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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 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

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[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>

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[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

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