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

[PATCH v4 0/8] iio: adc: dfsdm: add scaling support
by Olivier Moysan 04 Jul '24

04 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 Changes in v4: - Remove allOf and rework examples in SD modulator binding - Wrap lines to 80 characters in IIO code 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 | 23 +- .../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 | 116 ++++++- include/linux/iio/backend.h | 14 +- 7 files changed, 633 insertions(+), 65 deletions(-) base-commit: 529d2e1900642eba6df28307e26e19793e227546 -- 2.25.1

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[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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RE: [PATCH 1/1] Fix swapped left and right channels for Lenovo ThinBook 13x Gen4 sound
by Stefan Binding (Opensource) 04 Jul '24

04 Jul '24

Reviewed-by: Stefan Binding <sbinding(a)opensource.cirrus.com> > -----Original Message----- > From: MadEnvel <envelsavinds(a)gmail.com> > Sent: Monday, July 1, 2024 9:55 PM > To: tiwai(a)suse.com; rf(a)opensource.cirrus.com > Cc: david.rhodes(a)cirrus.com; alsa-devel(a)alsa-project.org; > patches(a)opensource.cirrus.com; MadEnvel <envelsavinds(a)gmail.com> > Subject: [PATCH 1/1] Fix swapped left and right channels for Lenovo ThinBook > 13x Gen4 sound > > Signed-off-by: MadEnvel <envelsavinds(a)gmail.com> > --- > sound/pci/hda/cs35l41_hda_property.c | 4 ++-- > 1 file changed, 2 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/hda/cs35l41_hda_property.c > b/sound/pci/hda/cs35l41_hda_property.c > index 51998d1c72ff..80c816922f78 100644 > --- a/sound/pci/hda/cs35l41_hda_property.c > +++ b/sound/pci/hda/cs35l41_hda_property.c > @@ -128,8 +128,8 @@ static const struct cs35l41_config > cs35l41_config_table[] = { > { "17AA38B5", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, > 0, 1, -1, 0, 0, 0 }, > { "17AA38B6", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, > 0, 1, -1, 0, 0, 0 }, > { "17AA38B7", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, > 0, 1, -1, 0, 0, 0 }, > - { "17AA38C7", 4, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, > CS35L41_LEFT, CS35L41_RIGHT }, 0, 2, -1, 1000, 4500, 24 }, > - { "17AA38C8", 4, INTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, > CS35L41_LEFT, CS35L41_RIGHT }, 0, 2, -1, 1000, 4500, 24 }, > + { "17AA38C7", 4, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, > CS35L41_RIGHT, CS35L41_LEFT }, 0, 2, -1, 1000, 4500, 24 }, > + { "17AA38C8", 4, INTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, > CS35L41_RIGHT, CS35L41_LEFT }, 0, 2, -1, 1000, 4500, 24 }, > { "17AA38F9", 2, EXTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, > 0, 2, -1, 0, 0, 0 }, > { "17AA38FA", 2, EXTERNAL, { CS35L41_RIGHT, CS35L41_LEFT, 0, 0 }, > 0, 2, -1, 0, 0, 0 }, > {} > -- > 2.45.2

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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

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[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 v23 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 04 Jul '24

04 Jul '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Wesley Cheng (32): ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Introduce SND kcontrols to track USB offloading state ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request ASoC: usb: Rediscover USB SND devices on USB port add ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 603 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/xhci-plat.c | 2 + .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 188 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 417 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 684 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1915 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 41 files changed, 5705 insertions(+), 29 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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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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[PATCH 00/11] ASoC: Simplify code with cleanup.h
by Krzysztof Kozlowski 04 Jul '24

04 Jul '24

Allocate the memory with scoped/cleanup.h to reduce error handling (simpler error paths) and make the code a bit smaller. Best regards, Krzysztof --- Krzysztof Kozlowski (11): ASoC: codecs: audio-iio-aux: Simplify audio_iio_aux_add_dapms() with cleanup.h ASoC: codecs: audio-iio-aux: Simplify audio_iio_aux_probe() with cleanup.h ASoC: codecs: wcd9335: Simplify with cleanup.h ASoC: codecs: wcd934x: Simplify with cleanup.h ASoC: simple-card-utils: Simplify with cleanup.h ASoC: audio-graph-card: Use cleanup.h instead of devm_kfree() ASoC: audio-graph-card2: Use cleanup.h instead of devm_kfree() ASoC: simple-card: Use cleanup.h instead of devm_kfree() ASoC: ops: Simplify with cleanup.h ASoC: dapm: Simplify dapm_cnew_widget() with cleanup.h ASoC: dapm: Simplify snd_soc_dai_link_event_pre_pmu() with cleanup.h sound/soc/codecs/audio-iio-aux.c | 83 ++++++++++++----------------------- sound/soc/codecs/wcd9335.c | 22 ++++------ sound/soc/codecs/wcd934x.c | 22 ++++------ sound/soc/generic/audio-graph-card.c | 5 +-- sound/soc/generic/audio-graph-card2.c | 5 +-- sound/soc/generic/simple-card-utils.c | 13 +++--- sound/soc/generic/simple-card.c | 5 +-- sound/soc/soc-dapm.c | 47 ++++++++------------ sound/soc/soc-ops.c | 26 +++++------ 9 files changed, 87 insertions(+), 141 deletions(-) --- base-commit: cf601827c51b0496f6bb8a1865f1bf69ebd246b7 change-id: 20240703-asoc-cleanup-h-ef9cbfb67b66 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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