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[tiwai-sound:for-next 5/6] sound/core/pcm_lib.c:547:9: warning: 'strncpy' output truncated before terminating nul copying 12 bytes from a string of the same length
by kernel test robot 05 Jul '24

05 Jul '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 5f9f982dd71b418aeba7a0b37f87312545c06df4 commit: d712c58c55d9a4b4cc88ec2e1f8cd2e3b82359b5 [5/6] ALSA: pcm: optimize and clarify stream synchronization ID API config: x86_64-defconfig (https://download.01.org/0day-ci/archive/20240705/202407050501.o5Z3bibi-lkp@…) compiler: gcc-13 (Ubuntu 13.2.0-4ubuntu3) 13.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240705/202407050501.o5Z3bibi-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/202407050501.o5Z3bibi-lkp@intel.com/ All warnings (new ones prefixed by >>): In function 'snd_pcm_set_sync_per_card', inlined from 'snd_pcm_lib_ioctl_sync_id' at sound/core/pcm_lib.c:1837:3, inlined from 'snd_pcm_lib_ioctl' at sound/core/pcm_lib.c:1863:10: >> sound/core/pcm_lib.c:547:9: warning: 'strncpy' output truncated before terminating nul copying 12 bytes from a string of the same length [-Wstringop-truncation] 547 | strncpy(params->sync + 4, id, len); | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vim +/strncpy +547 sound/core/pcm_lib.c 517 518 /** 519 * snd_pcm_set_sync_per_card - set the PCM sync id with card number 520 * @substream: the pcm substream 521 * @params: modified hardware parameters 522 * @id: identifier (max 12 bytes) 523 * @len: identifier length (max 12 bytes) 524 * 525 * Sets the PCM sync identifier for the card with zero padding. 526 * 527 * User space or any user should use this 16-byte identifier for a comparison only 528 * to check if two IDs are similar or different. Special case is the identifier 529 * containing only zeros. Interpretation for this combination is - empty (not set). 530 * The contents of the identifier should not be interpreted in any other way. 531 * 532 * The synchronization ID must be unique per clock source (usually one sound card, 533 * but multiple soundcard may use one PCM word clock source which means that they 534 * are fully synchronized). 535 * 536 * This routine composes this ID using card number in first four bytes and 537 * 12-byte additional ID. When other ID composition is used (e.g. for multiple 538 * sound cards), make sure that the composition does not clash with this 539 * composition scheme. 540 */ 541 void snd_pcm_set_sync_per_card(struct snd_pcm_substream *substream, 542 struct snd_pcm_hw_params *params, 543 const unsigned char *id, unsigned int len) 544 { 545 *(__u32 *)params->sync = cpu_to_le32(substream->pcm->card->number); 546 len = max(12, len); > 547 strncpy(params->sync + 4, id, len); 548 memset(params->sync + 4 + len, 0, 12 - len); 549 } 550 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[PATCH v3 0/6] ASoC: codecs: lpass-rx-macro: Few code cleanups
by Krzysztof Kozlowski 04 Jul '24

04 Jul '24

Hi, Changes in v3: - New patch #1 to fix clang jump warning ("ASoC: codecs: lpass-rx-macro: Simplify PDS cleanup with devm") - Link to v2: https://lore.kernel.org/r/20240628-b4-qcom-audio-lpass-codec-cleanups-v2-0-… Changes in v2: - Use cleanup.h instead of devm(), therefore not adding Dmitry's review. - New patch #5. - Link to v1: https://lore.kernel.org/r/20240627-b4-qcom-audio-lpass-codec-cleanups-v1-0-… Improve a bit the Qualcomm LPASS RX macro driver and align similar parts of code with LPASS WSA macro driver for consistency. No external dependencies. Best regards, Krzysztof --- Krzysztof Kozlowski (6): ASoC: codecs: lpass-rx-macro: Simplify PDS cleanup with devm ASoC: codecs: lpass-rx-macro: Simplify with cleanup.h ASoC: codecs: lpass-rx-macro: Keep static regmap_config as const ASoC: dapm: Use unsigned for number of widgets in snd_soc_dapm_new_controls() ASoC: codecs: lpass-rx-macro: Use unsigned for number of widgets ASoC: codecs: lpass-wsa-macro: Simplify with cleanup.h include/sound/soc-dapm.h | 2 +- sound/soc/codecs/lpass-macro-common.h | 5 +++ sound/soc/codecs/lpass-rx-macro.c | 63 +++++++++++++++++------------------ sound/soc/codecs/lpass-wsa-macro.c | 22 ++++++------ sound/soc/soc-dapm.c | 2 +- 5 files changed, 47 insertions(+), 47 deletions(-) --- base-commit: feca1ff0cd5ab7bc3990ec5a387d81d4dff88068 change-id: 20240627-b4-qcom-audio-lpass-codec-cleanups-27175f1d069f Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

2 7

[A52] Pipeware errors
by GitHub issues - opened 04 Jul '24

04 Jul '24

alsa-project/alsa-plugins issue #59 was opened from ManuLinares: https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/4085 I'm getting this errors: ``` jul 03 21:12:38 thought pipewire[63110]: ALSA lib confmisc.c:165:(snd_config_get_card) Cannot get card index for %f jul 03 21:12:38 thought pipewire[63110]: spa.alsa: 'plug:{SLAVE="a52:0,'hw:%f,3'"}': playback open failed: No such device jul 03 21:12:38 thought pipewire[63110]: mod.adapter: 0x5b856abf1c30: can't get format: No such device jul 03 21:12:38 thought pipewire[63110]: mod.adapter: usage: node.name=<string> jul 03 21:14:21 thought wireplumber[63111]: wplua: [string "alsa.lua"]:182: attempt to concatenate a nil value (local 'node_name') stack traceback: [string "alsa.lua"]:182: in function <[string "alsa.lua"]:175> ``` could this be related to the this commit? https://github.com/alsa-project/alsa-plugins/commit/dfc65f2d1f239256c201c68… Issue URL : https://github.com/alsa-project/alsa-plugins/issues/59 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v1] ASoc: tas2781: Set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563
by Shenghao Ding 04 Jul '24

04 Jul '24

Set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563 and move it into newly-created tasdevice_snd_controls, and keep the digital gain and analog gain in tas2781_snd_controls. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Rename driver_name from tas2781-codec to tasdev-codec, request from some tas2563 users since the driver supports both tas2563 and tas2781, the drvier name seemed unfair to tas2563. - Set "Speaker Force Firmware Load" as the common kcontrol for both tas27871 and tas2563 and move it into newly-created tasdevice_snd_controls. --- sound/soc/codecs/tas2781-i2c.c | 29 ++++++++++++++++++++++------- 1 file changed, 22 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index e0e630f66101..b412f8f2e5d5 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -104,7 +104,7 @@ static int tas2781_amp_putvol(struct snd_kcontrol *kcontrol, return tasdevice_amp_putvol(tas_priv, ucontrol, mc); } -static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -119,7 +119,7 @@ static int tas2781_force_fwload_get(struct snd_kcontrol *kcontrol, return 0; } -static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, +static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_component *component = @@ -140,6 +140,11 @@ static int tas2781_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static const struct snd_kcontrol_new tasdevice_snd_controls[] = { + SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, + tasdev_force_fwload_get, tasdev_force_fwload_put), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -147,8 +152,6 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Gain", TAS2781_DVC_LVL, 0, 0, 200, 1, tas2781_digital_getvol, tas2781_digital_putvol, dvc_tlv), - SOC_SINGLE_BOOL_EXT("Speaker Force Firmware Load", 0, - tas2781_force_fwload_get, tas2781_force_fwload_put), }; static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, @@ -590,6 +593,18 @@ static struct snd_soc_dai_driver tasdevice_dai_driver[] = { static int tasdevice_codec_probe(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int rc; + + if (tas_priv->chip_id == TAS2781) { + rc = snd_soc_add_component_controls(codec, + tas2781_snd_controls, + ARRAY_SIZE(tas2781_snd_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add control err rc = %d", + __func__, rc); + return rc; + } + } tas_priv->name_prefix = codec->name_prefix; return tascodec_init(tas_priv, codec, THIS_MODULE, tasdevice_fw_ready); @@ -617,8 +632,8 @@ static const struct snd_soc_component_driver soc_codec_driver_tasdevice = { .probe = tasdevice_codec_probe, .remove = tasdevice_codec_remove, - .controls = tas2781_snd_controls, - .num_controls = ARRAY_SIZE(tas2781_snd_controls), + .controls = tasdevice_snd_controls, + .num_controls = ARRAY_SIZE(tasdevice_snd_controls), .dapm_widgets = tasdevice_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), .dapm_routes = tasdevice_audio_map, @@ -759,7 +774,7 @@ MODULE_DEVICE_TABLE(acpi, tasdevice_acpi_match); static struct i2c_driver tasdevice_i2c_driver = { .driver = { - .name = "tas2781-codec", + .name = "tasdev-codec", .of_match_table = of_match_ptr(tasdevice_of_match), #ifdef CONFIG_ACPI .acpi_match_table = ACPI_PTR(tasdevice_acpi_match), -- 2.34.1

2 1

[PATCH] ASoC: SOF: Intel: hda-pcm: Limit the maximum number of periods by MAX_BDL_ENTRIES
by Pierre-Louis Bossart 04 Jul '24

04 Jul '24

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> The HDaudio specification Section 3.6.2 limits the number of BDL entries to 256. Make sure we don't allow more periods than this normative value. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-pcm.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/sof/intel/hda-pcm.c b/sound/soc/sof/intel/hda-pcm.c index 9fb8521b896b..f6e24edd7adb 100644 --- a/sound/soc/sof/intel/hda-pcm.c +++ b/sound/soc/sof/intel/hda-pcm.c @@ -258,6 +258,12 @@ int hda_dsp_pcm_open(struct snd_sof_dev *sdev, snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); + /* Limit the maximum number of periods to not exceed the BDL entries count */ + if (runtime->hw.periods_max > HDA_DSP_MAX_BDL_ENTRIES) + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIODS, + runtime->hw.periods_min, + HDA_DSP_MAX_BDL_ENTRIES); + /* Only S16 and S32 supported by HDA hardware when used without DSP */ if (sdev->dspless_mode_selected) snd_pcm_hw_constraint_mask64(substream->runtime, SNDRV_PCM_HW_PARAM_FORMAT, -- 2.43.0

2 1

[PATCH] ASoC: SOF: Intel: hda: fix null deref on system suspend entry
by Pierre-Louis Bossart 04 Jul '24

04 Jul '24

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> When system enters suspend with an active stream, SOF core calls hw_params_upon_resume(). On Intel platforms with HDA DMA used to manage the link DMA, this leads to call chain of hda_dsp_set_hw_params_upon_resume() -> hda_dsp_dais_suspend() -> hda_dai_suspend() -> hda_ipc4_post_trigger() A bug is hit in hda_dai_suspend() as hda_link_dma_cleanup() is run first, which clears hext_stream->link_substream, and then hda_ipc4_post_trigger() is called with a NULL snd_pcm_substream pointer. Fixes: 2b009fa0823c ("ASoC: SOF: Intel: hda: Unify DAI drv ops for IPC3 and IPC4") Link: https://github.com/thesofproject/linux/issues/5080 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/intel/hda-dai.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index c61d298ea6b3..1c823f9eea57 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -617,12 +617,6 @@ static int hda_dai_suspend(struct hdac_bus *bus) sdai = swidget->private; ops = sdai->platform_private; - ret = hda_link_dma_cleanup(hext_stream->link_substream, - hext_stream, - cpu_dai); - if (ret < 0) - return ret; - /* for consistency with TRIGGER_SUSPEND */ if (ops->post_trigger) { ret = ops->post_trigger(sdev, cpu_dai, @@ -631,6 +625,12 @@ static int hda_dai_suspend(struct hdac_bus *bus) if (ret < 0) return ret; } + + ret = hda_link_dma_cleanup(hext_stream->link_substream, + hext_stream, + cpu_dai); + if (ret < 0) + return ret; } } -- 2.43.0

2 1

[PATCH] ASoC: SOF: ipc4-topology: Use single token list for the copiers
by Pierre-Louis Bossart 04 Jul '24

04 Jul '24

From: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> There is no need to keep separate token list for dai and 'common' copier token list when the 'common' list is actually the aif list, the SOF_COPIER_DEEP_BUFFER_TOKENS are not applicable for buffers. We could have separate lists for all types but it is probably simpler to just use a single list for all types of copiers. Function specific tokens will be only parsed by function specific code anyways. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index 7755c01735f5..90f6856ee80c 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -3307,14 +3307,17 @@ static int sof_ipc4_link_setup(struct snd_sof_dev *sdev, struct snd_soc_dai_link return 0; } -static enum sof_tokens common_copier_token_list[] = { +/* Tokens needed for different copier variants (aif, dai and buffer) */ +static enum sof_tokens copier_token_list[] = { SOF_COMP_TOKENS, + SOF_COPIER_TOKENS, SOF_AUDIO_FMT_NUM_TOKENS, SOF_IN_AUDIO_FORMAT_TOKENS, SOF_OUT_AUDIO_FORMAT_TOKENS, - SOF_COPIER_DEEP_BUFFER_TOKENS, - SOF_COPIER_TOKENS, SOF_COMP_EXT_TOKENS, + + SOF_COPIER_DEEP_BUFFER_TOKENS, /* for AIF copier */ + SOF_DAI_TOKENS, /* for DAI copier */ }; static enum sof_tokens pipeline_token_list[] = { @@ -3322,16 +3325,6 @@ static enum sof_tokens pipeline_token_list[] = { SOF_PIPELINE_TOKENS, }; -static enum sof_tokens dai_token_list[] = { - SOF_COMP_TOKENS, - SOF_AUDIO_FMT_NUM_TOKENS, - SOF_IN_AUDIO_FORMAT_TOKENS, - SOF_OUT_AUDIO_FORMAT_TOKENS, - SOF_COPIER_TOKENS, - SOF_DAI_TOKENS, - SOF_COMP_EXT_TOKENS, -}; - static enum sof_tokens pga_token_list[] = { SOF_COMP_TOKENS, SOF_GAIN_TOKENS, @@ -3368,23 +3361,23 @@ static enum sof_tokens process_token_list[] = { static const struct sof_ipc_tplg_widget_ops tplg_ipc4_widget_ops[SND_SOC_DAPM_TYPE_COUNT] = { [snd_soc_dapm_aif_in] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, - common_copier_token_list, ARRAY_SIZE(common_copier_token_list), + copier_token_list, ARRAY_SIZE(copier_token_list), NULL, sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_aif_out] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, - common_copier_token_list, ARRAY_SIZE(common_copier_token_list), + copier_token_list, ARRAY_SIZE(copier_token_list), NULL, sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_dai_in] = {sof_ipc4_widget_setup_comp_dai, sof_ipc4_widget_free_comp_dai, - dai_token_list, ARRAY_SIZE(dai_token_list), NULL, + copier_token_list, ARRAY_SIZE(copier_token_list), NULL, sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_dai_out] = {sof_ipc4_widget_setup_comp_dai, sof_ipc4_widget_free_comp_dai, - dai_token_list, ARRAY_SIZE(dai_token_list), NULL, + copier_token_list, ARRAY_SIZE(copier_token_list), NULL, sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_buffer] = {sof_ipc4_widget_setup_pcm, sof_ipc4_widget_free_comp_pcm, - common_copier_token_list, ARRAY_SIZE(common_copier_token_list), + copier_token_list, ARRAY_SIZE(copier_token_list), NULL, sof_ipc4_prepare_copier_module, sof_ipc4_unprepare_copier_module}, [snd_soc_dapm_scheduler] = {sof_ipc4_widget_setup_comp_pipeline, -- 2.43.0

2 1

[PATCH v1] ALSA: hda: cs35l41: Fix missing Speaker ID GPIO description in _DSD
by Stefan Binding 04 Jul '24

04 Jul '24

Laptop 10431A63 contains valid _DSD, but missing Speaker ID description. Add this discription, but keep the rest of the _DSD to ensure the correct firmware and tuning is loaded for this laptop. Signed-off-by: Stefan Binding <sbinding(a)opensource.cirrus.com> --- sound/pci/hda/cs35l41_hda.c | 86 ++++++++++++++++------------ sound/pci/hda/cs35l41_hda.h | 1 + sound/pci/hda/cs35l41_hda_property.c | 15 +++++ 3 files changed, 65 insertions(+), 37 deletions(-) diff --git a/sound/pci/hda/cs35l41_hda.c b/sound/pci/hda/cs35l41_hda.c index ee9f83b737de..4b411ed8c3fe 100644 --- a/sound/pci/hda/cs35l41_hda.c +++ b/sound/pci/hda/cs35l41_hda.c @@ -1753,38 +1753,14 @@ int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int return speaker_id; } -static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, int id) +int cs35l41_hda_parse_acpi(struct cs35l41_hda *cs35l41, struct device *physdev, int id) { struct cs35l41_hw_cfg *hw_cfg = &cs35l41->hw_cfg; u32 values[HDA_MAX_COMPONENTS]; - struct acpi_device *adev; - struct device *physdev; - struct spi_device *spi; - const char *sub; char *property; size_t nval; int i, ret; - adev = acpi_dev_get_first_match_dev(hid, NULL, -1); - if (!adev) { - dev_err(cs35l41->dev, "Failed to find an ACPI device for %s\n", hid); - return -ENODEV; - } - - cs35l41->dacpi = adev; - physdev = get_device(acpi_get_first_physical_node(adev)); - - sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); - if (IS_ERR(sub)) - sub = NULL; - cs35l41->acpi_subsystem_id = sub; - - ret = cs35l41_add_dsd_properties(cs35l41, physdev, id, hid); - if (!ret) { - dev_info(cs35l41->dev, "Using extra _DSD properties, bypassing _DSD in ACPI\n"); - goto out; - } - property = "cirrus,dev-index"; ret = device_property_count_u32(physdev, property); if (ret <= 0) @@ -1816,8 +1792,9 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i /* To use the same release code for all laptop variants we can't use devm_ version of * gpiod_get here, as CLSA010* don't have a fully functional bios with an _DSD node */ - cs35l41->reset_gpio = fwnode_gpiod_get_index(acpi_fwnode_handle(adev), "reset", cs35l41->index, - GPIOD_OUT_LOW, "cs35l41-reset"); + cs35l41->reset_gpio = fwnode_gpiod_get_index(acpi_fwnode_handle(cs35l41->dacpi), "reset", + cs35l41->index, GPIOD_OUT_LOW, + "cs35l41-reset"); property = "cirrus,speaker-position"; ret = device_property_read_u32_array(physdev, property, values, nval); @@ -1873,6 +1850,51 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i hw_cfg->bst_type = CS35L41_EXT_BOOST; hw_cfg->valid = true; + + return 0; +err: + dev_err(cs35l41->dev, "Failed property %s: %d\n", property, ret); + hw_cfg->valid = false; + hw_cfg->gpio1.valid = false; + hw_cfg->gpio2.valid = false; + acpi_dev_put(cs35l41->dacpi); + + return ret; +} + +static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, int id) +{ + struct acpi_device *adev; + struct device *physdev; + struct spi_device *spi; + const char *sub; + int ret; + + adev = acpi_dev_get_first_match_dev(hid, NULL, -1); + if (!adev) { + dev_err(cs35l41->dev, "Failed to find an ACPI device for %s\n", hid); + return -ENODEV; + } + + cs35l41->dacpi = adev; + physdev = get_device(acpi_get_first_physical_node(adev)); + + sub = acpi_get_subsystem_id(ACPI_HANDLE(physdev)); + if (IS_ERR(sub)) + sub = NULL; + cs35l41->acpi_subsystem_id = sub; + + ret = cs35l41_add_dsd_properties(cs35l41, physdev, id, hid); + if (!ret) { + dev_info(cs35l41->dev, "Using extra _DSD properties, bypassing _DSD in ACPI\n"); + goto out; + } + + ret = cs35l41_hda_parse_acpi(cs35l41, physdev, id); + if (ret) { + put_device(physdev); + return ret; + } out: put_device(physdev); @@ -1888,16 +1910,6 @@ static int cs35l41_hda_read_acpi(struct cs35l41_hda *cs35l41, const char *hid, i } return 0; - -err: - dev_err(cs35l41->dev, "Failed property %s: %d\n", property, ret); - hw_cfg->valid = false; - hw_cfg->gpio1.valid = false; - hw_cfg->gpio2.valid = false; - acpi_dev_put(cs35l41->dacpi); - put_device(physdev); - - return ret; } int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int irq, diff --git a/sound/pci/hda/cs35l41_hda.h b/sound/pci/hda/cs35l41_hda.h index b0bebb778462..c730b3351589 100644 --- a/sound/pci/hda/cs35l41_hda.h +++ b/sound/pci/hda/cs35l41_hda.h @@ -104,5 +104,6 @@ int cs35l41_hda_probe(struct device *dev, const char *device_name, int id, int i struct regmap *regmap, enum control_bus control_bus); void cs35l41_hda_remove(struct device *dev); int cs35l41_get_speaker_id(struct device *dev, int amp_index, int num_amps, int fixed_gpio_id); +int cs35l41_hda_parse_acpi(struct cs35l41_hda *cs35l41, struct device *physdev, int id); #endif /*__CS35L41_HDA_H__*/ diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 51998d1c72ff..5860379a0412 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -428,6 +428,20 @@ static int lenovo_legion_no_acpi(struct cs35l41_hda *cs35l41, struct device *phy return 0; } +static int missing_speaker_id_gpio2(struct cs35l41_hda *cs35l41, struct device *physdev, int id, + const char *hid) +{ + int ret; + + ret = cs35l41_add_gpios(cs35l41, physdev, -1, 2, -1, 2); + if (ret) { + dev_err(cs35l41->dev, "Error adding GPIO mapping: %d\n", ret); + return ret; + } + + return cs35l41_hda_parse_acpi(cs35l41, physdev, id); +} + struct cs35l41_prop_model { const char *hid; const char *ssid; @@ -501,6 +515,7 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "104317F3", generic_dsd_config }, { "CSC3551", "10431863", generic_dsd_config }, { "CSC3551", "104318D3", generic_dsd_config }, + { "CSC3551", "10431A63", missing_speaker_id_gpio2 }, { "CSC3551", "10431A83", generic_dsd_config }, { "CSC3551", "10431B93", generic_dsd_config }, { "CSC3551", "10431C9F", generic_dsd_config }, -- 2.43.0

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Virtual birthday cards
by hobefaj342＠devncie.com 04 Jul '24

04 Jul '24

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[alsa-devel] unsupported variant of snd_hda_intel
by Alexandre 04 Jul '24

04 Jul '24

Hello, I have an Acer Aspire 4736G laptop. The soundcard that is shipped in this laptop is intel-hda based as shown (actually it's a realtek): cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 46 Basic sound output is correct but few feature are missing: - internal mic seems to be found but do not works. - sound output stop working when trying to send it using HDMI output - heads do not work when plugged neither when forcing pulseaudio to use it. I have an ubuntu 10.10 which is shipped with alsa 1.0.23. I have googled around and tried loading the snd-hda-intel with several parameters () as found in HD-Audio-Models.txt to no avail. cat /proc/asound/pcm 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 00-01: ALC888 Digital : ALC888 Digital : playback 1 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 Where can I begin to help implement the support for this sound card? I have very skills in developement in general but am ready to try if it is not to hard (I supose I can use previous code for similar models).. I just need a shepherd. Thanks

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