DEFAULT_CLK_FREQ, -- 2.34.1

1 0

[PATCH V2 4/7] soundwire: amd: add support for ACP7.0 & ACP7.1 platforms
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Add SoundWire support for ACP7.0 and ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 22 ++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 18 ++++++++++++++++++ include/linux/soundwire/sdw_amd.h | 2 ++ 3 files changed, 42 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 6831b3729db5..d611498d5488 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -446,6 +446,10 @@ static int amd_sdw_port_params(struct sdw_bus *bus, struct sdw_port_params *p_pa return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + frame_fmt_reg = acp70_sdw_dp_reg[p_params->num].frame_fmt_reg; + break; default: return -EINVAL; } @@ -494,6 +498,14 @@ static int amd_sdw_transport_params(struct sdw_bus *bus, return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + frame_fmt_reg = acp70_sdw_dp_reg[params->port_num].frame_fmt_reg; + sample_int_reg = acp70_sdw_dp_reg[params->port_num].sample_int_reg; + hctrl_dp0_reg = acp70_sdw_dp_reg[params->port_num].hctrl_dp0_reg; + offset_reg = acp70_sdw_dp_reg[params->port_num].offset_reg; + lane_ctrl_ch_en_reg = acp70_sdw_dp_reg[params->port_num].lane_ctrl_ch_en_reg; + break; default: return -EINVAL; } @@ -549,6 +561,10 @@ static int amd_sdw_port_enable(struct sdw_bus *bus, return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + lane_ctrl_ch_en_reg = acp70_sdw_dp_reg[enable_ch->port_num].lane_ctrl_ch_en_reg; + break; default: return -EINVAL; } @@ -898,6 +914,7 @@ int amd_sdw_manager_start(struct amd_sdw_manager *amd_manager) ret = amd_enable_sdw_manager(amd_manager); if (ret) return ret; + amd_sdw_set_frameshape(amd_manager); } /* Enable runtime PM */ @@ -969,6 +986,11 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + amd_manager->num_dout_ports = AMD_ACP70_SDW_MAX_TX_PORTS; + amd_manager->num_din_ports = AMD_ACP70_SDW_MAX_RX_PORTS; + break; default: return -EINVAL; } diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index cc2170e4521e..30244a31c21c 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -159,8 +159,11 @@ #define AMD_ACP63_SDW0_MAX_RX_PORTS 3 #define AMD_ACP63_SDW1_MAX_TX_PORTS 1 #define AMD_ACP63_SDW1_MAX_RX_PORTS 1 +#define AMD_ACP70_SDW_MAX_TX_PORTS 3 +#define AMD_ACP70_SDW_MAX_RX_PORTS 3 #define AMD_ACP63_SDW0_MAX_DAI 6 #define AMD_ACP63_SDW1_MAX_DAI 2 +#define AMD_ACP70_SDW_MAX_DAI 6 #define AMD_SDW_SLAVE_0_ATTACHED 5 #define AMD_SDW_SSP_COUNTER_VAL 3 @@ -244,6 +247,21 @@ static struct sdw_manager_dp_reg acp63_sdw1_dp_reg[AMD_ACP63_SDW1_MAX_DAI] = { ACP_SW_AUDIO1_RX_OFFSET, ACP_SW_AUDIO1_RX_CHANNEL_ENABLE_DP0} }; +static struct sdw_manager_dp_reg acp70_sdw_dp_reg[AMD_ACP70_SDW_MAX_DAI] = { + {ACP_SW_AUDIO0_TX_FRAME_FORMAT, ACP_SW_AUDIO0_TX_SAMPLEINTERVAL, ACP_SW_AUDIO0_TX_HCTRL_DP0, + ACP_SW_AUDIO0_TX_OFFSET_DP0, ACP_SW_AUDIO0_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO1_TX_FRAME_FORMAT, ACP_SW_AUDIO1_TX_SAMPLEINTERVAL, ACP_SW_AUDIO1_TX_HCTRL, + ACP_SW_AUDIO1_TX_OFFSET, ACP_SW_AUDIO1_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO2_TX_FRAME_FORMAT, ACP_SW_AUDIO2_TX_SAMPLEINTERVAL, ACP_SW_AUDIO2_TX_HCTRL, + ACP_SW_AUDIO2_TX_OFFSET, ACP_SW_AUDIO2_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO0_RX_FRAME_FORMAT, ACP_SW_AUDIO0_RX_SAMPLEINTERVAL, ACP_SW_AUDIO0_RX_HCTRL_DP0, + ACP_SW_AUDIO0_RX_OFFSET_DP0, ACP_SW_AUDIO0_RX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO1_RX_FRAME_FORMAT, ACP_SW_AUDIO1_RX_SAMPLEINTERVAL, ACP_SW_AUDIO1_RX_HCTRL, + ACP_SW_AUDIO1_RX_OFFSET, ACP_SW_AUDIO1_RX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO2_RX_FRAME_FORMAT, ACP_SW_AUDIO2_RX_SAMPLEINTERVAL, ACP_SW_AUDIO2_RX_HCTRL, + ACP_SW_AUDIO2_RX_OFFSET, ACP_SW_AUDIO2_RX_CHANNEL_ENABLE_DP0}, +}; + static u32 sdw_manager_reg_mask_array[AMD_SDW_MAX_MANAGER_COUNT] = { AMD_SDW0_EXT_INTR_MASK, AMD_SDW1_EXT_INTR_MASK diff --git a/include/linux/soundwire/sdw_amd.h b/include/linux/soundwire/sdw_amd.h index 799f8578137b..6b839987f14c 100644 --- a/include/linux/soundwire/sdw_amd.h +++ b/include/linux/soundwire/sdw_amd.h @@ -28,6 +28,8 @@ #define ACP_SDW1 1 #define AMD_SDW_MAX_MANAGER_COUNT 2 #define ACP63_PCI_REV_ID 0x63 +#define ACP70_PCI_REV_ID 0x70 +#define ACP71_PCI_REV_ID 0x71 struct acp_sdw_pdata { u16 instance; -- 2.34.1

1 0

[PATCH V2 2/7] soundwire: amd: add debug log for soundwire wake event
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Add debug log in amd_sdw_process_wake_event() function. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 187006e68ad6..60be5805715e 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -849,6 +849,7 @@ static void amd_sdw_update_slave_status(u32 status_change_0to7, u32 status_chang static void amd_sdw_process_wake_event(struct amd_sdw_manager *amd_manager) { + dev_dbg(amd_manager->dev, "SoundWire Wake event reported\n"); pm_request_resume(amd_manager->dev); writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_8TO11); -- 2.34.1

1 0

[PATCH V2 1/7] soundwire: amd: change the soundwire wake enable/disable sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

During runtime suspend scenario, SoundWire wake should be enabled and during system level suspend scenario SoundWire wake should be disabled. Implement the SoundWire wake enable/disable sequence as per design flow for SoundWire poweroff mode. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 71b3cf2df89d..187006e68ad6 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1139,6 +1139,7 @@ static int __maybe_unused amd_suspend(struct device *dev) amd_sdw_wake_enable(amd_manager, false); return amd_sdw_clock_stop(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + amd_sdw_wake_enable(amd_manager, false); /* * As per hardware programming sequence on AMD platforms, * clock stop should be invoked first before powering-off @@ -1166,6 +1167,7 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) amd_sdw_wake_enable(amd_manager, true); return amd_sdw_clock_stop(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + amd_sdw_wake_enable(amd_manager, true); ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; -- 2.34.1

1 0

[PATCH v1 1/3] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 20 Jan '25

20 Jan '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - update the year from 2024 to 2025. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. --- include/sound/tas2781-dsp.h | 26 +++++++++++++++++++++++--- include/sound/tas2781.h | 13 ++++++++++++- 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..5cfbb9e0a626 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); -- 2.34.1

1 2

Is function delay of the pulse plugin correct?
by Pavel Hofman 19 Jan '25

19 Jan '25

Hi, Please correct me if I am wrong: "avail" means (buffer - buffer_fill), while "delay" means (buffer_fill + device/downstream delay/latency). Some drivers/devices do not track the device delay, and then delay = buffer_fill. While troubleshooting some playback code, I hit that snd_pcm_delay on the plugin pulse (device pulse) was always returning 0 frames, while snd_pcm_avail was returning dynamic (correct) values. IIUC while pulse_pcm.c uses generic code for avail, it overrides the generic snd_pcm_ioplug_delay https://github.com/alsa-project/alsa-lib/blob/35d2efefa95d28e0cd963a06e051f… with specific pulse_delay https://github.com/alsa-project/alsa-plugins/blob/52574cb5ccbb8b546df2759e4… . IIUC this method returns only delay obtained from Pulseaudio, i.e. no fill of the pulse plugin buffer is included. And IIUC in this specific case pulseaudio keeps returning zero for the pa_stream_get_latency call, making snd_pcm_delay return zero too. Therefore the equation (buffer size = avail + delay - driver/device delay) does not seem to hold for the pulse plugin. The new pipewire plugin does not seem to override the generic snd_pcm_ioplug_delay method https://github.com/PipeWire/pipewire-alsa/blob/master/alsa-plugins/pcm_pipe… . Please, is that behavior correct? With regards, Pavel.

1 0

[tiwai-sound:for-next 50/52] sound/usb/fcp.c:889:15: error: incompatible function pointer types assigning to 'long (*)(struct snd_hwdep *, char *, long, loff_t *)' (aka 'long (*)(struct snd_hwdep *, char *, long, long long *)') from 'ssize_t (struct snd_hwdep *, char *, ssize_t, loff...
by kernel test robot 19 Jan '25

19 Jan '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: dd85bbb7fc482b5cdb78f8b5e8f9ef7a3919d6ef commit: 46757a3e7d50dac923888e7fbe68377736f13c70 [50/52] ALSA: FCP: Add Focusrite Control Protocol driver config: hexagon-allmodconfig (https://download.01.org/0day-ci/archive/20250119/202501192122.ngosycnT-lkp@…) compiler: clang version 20.0.0git (https://github.com/llvm/llvm-project c23f2417dc5f6dc371afb07af5627ec2a9d373a0) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250119/202501192122.ngosycnT-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202501192122.ngosycnT-lkp@intel.com/ All errors (new ones prefixed by >>): >> sound/usb/fcp.c:889:15: error: incompatible function pointer types assigning to 'long (*)(struct snd_hwdep *, char *, long, loff_t *)' (aka 'long (*)(struct snd_hwdep *, char *, long, long long *)') from 'ssize_t (struct snd_hwdep *, char *, ssize_t, loff_t *)' (aka 'int (struct snd_hwdep *, char *, int, long long *)') [-Wincompatible-function-pointer-types] 889 | hw->ops.read = fcp_hwdep_read; | ^ ~~~~~~~~~~~~~~ 1 error generated. vim +889 sound/usb/fcp.c 874 875 static int fcp_hwdep_init(struct usb_mixer_interface *mixer) 876 { 877 struct snd_hwdep *hw; 878 int err; 879 880 err = snd_hwdep_new(mixer->chip->card, "Focusrite Control", 0, &hw); 881 if (err < 0) 882 return err; 883 884 hw->private_data = mixer; 885 hw->exclusive = 1; 886 hw->ops.open = fcp_hwdep_open; 887 hw->ops.ioctl = fcp_hwdep_ioctl; 888 hw->ops.ioctl_compat = fcp_hwdep_ioctl; > 889 hw->ops.read = fcp_hwdep_read; 890 hw->ops.poll = fcp_hwdep_poll; 891 hw->ops.release = fcp_hwdep_release; 892 893 return 0; 894 } 895 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 49/52] sound/pci/hda/tas2781_hda_spi.c:110:6: warning: variable 'ret' is used uninitialized whenever 'if' condition is false
by kernel test robot 19 Jan '25

19 Jan '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: dd85bbb7fc482b5cdb78f8b5e8f9ef7a3919d6ef commit: bb5f86ea50ffb292f42eb1ebdb99991d5c5ac3ba [49/52] ALSA: hda/tas2781: Add tas2781 hda SPI driver config: arm64-allmodconfig (https://download.01.org/0day-ci/archive/20250119/202501192006.Hm9GmKiV-lkp@…) compiler: clang version 18.1.8 (https://github.com/llvm/llvm-project 3b5b5c1ec4a3095ab096dd780e84d7ab81f3d7ff) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250119/202501192006.Hm9GmKiV-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202501192006.Hm9GmKiV-lkp@intel.com/ All warnings (new ones prefixed by >>): >> sound/pci/hda/tas2781_hda_spi.c:110:6: warning: variable 'ret' is used uninitialized whenever 'if' condition is false [-Wsometimes-uninitialized] 110 | if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/tas2781_hda_spi.c:119:9: note: uninitialized use occurs here 119 | return ret; | ^~~ sound/pci/hda/tas2781_hda_spi.c:110:2: note: remove the 'if' if its condition is always true 110 | if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/tas2781_hda_spi.c:108:9: note: initialize the variable 'ret' to silence this warning 108 | int ret; | ^ | = 0 1 warning generated. vim +110 sound/pci/hda/tas2781_hda_spi.c 104 105 static int tasdevice_spi_switch_book(struct tasdevice_priv *tas_priv, int reg) 106 { 107 struct regmap *map = tas_priv->regmap; 108 int ret; 109 > 110 if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { 111 ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, 112 TASDEVICE_BOOK_ID(reg)); 113 if (ret < 0) { 114 dev_err(tas_priv->dev, "Switch Book E=%d\n", ret); 115 return ret; 116 } 117 tas_priv->cur_book = TASDEVICE_BOOK_ID(reg); 118 } 119 return ret; 120 } 121 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 50/51] sound/usb/fcp.c:889:22: error: assignment to 'long int (*)(struct snd_hwdep *, char *, long int, loff_t *)' {aka 'long int (*)(struct snd_hwdep *, char *, long int, long long int *)'} from incompatible pointer type 'ssize_t (*)(struct snd_hwdep *, char ...
by kernel test robot 18 Jan '25

18 Jan '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 0ce204d3af3beca1825018e9ca128635ccc8aa85 commit: 46757a3e7d50dac923888e7fbe68377736f13c70 [50/51] ALSA: FCP: Add Focusrite Control Protocol driver config: i386-buildonly-randconfig-004-20250119 (https://download.01.org/0day-ci/archive/20250119/202501190536.A2CiOP0a-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250119/202501190536.A2CiOP0a-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202501190536.A2CiOP0a-lkp@intel.com/ All errors (new ones prefixed by >>): sound/usb/fcp.c: In function 'fcp_hwdep_init': >> sound/usb/fcp.c:889:22: error: assignment to 'long int (*)(struct snd_hwdep *, char *, long int, loff_t *)' {aka 'long int (*)(struct snd_hwdep *, char *, long int, long long int *)'} from incompatible pointer type 'ssize_t (*)(struct snd_hwdep *, char *, ssize_t, loff_t *)' {aka 'int (*)(struct snd_hwdep *, char *, int, long long int *)'} [-Werror=incompatible-pointer-types] 889 | hw->ops.read = fcp_hwdep_read; | ^ cc1: some warnings being treated as errors vim +889 sound/usb/fcp.c 874 875 static int fcp_hwdep_init(struct usb_mixer_interface *mixer) 876 { 877 struct snd_hwdep *hw; 878 int err; 879 880 err = snd_hwdep_new(mixer->chip->card, "Focusrite Control", 0, &hw); 881 if (err < 0) 882 return err; 883 884 hw->private_data = mixer; 885 hw->exclusive = 1; 886 hw->ops.open = fcp_hwdep_open; 887 hw->ops.ioctl = fcp_hwdep_ioctl; 888 hw->ops.ioctl_compat = fcp_hwdep_ioctl; > 889 hw->ops.read = fcp_hwdep_read; 890 hw->ops.poll = fcp_hwdep_poll; 891 hw->ops.release = fcp_hwdep_release; 892 893 return 0; 894 } 895 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

No audio on Acer Swift SF315-52
by GitHub issues - opened 18 Jan '25

18 Jan '25

alsa-project/alsa-ucm-conf issue #497 was opened from pavethem: [Alsa info](http://alsa-project.org/db/?f=8e6dc59510d7aaa42899f81a8f29a740032e9ce… Ever since kernel 6.12 dropped for Fedora, my laptop only shows a dummy output. This is probably due to the kernel removing legacy sound drivers and replacing them with avs, which doesn’t seem work for the kaby lake architecture present in this model. ([phoronix](https://www.phoronix.com/news/Linux-6.12-Sound-Drivers)). I am unsure whether or not I am filing this bug in the right place. Since I am able to get the microphone to work by copying a file: `cp /usr/share/alsa/ucm2/conf.d/avs_dmic/Acer-Lars-1.0.conf /usr/share/alsa/ucm2/conf.d/avs_dmic/avs_dmic.conf` I am guessing this is a case of a missing ucm file for this particular hardware. If this is incorrect, I would appreciate if someone could point me in the right direction and tell me if this is rather a bug with alsa-firmware or the kernel itself. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/497 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.13-rc7
by Mark Brown 18 Jan '25

18 Jan '25

The following changes since commit 139fa599cea0fd9d38e00246ea9f79af6c59acbd: ASoC: rsnd: check rsnd_adg_clk_enable() return value (2025-01-09 13:17:02 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc7 for you to fetch changes up to 75a7f7d6e7e10b9052d287be37f965103e1abae1: ASoC: Intel: sof_sdw: Fix DMI match entries for a (2025-01-15 19:50:01 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A few final driver specific fixes - a couple of x86 ID field changes, plus bug fixes for simple-card-utils and nau8824. ---------------------------------------------------------------- Laurentiu Mihalcea (1): ASoC: simple-card-utils: fix priv->dai_props indexing Mark Brown (1): ASoC: Intel: sof_sdw: Fix DMI match entries for a Maxim Kochetkov (1): ASoC: codecs: nau8824: fix max volume for Speaker output Simon Trimmer (2): ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83LC ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83JX, 83MC and 83NM include/sound/simple_card_utils.h | 7 +++++++ sound/soc/codecs/nau8824.c | 8 ++++---- sound/soc/generic/simple-card-utils.c | 10 +++++----- sound/soc/intel/boards/sof_sdw.c | 23 +++++++++++++++++++---- 4 files changed, 35 insertions(+), 13 deletions(-)

2 1

RE: [REGRESSION] Distorted sound on Acer Aspire A115-31 laptop
by Kailang 17 Jan '25

17 Jan '25

Attached. > -----Original Message----- > From: Takashi Iwai <tiwai(a)suse.de> > Sent: Thursday, January 16, 2025 10:27 PM > To: Evgeny Kapun <abacabadabacaba(a)gmail.com> > Cc: Kailang <kailang(a)realtek.com>; Takashi Iwai <tiwai(a)suse.de>; Linux > Sound Mailing List <linux-sound(a)vger.kernel.org>; Linux Kernel Mailing List > <linux-kernel(a)vger.kernel.org>; Linux Regressions Mailing List > <regressions(a)lists.linux.dev>; Linux Stable Mailing List > <stable(a)vger.kernel.org> > Subject: Re: [REGRESSION] Distorted sound on Acer Aspire A115-31 laptop > > > External mail. > > > > On Sat, 11 Jan 2025 16:00:33 +0100, > Evgeny Kapun wrote: > > > > On 12/24/24 04:54, Kailang wrote: > > > Please test attach patch. > > > > This patch, when applied to kernel version 6.12.8, appears to fix the > > issue. There are no distortions, and the left and the right channel > > can be controlled independently. > > Good to hear. > > Kailang, care to submit a proper patch for merging? > > > thanks, > > Takashi

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Topology: NHLT: Intel: Fix DMA slots config in SSP blob
by GitHub pull_request - opened 16 Jan '25

16 Jan '25

alsa-project/alsa-utils pull request #289 was opened from singalsu: This fixes a mistake in function ssp_calculate_intern_v30(). The rx_dir[0] and tx_dir[0] value was hard-coded to 0x3 for typical stereo slots configuration. It should be instead retrieved from topology parameters for the SSP. Request URL : https://github.com/alsa-project/alsa-utils/pull/289 Patch URL : https://github.com/alsa-project/alsa-utils/pull/289.patch Repository URL: https://github.com/alsa-project/alsa-utils

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no internal sound with snd_hda_intel, ALC256/ Lenovo ThinkCentre M90a Pro Gen4
by Ronny Sattler 15 Jan '25

15 Jan '25

Hello, I have a problem with the snd-hda-intel. There is no sound. Only when I plug in headphones do I hear the sound on the headphones. The speakers are onboard (All in One - PC) and always remain mute. I suspect it is because the amplifier is not activated or a PIN assignment is incorrect. Please let me know if there is anything else I can do for you on the PC. Attached are previous log files and general information. Thank you very much for your work :-) Best regards Ronny

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[RFC PATCH] soc: audio-graph-card2: use correct endpoint when getting link parameters
by Ivaylo Dimitrov 15 Jan '25

15 Jan '25

We may have multiple links between ports, with each link having different parameters. Currently, no matter the topology, it is always port endpoint 0 that is used when setting parameters. On a complex sound system, like the one found on Motorola droid4, hifi and voice DAIs require differents formats (i2s vs dsp_a) and curently it is impossible to use DT to set that. Implementing the change leads to partially dropping of at least 0dedbde5062d (ASoC: cpcap: Implement set_tdm_slot for voice call support), as core does most of what is needed to configure voice DAI. We (on Maemo Leste ) use the patch (along with few others) to have voice calls working properly on d4 through UCM. The patch is for linux 6.6, I want to know whether the approach would be accepted before sending a proper patch for current master. the original commit message follows: When link parameters are parsed, it is always endpoint@0 that is used and parameters set to other endpoints are ignored. Fix that by using endpoint that is set in DT when parsing link parameters. Signed-off-by: Ivaylo Dimitrov <ivo.g.dimitrov.75(a)gmail.com> --- sound/soc/generic/audio-graph-card2.c | 59 +++++++++++++-------------- 1 file changed, 28 insertions(+), 31 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index b1c675c6b6db..163a20c8ffee 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -508,17 +508,16 @@ static int __graph_parse_node(struct asoc_simple_priv *priv, static int graph_parse_node(struct asoc_simple_priv *priv, enum graph_type gtype, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu) { - struct device_node *ep; int ret = 0; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); - if (graph_lnk_is_multi(port)) { + if (is_multi) { int idx; - of_node_get(port); - for (idx = 0;; idx++) { ep = graph_get_next_multi_ep(&port); if (!ep) @@ -532,9 +531,8 @@ static int graph_parse_node(struct asoc_simple_priv *priv, } } else { /* Single CPU / Codec */ - ep = port_to_endpoint(port); + of_node_put(port); ret = __graph_parse_node(priv, gtype, ep, li, is_cpu, 0); - of_node_put(ep); } return ret; @@ -591,22 +589,20 @@ static void graph_parse_daifmt(struct device_node *node, } static void graph_link_init(struct asoc_simple_priv *priv, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu_node) { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct device_node *ep; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); struct device_node *ports; unsigned int daifmt = 0, daiclk = 0; unsigned int bit_frame = 0; - if (graph_lnk_is_multi(port)) { - of_node_get(port); + if (is_multi) { ep = graph_get_next_multi_ep(&port); port = of_get_parent(ep); - } else { - ep = port_to_endpoint(port); } ports = of_get_parent(port); @@ -642,6 +638,9 @@ static void graph_link_init(struct asoc_simple_priv *priv, dai_link->ops = &graph_ops; if (priv->ops) dai_link->ops = priv->ops; + + of_node_put(port); + of_node_put(ports); } int audio_graph2_link_normal(struct asoc_simple_priv *priv, @@ -650,7 +649,7 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, { struct device_node *cpu_port = lnk; struct device_node *cpu_ep = port_to_endpoint(cpu_port); - struct device_node *codec_port = of_graph_get_remote_port(cpu_ep); + struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); int ret; /* @@ -658,20 +657,20 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_NORMAL, codec_port, li, 0); + ret = graph_parse_node(priv, GRAPH_NORMAL, codec_ep, li, 0); if (ret < 0) goto err; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_port, li, 1); + ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_ep, li, 1); if (ret < 0) goto err; - graph_link_init(priv, cpu_port, li, 1); + graph_link_init(priv, cpu_ep, li, 1); err: - of_node_put(codec_port); + of_node_put(codec_ep); of_node_put(cpu_ep); return ret; @@ -684,7 +683,6 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, { struct device_node *ep = port_to_endpoint(lnk); struct device_node *rep = of_graph_get_remote_endpoint(ep); - struct device_node *rport = of_graph_get_remote_port(ep); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); int is_cpu = asoc_graph_is_ports0(lnk); @@ -718,7 +716,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 1); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 1); if (ret) goto err; } else { @@ -751,7 +749,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 0); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 0); if (ret < 0) goto err; } @@ -761,11 +759,10 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, snd_soc_dai_link_set_capabilities(dai_link); - graph_link_init(priv, rport, li, is_cpu); + graph_link_init(priv, rep, li, is_cpu); err: of_node_put(ep); of_node_put(rep); - of_node_put(rport); return ret; } @@ -777,7 +774,7 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct device_node *port0, *port1, *ports; - struct device_node *codec0_port, *codec1_port; + struct device_node *codec0_ep, *codec1_ep; struct device_node *ep0, *ep1; u32 val = 0; int ret = -EINVAL; @@ -834,31 +831,31 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, ep0 = port_to_endpoint(port0); ep1 = port_to_endpoint(port1); - codec0_port = of_graph_get_remote_port(ep0); - codec1_port = of_graph_get_remote_port(ep1); + codec0_ep = of_graph_get_remote_endpoint(ep0); + codec1_ep = of_graph_get_remote_endpoint(ep1); /* * call Codec first. * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_C2C, codec1_port, li, 0); + ret = graph_parse_node(priv, GRAPH_C2C, codec1_ep, li, 0); if (ret < 0) goto err2; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_C2C, codec0_port, li, 1); + ret = graph_parse_node(priv, GRAPH_C2C, codec0_ep, li, 1); if (ret < 0) goto err2; - graph_link_init(priv, codec0_port, li, 1); + graph_link_init(priv, codec0_ep, li, 1); err2: of_node_put(ep0); of_node_put(ep1); - of_node_put(codec0_port); - of_node_put(codec1_port); + of_node_put(codec0_ep); + of_node_put(codec1_ep); err1: of_node_put(ports); of_node_put(port0); -- 2.30.2

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[PATCH v2 0/5] Updates for rawmidi / sequencer
by Takashi Iwai 14 Jan '25

14 Jan '25

The kernel 6.14 will receive a few small enhancements for rawmidi and sequencer APIs, and this patch set covers those. v1->v2: sync with the latest rawmidi API change to pass 0 to tied_device as default v1: https://lore.kernel.org/20250114085515.11880-1-tiwai@suse.de Takashi === Takashi Iwai (5): Sync UAPI asound.h with 6.14 kernel Sync UAPI asequencer.h with 6.14 kernel rawmidi: Extensions for tied device and substream inactive flag rawmidi: Make rawmidi flag bits doxygen-style comments seq: Define new events for UMP EP/FB change notifications include/local.h | 1 + include/rawmidi.h | 4 +++- include/seq_event.h | 12 ++++++++++++ include/sound/uapi/asequencer.h | 12 +++++++++++- include/sound/uapi/asound.h | 8 ++++++-- src/rawmidi/rawmidi.c | 16 ++++++++++++++++ 6 files changed, 49 insertions(+), 4 deletions(-) -- 2.43.0

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[PATCH alsa-lib 0/5] Updates for rawmidi / sequencer
by Takashi Iwai 14 Jan '25

14 Jan '25

The kernel 6.14 will receive a few small enhancements for rawmidi and sequencer APIs, and this patch set covers those. Takashi === Takashi Iwai (5): Sync UAPI asound.h with 6.14 kernel Sync UAPI asequencer.h with 6.14 kernel rawmidi: Extensions for tied device and substream inactive flag rawmidi: Make rawmidi flag bits doxygen-style comments seq: Define new events for UMP EP/FB change notifications include/local.h | 1 + include/rawmidi.h | 4 +++- include/seq_event.h | 12 ++++++++++++ include/sound/uapi/asequencer.h | 12 +++++++++++- include/sound/uapi/asound.h | 8 ++++++-- src/rawmidi/rawmidi.c | 22 +++++++++++++++++++++- 6 files changed, 54 insertions(+), 5 deletions(-) -- 2.43.0

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rt722: change output volume of headphone to 0dB
by GitHub pull_request - opened 14 Jan '25

14 Jan '25

alsa-project/alsa-ucm-conf pull request #496 was opened from shumingfan: rt722: change output volume of headphone to 0dB Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/496 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/496.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2 1/1] ASoC: codecs: nau8824: fix max volume for Speaker output
by Maxim Kochetkov 13 Jan '25

13 Jan '25

There is no audio output if Speaker volume is set above 25. According to datasheet Rev 2.5 maximum allowed value for the Speaker output is 0b11001 (25) 0x6D CLASSD_GAIN_1/ 0x6E CLASSD_GAIN_2: Left/Right Channel Class-D Driver Gain For DAC Left/Right Input (Step size is 1dB.) 00000 = Mute (DEFAULT) 00001 = 0dB 00002 = 1dB ... 11000 = 23dB 11001 = 24dB So adjust this value in accordance with the datasheet. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/codecs/nau8824.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 12540397fd4d..5aaf8c496300 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -368,13 +368,13 @@ static const struct snd_kcontrol_new nau8824_snd_controls[] = { SOC_ENUM("DAC Oversampling Rate", nau8824_dac_oversampl_enum), SOC_SINGLE_TLV("Speaker Right DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Right DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Headphone Right DACR Volume", NAU8824_REG_ATT_PORT0, 8, 0x1f, 0, hp_vol_tlv), -- 2.45.2

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speaker-test.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 11 Jan '25

11 Jan '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' <file>" to find trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:183: misuse, warning: .BR is for at least 2 arguments, got 1 Use macro '.B' for one argument or split argument. * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.6-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.40-4 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5-2+b1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5-2+b1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

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[GIT PULL] ASoC fixes for v6.13-rc6
by Mark Brown 10 Jan '25

10 Jan '25

The following changes since commit 32c9c06adb5b157ef259233775a063a43746d699: ASoC: mediatek: disable buffer pre-allocation (2024-12-19 11:15:09 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc6 for you to fetch changes up to 139fa599cea0fd9d38e00246ea9f79af6c59acbd: ASoC: rsnd: check rsnd_adg_clk_enable() return value (2025-01-09 13:17:02 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A collection of device specific fixes that came in over the holidays, plus a MAINTAINERS update and some documentation to help users debug problems with some of the Cirrus CODECs found in modern laptops. ---------------------------------------------------------------- Charles Keepax (3): ASoC: wm8994: Add depends on MFD core ASoC: samsung: Add missing selects for MFD_WM8994 ASoC: samsung: Add missing depends on I2C Kuninori Morimoto (2): MAINTAINERS: add missing maintainers for Simple Audio Card ASoC: rsnd: check rsnd_adg_clk_enable() return value Maciej Strozek (1): ASoC: cs42l43: Add codec force suspend/resume ops Marian Postevca (1): ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK Richard Fitzgerald (2): ALSA: doc: cs35l56: Add information about Cirrus Logic CS35L54/56/57 ALSA: doc: Add codecs/index.rst to top-level index Shenghao Ding (1): ASoC: tas2781: Fix occasional calibration failture Zhang Yi (1): ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise Documentation/sound/codecs/cs35l56.rst | 292 +++++++++++++++++++++++++++++++++ Documentation/sound/codecs/index.rst | 9 + Documentation/sound/index.rst | 1 + MAINTAINERS | 2 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cs42l43.c | 1 + sound/soc/codecs/es8316.c | 10 +- sound/soc/codecs/es8326.c | 23 ++- sound/soc/codecs/tas2781-i2c.c | 2 +- sound/soc/renesas/rcar/adg.c | 28 +++- sound/soc/renesas/rcar/core.c | 4 +- sound/soc/renesas/rcar/rsnd.h | 2 +- sound/soc/samsung/Kconfig | 6 +- 13 files changed, 364 insertions(+), 17 deletions(-) create mode 100644 Documentation/sound/codecs/cs35l56.rst create mode 100644 Documentation/sound/codecs/index.rst

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