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

[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

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[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

4 4

[PATCH 0/4] ASoC: tas*: Fix up GPIO usage
by Linus Walleij 04 Jul '24

04 Jul '24

The TI TAS drivers use some legacy GPIO code and headers, this series fixes it up. The TAS2781 is a special case since it adds a handful of lines of deviating code to reconfigure a GPIO line for IRQ mode and then never actually use the IRQ obtained in the code. Is the line used by autonomous hardware? I'm puzzled by this. Anyways the patch suggest how to solve this properly by fixing the parent irqchip and I'm happy to help. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Linus Walleij (4): ASoC: tas5086: Convert to GPIO descriptors ASoC: tas2781-i2c: Drop weird GPIO code ASoC: tas2781-i2c: Get the right GPIO line ASoC: tas*: Drop unused GPIO includes include/sound/tas2781.h | 7 +------ sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/codecs/tas2552.c | 2 -- sound/soc/codecs/tas2764.c | 2 -- sound/soc/codecs/tas2770.c | 2 -- sound/soc/codecs/tas2780.c | 2 -- sound/soc/codecs/tas2781-comlib.c | 3 --- sound/soc/codecs/tas2781-fmwlib.c | 1 - sound/soc/codecs/tas2781-i2c.c | 26 ++++---------------------- sound/soc/codecs/tas5086.c | 27 ++++++++++++--------------- 10 files changed, 18 insertions(+), 56 deletions(-) --- base-commit: 1613e604df0cd359cf2a7fbd9be7a0bcfacfabd0 change-id: 20240701-asoc-tas-gpios-5c051d80d768 Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

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[PATCH v3 0/6] ASoC/SOF/PCI/Intel: add PantherLake support
by Pierre-Louis Bossart 04 Jul '24

04 Jul '24

Add initial support for the PantherLake platform, and initial ACPI configurations. This patchset depends on the first patch of "[PATCH 0/3] ALSA/PCI: add PantherLake audio support". To apply this on the ASoC tree, a merge of Takashi's for-next branch is required. v3: add configuration with rt722 on Link3 v2: changed order of patches to fix git bisect error reported by Intel kbuild bot https://lore.kernel.org/oe-kbuild-all/202406131144.L6gW0I47-lkp@intel.com/ Bard Liao (3): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support ASoC: Intel: soc-acpi-intel-ptl-match: add rt722 l3 adr Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Pierre-Louis Bossart (2): ASoC: Intel: soc-acpi: add PTL match tables ASoC: SOF: Intel: add initial support for PTL include/sound/soc-acpi-intel-match.h | 2 + sound/soc/intel/common/Makefile | 1 + .../intel/common/soc-acpi-intel-ptl-match.c | 145 ++++++++++++++++++ sound/soc/sof/intel/Kconfig | 17 ++ sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 ++++ sound/soc/sof/intel/mtl.c | 16 +- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 ++++++++++ sound/soc/sof/intel/shim.h | 1 + 12 files changed, 290 insertions(+), 2 deletions(-) create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c create mode 100644 sound/soc/sof/intel/pci-ptl.c -- 2.43.0

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USB-Audio: Add support for Solid State Labs SSL 2+
by GitHub pull_request - opened 04 Jul '24

04 Jul '24

alsa-project/alsa-ucm-conf pull request #429 was opened from kescherCode: This adds support for the [Solid State Labs SSL 2+](https://www.solidstatelogic.com/products/ssl2-plus). Note that this *is* a distinct device to the SSL 2, support for which was added in #377. The most notable difference is that this device has two additional outputs (physically four, but two of those are mirrored). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/429 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/429.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] slimbus: Fix struct and documentation alignment in stream.c
by Amit Vadhavana 03 Jul '24

03 Jul '24

The placement of the `segdist_codes` array documentation was corrected to conform with kernel documentation guidelines. The `@segdist_codes` was placed incorrectly within the struct `segdist_code` documentation block, which led to a potential misinterpretation of the code structure. The `segdist_codes` array documentation was moved outside the struct block, and a separate comment block was provided for it. This change ensures that clarity and proper alignment with kernel documentation standards are maintained. A kernel-doc warning was addressed: ./drivers/slimbus/stream.c:49: warning: Excess struct member 'segdist_codes' description in 'segdist_code' Signed-off-by: Amit Vadhavana <av2082000(a)gmail.com> --- Changes in v2: - Removed `static const` keyword from `segdist_code` structure declaration. drivers/slimbus/stream.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index 1d6b38657917..863ab3075d7e 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -18,15 +18,17 @@ * and the first slot of the next consecutive Segment. * @segdist_code: Segment Distribution Code SD[11:0] * @seg_offset_mask: Segment offset mask in SD[11:0] - * @segdist_codes: List of all possible Segmet Distribution codes. */ -static const struct segdist_code { +struct segdist_code { int ratem; int seg_interval; int segdist_code; u32 seg_offset_mask; -} segdist_codes[] = { +}; + +/* segdist_codes - List of all possible Segment Distribution codes. */ +static const struct segdist_code segdist_codes[] = { {1, 1536, 0x200, 0xdff}, {2, 768, 0x100, 0xcff}, {4, 384, 0x080, 0xc7f}, -- 2.25.1

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[PATCH 0/2] ASoC: cs35l56: Set correct upper volume limit
by Richard Fitzgerald 03 Jul '24

03 Jul '24

These two commits set the upper limit of the Speaker Volume control to +12dB instead of +100dB. This should have been a simple 1-line change to the #define in the header file, but only the HDA cs35l56 driver is using this define. The ASoC cs35l56 driver was using hardcoded numbers instead of the header defines. So the first commit changes the ASoC driver to use the #defined constants. The second commit corrects the value of the constant. Richard Fitzgerald (2): ASoC: cs35l56: Use header defines for Speaker Volume control definition ASoC: cs35l56: Limit Speaker Volume to +12dB maximum include/sound/cs35l56.h | 2 +- sound/soc/codecs/cs35l56.c | 6 +++++- 2 files changed, 6 insertions(+), 2 deletions(-) -- 2.39.2

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[PATCH] ASoC: fsl_xcvr: Improve suspend/resume flow in fsl_xcvr_trigger()
by Chancel Liu 03 Jul '24

03 Jul '24

In the current flow all interrupts are disabled in runtime suspend phase. However interrupts enablement only exists in fsl_xcvr_prepare(). After resume fsl_xcvr_prepare() may not be called so it will cause all interrupts still disabled even if resume from suspend. Interrupts should be explictily enabled after resume. Also, DPATH reset setting only exists in fsl_xcvr_prepare(). After resume from suspend DPATH should be reset otherwise there'll be channel swap issue. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 43 +++++++++++++++++++++------------------- 1 file changed, 23 insertions(+), 20 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 337da46a2f90..bf9a4e90978e 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -529,16 +529,6 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, break; } - ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, - FSL_XCVR_IRQ_EARC_ALL, FSL_XCVR_IRQ_EARC_ALL); - if (ret < 0) { - dev_err(dai->dev, "Error while setting IER0: %d\n", ret); - return ret; - } - - /* set DPATH RESET */ - m_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - v_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, m_ctl, v_ctl); if (ret < 0) { dev_err(dai->dev, "Error while setting EXT_CTRL: %d\n", ret); @@ -679,6 +669,15 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* set DPATH RESET */ + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, + FSL_XCVR_EXT_CTRL_DPTH_RESET(tx), + FSL_XCVR_EXT_CTRL_DPTH_RESET(tx)); + if (ret < 0) { + dev_err(dai->dev, "Failed to set DPATH RESET: %d\n", ret); + return ret; + } + if (tx) { switch (xcvr->mode) { case FSL_XCVR_MODE_EARC: @@ -711,6 +710,13 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, + FSL_XCVR_IRQ_EARC_ALL, FSL_XCVR_IRQ_EARC_ALL); + if (ret < 0) { + dev_err(dai->dev, "Error while setting IER0: %d\n", ret); + return ret; + } + /* clear DPATH RESET */ ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, FSL_XCVR_EXT_CTRL_DPTH_RESET(tx), @@ -733,6 +739,13 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, + FSL_XCVR_IRQ_EARC_ALL, 0); + if (ret < 0) { + dev_err(dai->dev, "Failed to clear IER0: %d\n", ret); + return ret; + } + if (tx) { switch (xcvr->mode) { case FSL_XCVR_MODE_SPDIF: @@ -1411,16 +1424,6 @@ static int fsl_xcvr_runtime_suspend(struct device *dev) struct fsl_xcvr *xcvr = dev_get_drvdata(dev); int ret; - /* - * Clear interrupts, when streams starts or resumes after - * suspend, interrupts are enabled in prepare(), so no need - * to enable interrupts in resume(). - */ - ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, - FSL_XCVR_IRQ_EARC_ALL, 0); - if (ret < 0) - dev_err(dev, "Failed to clear IER0: %d\n", ret); - if (!xcvr->soc_data->spdif_only) { /* Assert M0+ reset */ ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, -- 2.43.0

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[PATCH 0/9] ASoC: Constify struct regmap_config
by Javier Carrasco 03 Jul '24

03 Jul '24

This series adds the const modifier to the remaining regmap_config structs under sound/soc that are effectively used as const (i.e., only read after their declaration), but kept as writtable data. Signed-off-by: Javier Carrasco <javier.carrasco.cruz(a)gmail.com> --- Javier Carrasco (9): ASoC: cs35l34: Constify struct regmap_config ASoC: cs35l35: Constify struct regmap_config ASoC: cs35l36: Constify struct regmap_config ASoC: cs53l30: Constify struct regmap_config ASoC: jz4760: Constify struct regmap_config ASoC: jz4770: Constify struct regmap_config ASoC: wsa881x: Constify struct regmap_config ASoC: wsa883x: Constify struct regmap_config ASoC: wsa884x: Constify struct regmap_config sound/soc/codecs/cs35l34.c | 2 +- sound/soc/codecs/cs35l35.c | 2 +- sound/soc/codecs/cs35l36.c | 2 +- sound/soc/codecs/cs53l30.c | 2 +- sound/soc/codecs/jz4760.c | 2 +- sound/soc/codecs/jz4770.c | 2 +- sound/soc/codecs/wsa881x.c | 2 +- sound/soc/codecs/wsa883x.c | 2 +- sound/soc/codecs/wsa884x.c | 2 +- 9 files changed, 9 insertions(+), 9 deletions(-) --- base-commit: 0b58e108042b0ed28a71cd7edf5175999955b233 change-id: 20240703-sound-const-regmap_config-1d4d56b7e5ee Best regards, -- Javier Carrasco <javier.carrasco.cruz(a)gmail.com>

2 10

[PATCH v3 0/8] iio: adc: dfsdm: add scaling support
by Olivier Moysan 03 Jul '24

03 Jul '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Olivier Moysan (8): iio: add read scale and offset services to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add backend support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 24 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 153 ++++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 298 +++++++++++++++--- drivers/iio/industrialio-backend.c | 111 ++++++- include/linux/iio/backend.h | 11 +- 7 files changed, 626 insertions(+), 65 deletions(-) base-commit: 529d2e1900642eba6df28307e26e19793e227546 -- 2.25.1

1 1

[PATCH] ASoC: tas2781: Use of_property_read_reg()
by Rob Herring (Arm) 03 Jul '24

03 Jul '24

Replace the open-coded parsing of "reg" with of_property_read_reg(). The #ifdef is also easily replaced with IS_ENABLED(). Signed-off-by: Rob Herring (Arm) <robh(a)kernel.org> --- sound/soc/codecs/tas2781-i2c.c | 34 +++++++++++----------------------- 1 file changed, 11 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9350972dfefe..746e6964bfb3 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -21,6 +21,7 @@ #include <linux/interrupt.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_address.h> #include <linux/of_gpio.h> #include <linux/of_irq.h> #include <linux/regmap.h> @@ -622,33 +623,20 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) tas_priv->irq_info.irq_gpio = acpi_dev_gpio_irq_get(ACPI_COMPANION(&client->dev), 0); - } else { + } else if (IS_ENABLED(CONFIG_OF)) { struct device_node *np = tas_priv->dev->of_node; -#ifdef CONFIG_OF - const __be32 *reg, *reg_end; - int len, sw, aw; - - aw = of_n_addr_cells(np); - sw = of_n_size_cells(np); - if (sw == 0) { - reg = (const __be32 *)of_get_property(np, - "reg", &len); - reg_end = reg + len/sizeof(*reg); - ndev = 0; - do { - dev_addrs[ndev] = of_read_number(reg, aw); - reg += aw; - ndev++; - } while (reg < reg_end); - } else { - ndev = 1; - dev_addrs[0] = client->addr; + u64 addr; + + for (i = 0; i < TASDEVICE_MAX_CHANNELS; i++) { + if (of_property_read_reg(np, i, &addr, NULL)) + break; + dev_addrs[ndev++] = addr; } -#else + + tas_priv->irq_info.irq_gpio = of_irq_get(np, 0); + } else { ndev = 1; dev_addrs[0] = client->addr; -#endif - tas_priv->irq_info.irq_gpio = of_irq_get(np, 0); } tas_priv->ndev = ndev; for (i = 0; i < ndev; i++) -- 2.43.0

2 1

[PATCH v2 0/8] iio: adc: dfsdm: add scaling support
by Olivier Moysan 03 Jul '24

03 Jul '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve the voltage information from the backend. It is based on IIO framework read_raw API. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Conor, in this v2, I left the SD modulator driver & binding unchanged, regarding the naming issue you raised previously. The problem here is that we have two versions of the generic sigma delta modulator driver: one for legacy support and a new one to support new binding. Maybe an alternate, is to rename former sd modulator as "legacy" or something similar. I will address this point in a v3 if necessary. Olivier Moysan (8): iio: add read raw service to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add sigma delta modulator backend iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add sd modulator generic iio backend iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sd-modulator-backend.yaml | 39 +++ .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 157 ++++++++- drivers/iio/adc/Kconfig | 11 + drivers/iio/adc/Makefile | 1 + drivers/iio/adc/sd_adc_backend.c | 117 +++++++ drivers/iio/adc/stm32-dfsdm-adc.c | 302 +++++++++++++++--- drivers/iio/industrialio-backend.c | 108 +++++-- include/linux/iio/backend.h | 10 +- 8 files changed, 679 insertions(+), 66 deletions(-) create mode 100644 Documentation/devicetree/bindings/iio/adc/sd-modulator-backend.yaml create mode 100644 drivers/iio/adc/sd_adc_backend.c base-commit: 2dfa1b7bfc07e58acb9f9eaa8c871f37189dbfee -- 2.25.1

2 3

[PATCH v1] ASoc: tas2781: Add name_prefix as the prefix name of DSP firmwares and calibrated data files
by Shenghao Ding 03 Jul '24

03 Jul '24

Add name_prefix as the prefix name of DSP firmwares and calibrated data files which stored speaker calibrated impedance. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 4d1a0d836e77..cc765d45c6b5 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -394,8 +394,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; - scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", - tas_priv->dev_name); + if (tas_priv->name_prefix) + scnprintf(tas_priv->rca_binaryname, 64, "%s-%s_coef.bin", + tas_priv->name_prefix, tas_priv->dev_name); + else + scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", + tas_priv->dev_name); ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", @@ -418,8 +422,15 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * calibrated data inside algo. */ for (i = 0; i < tas_priv->ndev; i++) { - scnprintf(tas_priv->cal_binaryname[i], 64, "%s_cal_0x%02x.bin", - tas_priv->dev_name, tas_priv->tasdevice[i].dev_addr); + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); ret = tas2781_load_calibration(tas_priv, tas_priv->cal_binaryname[i], i); if (ret != 0) -- 2.34.1

3 5

[PATCH v3 0/2] Add master clock handling for nau8824
by Maxim Kochetkov 02 Jul '24

02 Jul '24

nau8824 has external MCLK pin. So add enable/disable external clock management. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- Changes in v3: - move bindings before their users - use devm_clk_get_optional() - let clock API care about NULL ptr Changes in v2: - move the devm_clk_get() to the nau8824_read_device() Maxim Kochetkov (2): ASoC: dt-bindings: nau8824: Add master clock handling ASoC: codecs: nau8824: Add master clock handling .../bindings/sound/nuvoton,nau8824.yaml | 8 +++++++ sound/soc/codecs/nau8824.c | 21 +++++++++++++++++-- sound/soc/codecs/nau8824.h | 1 + 3 files changed, 28 insertions(+), 2 deletions(-) -- 2.45.2

2 3

[PATCH] firmware: cs_dsp: Don't allow writes to read-only controls
by Richard Fitzgerald 02 Jul '24

02 Jul '24

Add a check to cs_dsp_coeff_write_ctrl() to abort if the control is not writeable. The cs_dsp code originated as an ASoC driver (wm_adsp) where all controls were exported as ALSA controls. It relied on ALSA to enforce the read-only permission. Now that the code has been separated from ALSA/ASoC it must perform its own permission check. This isn't currently causing any problems so there shouldn't be any need to backport this. If the client of cs_dsp exposes the control as an ALSA control, it should set permissions on that ALSA control to protect it. The few uses of cs_dsp_coeff_write_ctrl() inside drivers are for writable controls. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/firmware/cirrus/cs_dsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 16484ab9b09d..be8747d227db 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -796,6 +796,9 @@ int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl, lockdep_assert_held(&ctl->dsp->pwr_lock); + if (ctl->flags && !(ctl->flags & WMFW_CTL_FLAG_WRITEABLE)) + return -EPERM; + if (len + off * sizeof(u32) > ctl->len) return -EINVAL; -- 2.39.2

2 1

[PATCH 0/3] ASoC: cs35l56: Remove obsolete and redundant code
by Richard Fitzgerald 02 Jul '24

02 Jul '24

These commits remove various code that is either no longer needed, or is redundant. Richard Fitzgerald (3): ASoC: cs35l56: Revert support for dual-ownership of ASP registers ASoC: cs35l56: Remove support for A1 silicon ASoC: cs35l56: Remove redundant clearing of clk_stop_mode1 include/sound/cs35l56.h | 11 +- sound/soc/codecs/cs35l56-sdw.c | 75 ----------- sound/soc/codecs/cs35l56-shared.c | 122 ++++-------------- sound/soc/codecs/cs35l56.c | 205 ++---------------------------- sound/soc/codecs/cs35l56.h | 2 - 5 files changed, 45 insertions(+), 370 deletions(-) -- 2.39.2

2 4

[PATCH] ASoC: fsl: lpc3xxx-i2s: Include bitfield.h for FIELD_PREP
by Nathan Chancellor 02 Jul '24

02 Jul '24

bitfield.h is not explicitly included but it is required for FIELD_PREP to be expanded by the preprocessor. If it is not implicitly included, there will be a compiler error (as seen with ARCH=hexagon allmodconfig): sound/soc/fsl/lpc3xxx-i2s.c:169:10: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration] 169 | tmp |= LPC3XXX_I2S_WW8 | LPC3XXX_I2S_WS_HP(LPC3XXX_I2S_WW8_HP); | ^ sound/soc/fsl/lpc3xxx-i2s.h:42:30: note: expanded from macro 'LPC3XXX_I2S_WW8' 42 | #define LPC3XXX_I2S_WW8 FIELD_PREP(0x3, 0) /* Word width is 8bit */ | ^ sound/soc/fsl/lpc3xxx-i2s.c:205:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration] 205 | LPC3XXX_I2S_DMA1_TX_EN | LPC3XXX_I2S_DMA0_TX_DEPTH(4)); | ^ sound/soc/fsl/lpc3xxx-i2s.h:65:38: note: expanded from macro 'LPC3XXX_I2S_DMA0_TX_DEPTH' 65 | #define LPC3XXX_I2S_DMA0_TX_DEPTH(s) FIELD_PREP(0xF0000, s) /* Set the DMA1 TX Request level */ | ^ sound/soc/fsl/lpc3xxx-i2s.c:210:34: error: call to undeclared function 'FIELD_PREP'; ISO C99 and later do not support implicit function declarations [-Wimplicit-function-declaration] 210 | LPC3XXX_I2S_DMA0_RX_EN | LPC3XXX_I2S_DMA1_RX_DEPTH(4)); | ^ sound/soc/fsl/lpc3xxx-i2s.h:70:38: note: expanded from macro 'LPC3XXX_I2S_DMA1_RX_DEPTH' 70 | #define LPC3XXX_I2S_DMA1_RX_DEPTH(s) FIELD_PREP(0x700, s) /* Set the DMA1 RX Request level */ | ^ Include bitfield.h explicitly, so that FIELD_PREP is always expanded, clearing up the compiler error. Fixes: 0959de657a10 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/fsl/lpc3xxx-i2s.h | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/lpc3xxx-i2s.h b/sound/soc/fsl/lpc3xxx-i2s.h index eec755448478..b6657853017a 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.h +++ b/sound/soc/fsl/lpc3xxx-i2s.h @@ -9,6 +9,7 @@ #ifndef __SOUND_SOC_LPC3XXX_I2S_H #define __SOUND_SOC_LPC3XXX_I2S_H +#include <linux/bitfield.h> #include <linux/types.h> #include <linux/regmap.h> --- base-commit: 55f0a1fa6ea4e339c797e9a3292ca0caa4ab3885 change-id: 20240701-lpc32xx-asoc-fix-include-for-field_prep-140db845a209 Best regards, -- Nathan Chancellor <nathan(a)kernel.org>

2 1

[PATCH 1/2] ASoC: codecs: lpass-rx-macro: Fix using NULL pointer in probe() dev_err
by Krzysztof Kozlowski 02 Jul '24

02 Jul '24

The 'rx->dev' is assigned closer to the end of the probe() function, so the dev_err() must not use it - it is still NULL at this point. Instead there is already a local 'dev' variable. Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 08308a63d1b7..8d4a61f7debc 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -3843,7 +3843,7 @@ static int rx_macro_probe(struct platform_device *pdev) rx_2_5_defaults, sizeof(rx_2_5_defaults)); break; default: - dev_err(rx->dev, "Unsupported Codec version (%d)\n", rx->codec_version); + dev_err(dev, "Unsupported Codec version (%d)\n", rx->codec_version); ret = -EINVAL; goto err; } -- 2.43.0

3 4

[PATCH] drivers:soundwire: qcom: cleanup port maask calculations
by Srinivas Kandagatla 02 Jul '24

02 Jul '24

Cleanup the port map calculations, existing masks of having separate masks for in and out ports is not really required. Having a single mask for all the ports in the controller is simple and cuts of some unnecessary code. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 22 ++++++---------------- 1 file changed, 6 insertions(+), 16 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index ce5cf3ecceb5..aed57002fd0e 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -197,8 +197,7 @@ struct qcom_swrm_ctrl { int num_dout_ports; int cols_index; int rows_index; - unsigned long dout_port_mask; - unsigned long din_port_mask; + unsigned long port_mask; u32 intr_mask; u8 rcmd_id; u8 wcmd_id; @@ -1146,11 +1145,7 @@ static void qcom_swrm_stream_free_ports(struct qcom_swrm_ctrl *ctrl, mutex_lock(&ctrl->port_lock); list_for_each_entry(m_rt, &stream->master_list, stream_node) { - if (m_rt->direction == SDW_DATA_DIR_RX) - port_mask = &ctrl->dout_port_mask; - else - port_mask = &ctrl->din_port_mask; - + port_mask = &ctrl->port_mask; list_for_each_entry(p_rt, &m_rt->port_list, port_node) clear_bit(p_rt->num, port_mask); } @@ -1195,13 +1190,9 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, if (ctrl->bus.id != m_rt->bus->id) continue; - if (m_rt->direction == SDW_DATA_DIR_RX) { - maxport = ctrl->num_dout_ports; - port_mask = &ctrl->dout_port_mask; - } else { - maxport = ctrl->num_din_ports; - port_mask = &ctrl->din_port_mask; - } + port_mask = &ctrl->port_mask; + maxport = ctrl->num_dout_ports + ctrl->num_din_ports; + list_for_each_entry(s_rt, &m_rt->slave_rt_list, m_rt_node) { slave = s_rt->slave; @@ -1401,8 +1392,7 @@ static int qcom_swrm_get_port_config(struct qcom_swrm_ctrl *ctrl) return -EINVAL; /* Valid port numbers are from 1-14, so mask out port 0 explicitly */ - set_bit(0, &ctrl->dout_port_mask); - set_bit(0, &ctrl->din_port_mask); + set_bit(0, &ctrl->port_mask); ret = of_property_read_u8_array(np, "qcom,ports-offset1", off1, nports); --- base-commit: 1613e604df0cd359cf2a7fbd9be7a0bcfacfabd0 change-id: 20240618-soundwire-port-map-008f9dfae0f9 Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

2 1

[PATCH] ASoC: codecs: ES8326: suspend issue
by Zhang Yi 02 Jul '24

02 Jul '24

We find that we need to disable micbias for the codec to enter suspend So We modify the trigger conditions for enable_micbias and disable_micbias Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b246694ebb4f..60877116c0ef 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -805,6 +805,7 @@ static void es8326_jack_button_handler(struct work_struct *work) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2); button_to_report = 0; } + es8326_disable_micbias(es8326->component); } mutex_unlock(&es8326->lock); } @@ -878,7 +879,6 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x01); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x00); - es8326_enable_micbias(es8326->component); usleep_range(50000, 70000); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x03, 0x00); regmap_update_bits(es8326->regmap, ES8326_HPDET_TYPE, 0x10, 0x10); @@ -897,6 +897,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) dev_dbg(comp->dev, "button pressed\n"); regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); + es8326_enable_micbias(es8326->component); queue_delayed_work(system_wq, &es8326->button_press_work, 10); goto exit; } @@ -1067,6 +1068,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC_MUTE, 0x0f); regmap_write(es8326->regmap, ES8326_CLK_DIV_LRCK, 0xff); + es8326_disable_micbias(es8326->component); msleep(200); regmap_write(es8326->regmap, ES8326_INT_SOURCE, ES8326_INT_SRC_PIN9); -- 2.17.1

1 0

[Patch v4 00/10] Add audio support for LPC32XX CPUs
by Piotr Wojtaszczyk 02 Jul '24

02 Jul '24

This pach set is to bring back audio to machines with a LPC32XX CPU. The legacy LPC32XX SoC used to have audio spport in linux 2.6.27. The support was dropped due to lack of interest from mainaeners. Piotr Wojtaszczyk (10): dt-bindings: dma: pl08x: Add dma-cells description dt-bindings: dma: Add lpc32xx DMA mux binding ASoC: dt-bindings: lpc32xx: Add lpc32xx i2s DT binding ARM: dts: lpc32xx: Add missing dma and i2s properties clk: lpc32xx: initialize regmap using parent syscon dmaengine: Add dma router for pl08x in LPC32XX SoC ARM: lpc32xx: Remove pl08x platform data in favor for device tree mtd: rawnand: lpx32xx: Request DMA channels using DT entries ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs i2x: pnx: Use threaded irq to fix warning from del_timer_sync() .../devicetree/bindings/dma/arm-pl08x.yaml | 7 + .../bindings/dma/nxp,lpc3220-dmamux.yaml | 56 +++ .../bindings/sound/nxp,lpc3220-i2s.yaml | 73 ++++ MAINTAINERS | 21 + arch/arm/boot/dts/nxp/lpc/lpc32xx.dtsi | 53 ++- arch/arm/mach-lpc32xx/phy3250.c | 54 --- drivers/clk/Kconfig | 1 + drivers/clk/nxp/clk-lpc32xx.c | 10 +- drivers/dma/Kconfig | 9 + drivers/dma/Makefile | 1 + drivers/dma/lpc32xx-dmamux.c | 195 +++++++++ drivers/i2c/busses/i2c-pnx.c | 4 +- drivers/mtd/nand/raw/lpc32xx_mlc.c | 10 +- drivers/mtd/nand/raw/lpc32xx_slc.c | 10 +- sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/lpc3xxx-i2s.c | 376 ++++++++++++++++++ sound/soc/fsl/lpc3xxx-i2s.h | 79 ++++ sound/soc/fsl/lpc3xxx-pcm.c | 73 ++++ 19 files changed, 954 insertions(+), 87 deletions(-) create mode 100644 Documentation/devicetree/bindings/dma/nxp,lpc3220-dmamux.yaml create mode 100644 Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml create mode 100644 drivers/dma/lpc32xx-dmamux.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.h create mode 100644 sound/soc/fsl/lpc3xxx-pcm.c -- 2.25.1

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[PATCH] ASoC: tas5805m: demystify DSP volume control coefficients
by Felix Kaechele 02 Jul '24

02 Jul '24

The original manufacturer sample driver code uses direct DSP register writes to control digital volume on this amplifier without further context as to why it's not using the documented DIG_VOL_CTRL register. A thread in the manufacturer's forums [1] suggests this might have been done to work around volume ramping being used when controlling the volume through the DIG_VOL_CTRL register. When volume is controlled through this register, reading and/or writing any register is blocked until the volume ramping has concluded and the setpoint is reached. Additionally, the sample code uses a lookup table to map decibel values to 9.23 formatted coefficients. For posterity, add references to where this is documented and why a lookup table may be used. Signed-off-by: Felix Kaechele <felix(a)kaechele.ca> [1]: https://e2e.ti.com/support/audio-group/audio/f/audio-forum/1165952/tas5805m… --- sound/soc/codecs/tas5805m.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas5805m.c b/sound/soc/codecs/tas5805m.c index 3b53eba38a0b..59536c8b8e38 100644 --- a/sound/soc/codecs/tas5805m.c +++ b/sound/soc/codecs/tas5805m.c @@ -67,6 +67,14 @@ static const uint8_t dsp_cfg_preboot[] = { 0x00, 0x00, 0x7f, 0x00, 0x03, 0x02, }; +/* + * Lookup table for DSP volume coefficients. + * The formula uses floating point math, so a lookup table is used + * instead of computing values on the fly. + * Formula: round(10^(volume in dB/20)*2^23) + * The 9.23 format used here is documented in + * SLAA894 - "General Tuning Guide for TAS58xx Family" + */ static const uint32_t tas5805m_volume[] = { 0x0000001B, /* 0, -110dB */ 0x0000001E, /* 1, -109dB */ 0x00000021, /* 2, -108dB */ 0x00000025, /* 3, -107dB */ @@ -196,9 +204,13 @@ static void tas5805m_refresh(struct tas5805m_priv *tas5805m) regmap_write(rm, REG_BOOK, 0x8c); regmap_write(rm, REG_PAGE, 0x2a); - /* Refresh volume. The actual volume control documented in the - * datasheet doesn't seem to work correctly. This is a pair of - * DSP registers which are *not* documented in the datasheet. + /* Refresh volume. This writes the volume coefficients from + * the lookup table directly into the DSP registers. + * Digital volume control on this chip involves ramping which + * blocks register reads and writes until the desired setpoint + * is reached. + * The DSP memory maps are documented in + * SLOA263A - "TAS5805M, TAS5806M and TAS5806MD Process Flows" */ set_dsp_scale(rm, 0x24, tas5805m->vol[0]); set_dsp_scale(rm, 0x28, tas5805m->vol[1]); base-commit: 9fa5527b19b21848dfb09928ee66af1aac4a5700 -- 2.45.2

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[PATCH] ASoC: fsl: lpc3xxx-i2s: Avoid using ret uninitialized in lpc32xx_i2s_probe()
by Nathan Chancellor 01 Jul '24

01 Jul '24

clang points out that ret may be used uninitialized in lpc32xx_i2s_probe() in an error pointer path (which becomes fatal with CONFIG_WERROR): sound/soc/fsl/lpc3xxx-i2s.c:326:47: error: variable 'ret' is uninitialized when used here [-Werror,-Wuninitialized] 326 | "failed to init register map: %d\n", ret); | ^~~ sound/soc/fsl/lpc3xxx-i2s.c:310:9: note: initialize the variable 'ret' to silence this warning 310 | int ret; | ^ | = 0 1 error generated. One solution would be a small refactoring of the second parameter in dev_err_probe(), PTR_ERR(i2s_info_p->regs), to be the value of ret in the if statement. However, a nicer solution for debugging purposes, which is the point of this statement, would be to use the '%pe' specifier to symbolically print the error pointer value. Do so, which eliminates the uninitialized use of ret, clearing up the warning. Fixes: 0959de657a10 ("ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs") Signed-off-by: Nathan Chancellor <nathan(a)kernel.org> --- sound/soc/fsl/lpc3xxx-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c index 0e5b4d5202ff..af995ca081a3 100644 --- a/sound/soc/fsl/lpc3xxx-i2s.c +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -323,7 +323,7 @@ static int lpc32xx_i2s_probe(struct platform_device *pdev) i2s_info_p->regs = devm_regmap_init_mmio(dev, iomem, &lpc32xx_i2s_regconfig); if (IS_ERR(i2s_info_p->regs)) return dev_err_probe(dev, PTR_ERR(i2s_info_p->regs), - "failed to init register map: %d\n", ret); + "failed to init register map: %pe\n", i2s_info_p->regs); i2s_info_p->clk = devm_clk_get(dev, NULL); if (IS_ERR(i2s_info_p->clk)) --- base-commit: 878f4c36f9235e8a15fe0c2ecde066d92c50c8ff change-id: 20240701-lpc32xx-asoc-fix-uninitialized-ret-4700d6d556c1 Best regards, -- Nathan Chancellor <nathan(a)kernel.org>

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