July 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ASoC: soc-core: protect dlc->of_node under mutex
by Kuninori Morimoto 12 Jul '23

12 Jul '23

dlc->of_node will be set on snd_soc_get_dlc(), but we want 1) protect it by mutex, 2) set only when successed. This patch do it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 11bc5250ffd0..fcc798e0b8f0 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3262,8 +3262,6 @@ int snd_soc_get_dlc(const struct of_phandle_args *args, struct snd_soc_dai_link_ struct snd_soc_component *pos; int ret = -EPROBE_DEFER; - dlc->of_node = args->np; - mutex_lock(&client_mutex); for_each_component(pos) { struct device_node *component_of_node = soc_component_to_node(pos); @@ -3317,6 +3315,10 @@ int snd_soc_get_dlc(const struct of_phandle_args *args, struct snd_soc_dai_link_ break; } + + if (ret == 0) + dlc->of_node = args->np; + mutex_unlock(&client_mutex); return ret; } -- 2.25.1

2 1

[PATCH 0/8] ASoC: Intel: avs: New boards and fixes to existing ones
by Cezary Rojewski 12 Jul '23

12 Jul '23

Series adds support for two boards: es8336 and rt5663. The former is utilized by some KBL-based tablets whereas the latter unlocks Chromebooks with rt5663 i2c codecs. As existing implementation of es8336 (es8316.c) codec driver is not prepared to cope with KBL-based platforms, couple of small, clock-related changes precede anything avs-driver related. The tail of patchset cleans up existing implementation of rt5682. Amadeusz Sławiński (2): ASoC: Intel: avs: Add rt5663 machine board ASoC: Intel: avs: Load rt5663 board on KBL-based platforms Cezary Rojewski (6): ASoC: codecs: es8316: Add support for 24 MHz MCLK ASoC: codecs: es8316: Add support for S24_3LE format ASoC: Intel: avs: Add es8336 machine board ASoC: Intel: avs: Load es8336 board on KBL-based platforms ASoC: Intel: avs: rt5682: Add missing components ASoC: Intel: avs: rt5682: Tidy up hw_params() sound/soc/codecs/es8316.c | 5 +- sound/soc/intel/avs/board_selection.c | 16 ++ sound/soc/intel/avs/boards/Kconfig | 20 ++ sound/soc/intel/avs/boards/Makefile | 4 + sound/soc/intel/avs/boards/es8336.c | 315 ++++++++++++++++++++++++++ sound/soc/intel/avs/boards/rt5663.c | 254 +++++++++++++++++++++ sound/soc/intel/avs/boards/rt5682.c | 67 +++--- 7 files changed, 652 insertions(+), 29 deletions(-) create mode 100644 sound/soc/intel/avs/boards/es8336.c create mode 100644 sound/soc/intel/avs/boards/rt5663.c -- 2.25.1

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[PATCH] ASoC: imx-pcm-rpmsg: Set PCM hardware parameters separately
by Chancel Liu 12 Jul '23

12 Jul '23

Different PCM devices may have different PCM hardware parameters. It requires PCM hardware parameters set separately if there is more than one rpmsg sound card. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/imx-pcm-rpmsg.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index 765dad607bf6..d63782b8bdef 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -228,6 +228,10 @@ static int imx_rpmsg_pcm_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { struct rpmsg_info *info = dev_get_drvdata(component->dev); + struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream); + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct fsl_rpmsg *rpmsg = dev_get_drvdata(cpu_dai->dev); + struct snd_pcm_hardware pcm_hardware; struct rpmsg_msg *msg; int ret = 0; int cmd; @@ -255,10 +259,11 @@ static int imx_rpmsg_pcm_open(struct snd_soc_component *component, info->send_message(msg, info); - imx_rpmsg_pcm_hardware.period_bytes_max = - imx_rpmsg_pcm_hardware.buffer_bytes_max / 2; + pcm_hardware = imx_rpmsg_pcm_hardware; + pcm_hardware.buffer_bytes_max = rpmsg->buffer_size; + pcm_hardware.period_bytes_max = pcm_hardware.buffer_bytes_max / 2; - snd_soc_set_runtime_hwparams(substream, &imx_rpmsg_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, &pcm_hardware); ret = snd_pcm_hw_constraint_integer(substream->runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -597,7 +602,6 @@ static int imx_rpmsg_pcm_new(struct snd_soc_component *component, if (ret) return ret; - imx_rpmsg_pcm_hardware.buffer_bytes_max = rpmsg->buffer_size; return snd_pcm_set_fixed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV_WC, pcm->card->dev, rpmsg->buffer_size); } -- 2.25.1

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[PATCH] ASoC: dt-bindings: nau8821: Convert to dtschema
by Seven Lee 12 Jul '23

12 Jul '23

Convert the NAU8821 audio CODEC bindings to DT schema. Signed-off-by: Seven Lee <wtli(a)nuvoton.com> --- .../devicetree/bindings/sound/nau8821.txt | 55 -------- .../bindings/sound/nuvoton,nau8821.yaml | 125 ++++++++++++++++++ 2 files changed, 125 insertions(+), 55 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/nau8821.txt create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml diff --git a/Documentation/devicetree/bindings/sound/nau8821.txt b/Documentation/devicetree/bindings/sound/nau8821.txt deleted file mode 100644 index 7c84e7c7327a..000000000000 --- a/Documentation/devicetree/bindings/sound/nau8821.txt +++ /dev/null @@ -1,55 +0,0 @@ -Nuvoton NAU88L21 audio codec - -This device supports I2C only. - -Required properties: - - compatible : Must be "nuvoton,nau8821" - - - reg : the I2C address of the device. This is either 0x1B (CSB=0) or 0x54 (CSB=1). - -Optional properties: - - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. - - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, - otherwise pin in high impedance state. - - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. - - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. - - - nuvoton,vref-impedance: VREF Impedance selection - 0 - Open - 1 - 25 kOhm - 2 - 125 kOhm - 3 - 2.5 kOhm - - - nuvoton,micbias-voltage: Micbias voltage level. - 0 - VDDA - 1 - VDDA - 2 - VDDA * 1.1 - 3 - VDDA * 1.2 - 4 - VDDA * 1.3 - 5 - VDDA * 1.4 - 6 - VDDA * 1.53 - 7 - VDDA * 1.53 - - - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms - - - nuvoton,dmic-clk-threshold: the ADC threshold of DMIC clock. - - nuvoton,key_enable: Headset button detection switch. - -Example: - - headset: nau8821@1b { - compatible = "nuvoton,nau8821"; - reg = <0x1b>; - interrupt-parent = <&gpio>; - interrupts = <23 IRQ_TYPE_LEVEL_LOW>; - nuvoton,jkdet-enable; - nuvoton,jkdet-pull-enable; - nuvoton,jkdet-pull-up; - nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; - nuvoton,vref-impedance = <2>; - nuvoton,micbias-voltage = <6>; - nuvoton,jack-insert-debounce = <7>; - nuvoton,jack-eject-debounce = <7>; - nuvoton,dmic-clk-threshold = 3072000; - }; diff --git a/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml new file mode 100644 index 000000000000..fc2f4ce4db88 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nuvoton,nau8821.yaml @@ -0,0 +1,125 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nuvoton,nau8821.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NAU88L21 audio codec + +maintainers: + - Seven Lee <wtli(a)nuvoton.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: nuvoton,nau8821 + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + nuvoton,jkdet-enable: + description: Enable jack detection via JKDET pin. + type: boolean + + nuvoton,jkdet-pull-enable: + description: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + type: boolean + + nuvoton,jkdet-pull-up: + description: Pull-up JKDET pin. If set then JKDET pin is pull up, + otherwise pull down. + type: boolean + + nuvoton,key-enable: + description: handles key press detection. + type: boolean + + nuvoton,jkdet-polarity: + description: JKDET pin polarity. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # active high + - 1 # active low + default: 1 + + nuvoton,micbias-voltage: + description: MICBIAS output level select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # VDDA + - 1 # VDDA * 1 + - 2 # VDDA * 1.1 + - 3 # VDDA * 1.2 + - 4 # VDDA * 1.3 + - 5 # VDDA * 1.4 + - 6 # VDDA * 1.53 + - 7 # VDDA * 1.53 + default: 6 + + nuvoton,vref-impedance: + description: VMID Tie-off impedance select. + $ref: /schemas/types.yaml#/definitions/uint32 + enum: + - 0 # open + - 1 # 25KOhms + - 2 # 125KOhms + - 3 # 2.5KOhms + default: 2 + + nuvoton,jack-insert-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 7 + + nuvoton,jack-eject-debounce: + description: number from 0 to 7 that sets debounce time to 2^(n+2)ms. + $ref: /schemas/types.yaml#/definitions/uint32 + maximum: 7 + default: 0 + + nuvoton,dmic-clk-threshold: + description: DMIC clock speed expected value. Unit is Hz. + $ref: /schemas/types.yaml#/definitions/uint32 + default: 3072000 + + '#sound-dai-cells': + const: 0 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + #include <dt-bindings/interrupt-controller/irq.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + codec@1b { + compatible = "nuvoton,nau8821"; + reg = <0x1b>; + interrupt-parent = <&gpio>; + interrupts = <23 IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,key-enable; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,micbias-voltage = <6>; + nuvoton,vref-impedance = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <0>; + nuvoton,dmic-clk-threshold = <3072000>; + #sound-dai-cells = <0>; + }; + }; -- 2.25.1

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[PATCH v6 00/13] Add support for IIO devices in ASoC
by Herve Codina 12 Jul '23

12 Jul '23

Several weeks ago, I sent a series [1] for adding a potentiometer as an auxiliary device in ASoC. The feedback was that the potentiometer should be directly handled in IIO (as other potentiometers) and something more generic should be present in ASoC in order to have a binding to import some IIO devices into sound cards. The series related to the IIO potentiometer device is already applied. This series introduces audio-iio-aux. Its goal is to offer the binding between IIO and ASoC. It exposes attached IIO devices as ASoC auxiliary devices and allows to control them through mixer controls. On my system, the IIO device is a potentiometer and it is present in an amplifier design present in the audio path. Compare to the previous iteration https://lore.kernel.org/linux-kernel/20230615152631.224529-1-herve.codina@b… This v6 series mainly: - Fixes comments - Does some minor code modification Best regards, Hervé [1] https://lore.kernel.org/linux-kernel/20230203111422.142479-1-herve.codina@b… [2] https://lore.kernel.org/linux-kernel/20230421085245.302169-1-herve.codina@b… Changes v5 -> v6 - Patches 3, 4, 11, 13 No changes - Patch 1 Add 'Reviewed-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>' Add 'Reviewed-by: Rob Herring <robh(a)kernel.org>' - Patch 2 Add 'Reviewed-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>' - Patches 5, 6 Add 'Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>' - Patch 7 Fix comments: 'buf' -> 'buff', '__unconst_typeof' -> '__unconst_integer_typeof' - Patch 8, 9, 10 Add 'Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>' - Patch 12 Set struct snd_kcontrol_new fields for the constrol variable at variable initalization. Add a missing blank line Add a comment related to min/max swapping Add 'Reviewed-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>' Changes v4 -> v5 - Patches 1, 2, 3, 4, 5, 9, 10, 11, 12, 13 No changes. - Patch 6 Fix commit log. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' - Patch 7 Fix the macros to be able to use them with: - an array defined as int *buff; - an array defined as int buff[N]; - Rework the way to "unconstify" the temporary variable to avoid issues due to integer promotion. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' - Patch 8 Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>' Changes v3 -> v4 - Patches 1, 2 No changes. - Patches 3, 4, 5 Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 6 (new in v4) Fix headers inclusion order. - Patch 7 (patch 6 in v3) Add a comment related to __must_be_array() Use __array[0] of *__array - Patch 8 (patch 7 in v3) Fix minmax.h inclusion order. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 9 (patch 8 in v3) Add 'Suggested-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 10 (patch 9 in v3) Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 11 (patch 10 in v3) Fix a typo. Add 'Suggested-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. Add 'Reviewed-by: Andy Shevchenko <andy.shevchenko(a)gmail.com>'. - Patch 12 (patch 11 in v3) Fix typos in the commit log. Fix headers inclusion order. Removed unneeded variable initialization. Replace {0} by {}. Use struct device *dev in probe(). Check an error on the snd-control-invert-range property read. - Patch 13 (patch12 in v3) No changes. Changes v2 -> v3 - Patches 1, 2 No changes. - Patch 3, 4 Add 'Acked-by: Jonathan Cameron <Jonathan.Cameron(a)huawei.com>'. - Patch 5 (new in v3) Removed the 'unused' variable and check the null pointer when used. - Patch 6 (new in v3) Introduce {min,max}_array(). - Patch 7 (new in v3) Use max_array() in iio_channel_read_max(). - Patch 8 (new in v3) Replace a FIXME comment by a TODO one. - Patch 9 (patch 5 in v2) Removed the 'unused' variable and check the null pointer when used. Use min_array(). Remplace a FIXME comment by a TODO one. - Patch 10 (patch 6 in v2) Convert existing macros to return a compound litteral instead of adding a new helper. - Patch 11 (patch 7 in v2) Remove the file name from the C file header. Use directly converted DAPM macros. Replace <linux/module.h> by <linux/mod_devicetable.h>. Add <linux/platform_device.h>. Be sure that min <= max. Swap values if it is not the case. Move the bool structure member after the int ones. Remove unneeded assignements. Use dev_err_probe() when relevant. Use str_on_off(). Use static_assert() instead of BUILD_BUG_ON(). Remove unneeded comma and blank line. Use device_property_*() instead of the OF API. - patch 8 available in v2 removed as already applied - Patch 12 (patch 9 in v2) Use devm_add_action_or_reset(). Call simple_populate_aux() from simple_parse_of(). Changes v1 -> v2 - Patch 1 Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the /schemas/iio/iio-consumer.yaml reference Remove the unneeded '|' after description - Patch 2 (new in v2) Introduce the simple-audio-card additional-devs subnode - Patch 3 (new in v2) Check err before switch() in iio_channel_read_max() - Patch 4 (new in v2) Fix raw reads and raw writes documentation - Patch 5 (patch 2 in v1) Check err before switch() in iio_channel_read_min() Fix documentation - Patch 6 (path 3 in v1) No changes - Patch 7 (patch 4 in v1) Rename simple-iio-aux to audio-iio-aux Rename invert to snd-control-invert-range Remove the mask usage from audio_iio_aux_{get,put}_volsw helpers Use directly PTR_ERR() in dev_err_probe() parameter Remove the '!!' construction Remove of_match_ptr() - Patch 8 (new in v2) Add a missing of_node_put() in the simple-card driver - Patch 9 (new in v2) Handle additional-devs in the simple-card driver Herve Codina (13): ASoC: dt-bindings: Add audio-iio-aux ASoC: dt-bindings: simple-card: Add additional-devs subnode iio: inkern: Check error explicitly in iio_channel_read_max() iio: consumer.h: Fix raw values documentation notes iio: inkern: Remove the 'unused' variable usage in iio_channel_read_max() iio: inkern: Fix headers inclusion order minmax: Introduce {min,max}_array() iio: inkern: Use max_array() to get the maximum value from an array iio: inkern: Replace a FIXME comment by a TODO one iio: inkern: Add a helper to query an available minimum raw value ASoC: soc-dapm.h: Convert macros to return a compound literal ASoC: codecs: Add support for the generic IIO auxiliary devices ASoC: simple-card: Handle additional devices .../bindings/sound/audio-iio-aux.yaml | 64 ++++ .../bindings/sound/simple-card.yaml | 53 +++ drivers/iio/inkern.c | 86 ++++- include/linux/iio/consumer.h | 37 +- include/linux/minmax.h | 64 ++++ include/sound/soc-dapm.h | 138 ++++--- sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/audio-iio-aux.c | 344 ++++++++++++++++++ sound/soc/generic/simple-card.c | 46 ++- 10 files changed, 775 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/audio-iio-aux.yaml create mode 100644 sound/soc/codecs/audio-iio-aux.c -- 2.40.1

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[PATCH v2 00/10] Refactor acp legacy driver and add
by Syed Saba Kareem 12 Jul '23

12 Jul '23

This patch series to refactor acp leagacy driver and add pm ops support for rembrandt platforms. Changes since V1 - Fix below kernel test robot errors and warnings - ld.lld: error: undefined symbol: acp_enable_interrupts - referenced by acp-renoir.c:191 (sound/soc/amd/acp/acp-renoir.c:191) - ld.lld: error: undefined symbol: acp_disable_interrupts - sound/soc/amd/acp/acp-renoir.o:(renoir_audio_remove) in archive vmlinux.a - All warnings (new ones prefixed by >>): - sound/soc/amd/acp/acp-rembrandt.c:169:5: warning: no previous - prototype for 'acp6x_master_clock_generate' [-Wmissing-prototypes] - 169 | int acp6x_master_clock_generate(struct device *dev) Syed Saba Kareem (10): ASoC: amd: acp: refactor the acp init and de-init sequence ASoC: amd: acp: add acp i2s master clock generation for rembrandt platform ASoC: amd: acp: remove the redundant acp enable/disable interrupts functions ASoC: amd: acp: store platform device reference created in pci probe call ASoC: amd: acp: add pm ops support for acp pci driver ASoC: amd: acp: store xfer_resolution of the stream ASoC: amd: acp: export config_acp_dma() and config_pte_for_stream() symbols ASoC: amd: acp: store the pdm stream channel mask ASoC: amd: acp: move pdm macros to common header file ASoC: amd: acp: add pm ops support for rembrandt platform sound/soc/amd/acp/Kconfig | 5 + sound/soc/amd/acp/Makefile | 2 + sound/soc/amd/acp/acp-i2s.c | 2 + sound/soc/amd/acp/acp-legacy-common.c | 347 ++++++++++++++++++++++++++ sound/soc/amd/acp/acp-pci.c | 57 ++++- sound/soc/amd/acp/acp-pdm.c | 13 +- sound/soc/amd/acp/acp-platform.c | 6 +- sound/soc/amd/acp/acp-rembrandt.c | 186 +++++--------- sound/soc/amd/acp/acp-renoir.c | 115 +-------- sound/soc/amd/acp/amd.h | 51 ++++ 10 files changed, 531 insertions(+), 253 deletions(-) create mode 100644 sound/soc/amd/acp/acp-legacy-common.c -- 2.25.1

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[PATCH] ASoC: rt722-sdca: Remove redundant sdca mask
by Jack Yu 12 Jul '23

12 Jul '23

Remove redundant sdca mask for clear code. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt722-sdca-sdw.c | 2 +- sound/soc/codecs/rt722-sdca.c | 3 +-- 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt722-sdca-sdw.c b/sound/soc/codecs/rt722-sdca-sdw.c index cc57e4e27805..a9416bd163e8 100644 --- a/sound/soc/codecs/rt722-sdca-sdw.c +++ b/sound/soc/codecs/rt722-sdca-sdw.c @@ -175,7 +175,7 @@ static int rt722_sdca_update_status(struct sdw_slave *slave, * This also could sync with the cache value as the rt722_sdca_jack_init set. */ sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK1, - SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6); + SDW_SCP_SDCA_INTMASK_SDCA_6); sdw_write_no_pm(rt722->slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); } diff --git a/sound/soc/codecs/rt722-sdca.c b/sound/soc/codecs/rt722-sdca.c index 9c0d34366c9e..0e1c65a20392 100644 --- a/sound/soc/codecs/rt722-sdca.c +++ b/sound/soc/codecs/rt722-sdca.c @@ -191,8 +191,7 @@ static void rt722_sdca_jack_detect_handler(struct work_struct *work) return; /* SDW_SCP_SDCA_INT_SDCA_6 is used for jack detection */ - if (rt722->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6 || - rt722->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_0) { + if (rt722->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { ret = rt722_sdca_headset_detect(rt722); if (ret < 0) return; -- 2.40.1

2 3

[PATCH 1/1] ASoC: dwc: Add TDM mode support
by Maxim Kochetkov 12 Jul '23

12 Jul '23

Depending on hardware implementaion of DWC I2S controller may support TDM mode if enabled in SoC at design time. Unfortunately there is no way to detect TDM capability for DWC by reading registers. Anyway, if such capability enabled, TDM mode can be enabled and configured by dai-tdm-slot-* DT options. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/dwc/dwc-i2s.c | 65 +++++++++++++++++++++++++++++++++++++++-- sound/soc/dwc/dwc-pcm.c | 8 ++--- sound/soc/dwc/local.h | 24 +++++++++++++++ 3 files changed, 90 insertions(+), 7 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 97d652f0e84d..1f1ee14b04e6 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -183,7 +183,15 @@ static void i2s_start(struct dw_i2s_dev *dev, { struct i2s_clk_config_data *config = &dev->config; - i2s_write_reg(dev->i2s_base, IER, 1); + u32 reg = IER_IEN; + + if (dev->tdm_slots) { + reg |= (dev->tdm_slots - 1) << IER_TDM_SLOTS_SHIFT; + reg |= IER_INTF_TYPE; + reg |= dev->frame_offset << IER_FRAME_OFF_SHIFT; + } + + i2s_write_reg(dev->i2s_base, IER, reg); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) i2s_write_reg(dev->i2s_base, ITER, 1); @@ -233,13 +241,15 @@ static void dw_i2s_config(struct dw_i2s_dev *dev, int stream) dev->xfer_resolution); i2s_write_reg(dev->i2s_base, TFCR(ch_reg), dev->fifo_th - 1); - i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + i2s_write_reg(dev->i2s_base, TER(ch_reg), TER_TXCHEN | + dev->tdm_mask << TER_TXSLOT_SHIFT); } else { i2s_write_reg(dev->i2s_base, RCR(ch_reg), dev->xfer_resolution); i2s_write_reg(dev->i2s_base, RFCR(ch_reg), dev->fifo_th - 1); - i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + i2s_write_reg(dev->i2s_base, RER(ch_reg), RER_RXCHEN | + dev->tdm_mask << RER_RXSLOT_SHIFT); } } @@ -276,6 +286,9 @@ static int dw_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + if (dev->tdm_slots) + config->data_width = 32; + config->chan_nr = params_channels(params); switch (config->chan_nr) { @@ -384,14 +397,58 @@ static int dw_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ret = -EINVAL; break; } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_DSP_A: + dev->frame_offset = 1; + break; + case SND_SOC_DAIFMT_DSP_B: + dev->frame_offset = 0; + break; + default: + dev_err(dev->dev, "DAI format unsupported"); + return -EINVAL; + } + return ret; } +static int dw_i2s_set_tdm_slot(struct snd_soc_dai *cpu_dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + + if (slot_width != 32) + return -EINVAL; + + if (slots < 0 || slots > 16) + return -EINVAL; + + if (rx_mask != tx_mask) + return -EINVAL; + + if (!rx_mask) + return -EINVAL; + + dev->tdm_slots = slots; + dev->tdm_mask = rx_mask; + + dev->l_reg = RSLOT_TSLOT(ffs(rx_mask) - 1); + dev->r_reg = RSLOT_TSLOT(fls(rx_mask) - 1); + + return 0; +} + static const struct snd_soc_dai_ops dw_i2s_dai_ops = { .hw_params = dw_i2s_hw_params, .prepare = dw_i2s_prepare, .trigger = dw_i2s_trigger, .set_fmt = dw_i2s_set_fmt, + .set_tdm_slot = dw_i2s_set_tdm_slot, }; #ifdef CONFIG_PM @@ -726,6 +783,8 @@ static int dw_i2s_probe(struct platform_device *pdev) if (irq >= 0) { ret = dw_pcm_register(pdev); dev->use_pio = true; + dev->l_reg = LRBR_LTHR(0); + dev->r_reg = RRBR_RTHR(0); } else { ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); diff --git a/sound/soc/dwc/dwc-pcm.c b/sound/soc/dwc/dwc-pcm.c index 9f25631d43d3..f99262b89008 100644 --- a/sound/soc/dwc/dwc-pcm.c +++ b/sound/soc/dwc/dwc-pcm.c @@ -31,8 +31,8 @@ static unsigned int dw_pcm_tx_##sample_bits(struct dw_i2s_dev *dev, \ int i; \ \ for (i = 0; i < dev->fifo_th; i++) { \ - iowrite32(p[tx_ptr][0], dev->i2s_base + LRBR_LTHR(0)); \ - iowrite32(p[tx_ptr][1], dev->i2s_base + RRBR_RTHR(0)); \ + iowrite32(p[tx_ptr][0], dev->i2s_base + dev->l_reg); \ + iowrite32(p[tx_ptr][1], dev->i2s_base + dev->r_reg); \ period_pos++; \ if (++tx_ptr >= runtime->buffer_size) \ tx_ptr = 0; \ @@ -51,8 +51,8 @@ static unsigned int dw_pcm_rx_##sample_bits(struct dw_i2s_dev *dev, \ int i; \ \ for (i = 0; i < dev->fifo_th; i++) { \ - p[rx_ptr][0] = ioread32(dev->i2s_base + LRBR_LTHR(0)); \ - p[rx_ptr][1] = ioread32(dev->i2s_base + RRBR_RTHR(0)); \ + p[rx_ptr][0] = ioread32(dev->i2s_base + dev->l_reg); \ + p[rx_ptr][1] = ioread32(dev->i2s_base + dev->r_reg); \ period_pos++; \ if (++rx_ptr >= runtime->buffer_size) \ rx_ptr = 0; \ diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index ba4e397099be..4ce96bac2f39 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -25,6 +25,13 @@ #define RXFFR 0x014 #define TXFFR 0x018 +/* Enable register fields */ +#define IER_TDM_SLOTS_SHIFT 8 +#define IER_FRAME_OFF_SHIFT 5 +#define IER_FRAME_OFF BIT(5) +#define IER_INTF_TYPE BIT(1) +#define IER_IEN BIT(0) + /* Interrupt status register fields */ #define ISR_TXFO BIT(5) #define ISR_TXFE BIT(4) @@ -46,6 +53,15 @@ #define TFCR(x) (0x40 * x + 0x04C) #define RFF(x) (0x40 * x + 0x050) #define TFF(x) (0x40 * x + 0x054) +#define RSLOT_TSLOT(x) (0x4 * (x) + 0x224) + +/* Receive enable register fields */ +#define RER_RXSLOT_SHIFT 8 +#define RER_RXCHEN BIT(0) + +/* Transmit enable register fields */ +#define TER_TXSLOT_SHIFT 8 +#define TER_TXCHEN BIT(0) /* I2SCOMPRegisters */ #define I2S_COMP_PARAM_2 0x01F0 @@ -105,6 +121,8 @@ struct dw_i2s_dev { u32 ccr; u32 xfer_resolution; u32 fifo_th; + u32 l_reg; + u32 r_reg; /* data related to DMA transfers b/w i2s and DMAC */ union dw_i2s_snd_dma_data play_dma_data; @@ -114,6 +132,12 @@ struct dw_i2s_dev { /* data related to PIO transfers */ bool use_pio; + + /* data related to TDM mode */ + u32 tdm_slots; + u32 tdm_mask; + u32 frame_offset; + struct snd_pcm_substream __rcu *tx_substream; struct snd_pcm_substream __rcu *rx_substream; unsigned int (*tx_fn)(struct dw_i2s_dev *dev, -- 2.40.1

2 3

[PATCH -next] ASoC: tas2781: No need to set device_driver owner
by Yang Li 12 Jul '23

12 Jul '23

Remove .owner field if calls are used which set it automatically. to silence the warning: ./sound/soc/codecs/tas2781-i2c.c:746:3-8: No need to set .owner here. The core will do it. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Closes: https://bugzilla.openanolis.cn/show_bug.cgi?id=5589 Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/codecs/tas2781-i2c.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 4c59429a42b7..55cd5e3c23a5 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -743,7 +743,6 @@ MODULE_DEVICE_TABLE(acpi, tasdevice_acpi_match); static struct i2c_driver tasdevice_i2c_driver = { .driver = { .name = "tas2781-codec", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(tasdevice_of_match), #ifdef CONFIG_ACPI .acpi_match_table = ACPI_PTR(tasdevice_acpi_match), -- 2.20.1.7.g153144c

2 1

[PATCH v3] ASoC: Intel: sof_rt5682: add jsl_rt5650 board config
by Brent Lu 12 Jul '23

12 Jul '23

This configuration supports JSL boards which implement ALC5650 dual I2S interface codec. Two DAI links are added: AIF1 (on codec side) for headphone and AIF2 for speakers. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/intel/boards/Kconfig | 5 +- sound/soc/intel/boards/sof_rt5682.c | 80 ++++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 12 +++ 3 files changed, 93 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index f472f603ab75..1fe830af2b84 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -475,7 +475,7 @@ endif ## SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC || SND_SOC_SOF_HDA_AUDIO_CODEC if SND_SOC_SOF_HDA_LINK || SND_SOC_SOF_BAYTRAIL config SND_SOC_INTEL_SOF_RT5682_MACH - tristate "SOF with rt5682 codec in I2S Mode" + tristate "SOF with rt5650/rt5682 codec in I2S Mode" depends on I2C && ACPI depends on ((SND_HDA_CODEC_HDMI && SND_SOC_SOF_HDA_AUDIO_CODEC) &&\ (MFD_INTEL_LPSS || COMPILE_TEST)) ||\ @@ -485,6 +485,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_RT1011 select SND_SOC_RT1015 select SND_SOC_RT1015P + select SND_SOC_RT5645 select SND_SOC_RT5682_I2C select SND_SOC_RT5682S select SND_SOC_DMIC @@ -494,7 +495,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH select SND_SOC_INTEL_SOF_REALTEK_COMMON help This adds support for ASoC machine driver for SOF platforms - with rt5682 codec. + with rt5650 or rt5682 codec. Say Y if you have such a device. If unsure select "N". diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 7c034d671cf3..b4f07bdcf8b4 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -22,6 +22,7 @@ #include <sound/soc-acpi.h> #include "../../codecs/rt5682.h" #include "../../codecs/rt5682s.h" +#include "../../codecs/rt5645.h" #include "../../codecs/hdac_hdmi.h" #include "../common/soc-intel-quirks.h" #include "hda_dsp_common.h" @@ -60,6 +61,7 @@ #define SOF_MAX98390_SPEAKER_AMP_PRESENT BIT(24) #define SOF_MAX98390_TWEETER_SPEAKER_PRESENT BIT(25) #define SOF_RT1019_SPEAKER_AMP_PRESENT BIT(26) +#define SOF_RT5650_HEADPHONE_CODEC_PRESENT BIT(27) /* Default: MCLK on, MCLK 19.2M, SSP0 */ @@ -305,6 +307,7 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) struct sof_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_component *component = asoc_rtd_to_codec(rtd, 0)->component; struct snd_soc_jack *jack; + int extra_jack_data; int ret; /* need to enable ASRC function for 24MHz mclk rate */ @@ -315,7 +318,16 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) RT5682S_DA_STEREO1_FILTER | RT5682S_AD_STEREO1_FILTER, RT5682S_CLK_SEL_I2S1_ASRC); - else + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + rt5645_sel_asrc_clk_src(component, + RT5645_DA_STEREO_FILTER | + RT5645_AD_STEREO_FILTER, + RT5645_CLK_SEL_I2S1_ASRC); + rt5645_sel_asrc_clk_src(component, + RT5645_DA_MONO_L_FILTER | + RT5645_DA_MONO_R_FILTER, + RT5645_CLK_SEL_I2S2_ASRC); + } else rt5682_sel_asrc_clk_src(component, RT5682_DA_STEREO1_FILTER | RT5682_AD_STEREO1_FILTER, @@ -365,7 +377,12 @@ static int sof_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); - ret = snd_soc_component_set_jack(component, jack, NULL); + + if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + extra_jack_data = SND_JACK_MICROPHONE | SND_JACK_BTN_0; + ret = snd_soc_component_set_jack(component, jack, &extra_jack_data); + } else + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); @@ -402,6 +419,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_MCLK; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_MCLK; else pll_source = RT5682_PLL1_S_MCLK; @@ -422,6 +441,8 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, } else { if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) pll_source = RT5682S_PLL_S_BCLK1; + else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) + pll_source = RT5645_PLL1_S_BCLK1; else pll_source = RT5682_PLL1_S_BCLK1; @@ -431,6 +452,9 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { pll_id = RT5682S_PLL2; clk_id = RT5682S_SCLK_S_PLL2; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + pll_id = 0; /* not used in codec driver */ + clk_id = RT5645_SCLK_S_PLL1; } else { pll_id = RT5682_PLL1; clk_id = RT5682_SCLK_S_PLL1; @@ -559,11 +583,30 @@ static const struct snd_soc_dapm_route sof_map[] = { { "IN1P", NULL, "Headset Mic" }, }; +static const struct snd_soc_dapm_route rt5650_spk_dapm_routes[] = { + /* speaker */ + { "Left Spk", NULL, "SPOL" }, + { "Right Spk", NULL, "SPOR" }, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, }; +static int rt5650_spk_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_card *card = rtd->card; + int ret; + + ret = snd_soc_dapm_add_routes(&card->dapm, rt5650_spk_dapm_routes, + ARRAY_SIZE(rt5650_spk_dapm_routes)); + if (ret) + dev_err(rtd->dev, "fail to add dapm routes, ret=%d\n", ret); + + return ret; +} + static int dmic_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_card *card = rtd->card; @@ -614,6 +657,17 @@ static struct snd_soc_dai_link_component rt5682s_component[] = { } }; +static struct snd_soc_dai_link_component rt5650_components[] = { + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif1", + }, + { + .name = "i2c-10EC5650:00", + .dai_name = "rt5645-aif2", + } +}; + static struct snd_soc_dai_link_component dmic_component[] = { { .name = "dmic-codec", @@ -652,6 +706,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, if (sof_rt5682_quirk & SOF_RT5682S_HEADPHONE_CODEC_PRESENT) { links[id].codecs = rt5682s_component; links[id].num_codecs = ARRAY_SIZE(rt5682s_component); + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[0]; + links[id].num_codecs = 1; } else { links[id].codecs = rt5682_component; links[id].num_codecs = ARRAY_SIZE(rt5682_component); @@ -804,6 +861,11 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].init = max_98390_spk_codec_init; links[id].ops = &max_98390_ops; + } else if (sof_rt5682_quirk & SOF_RT5650_HEADPHONE_CODEC_PRESENT) { + links[id].codecs = &rt5650_components[1]; + links[id].num_codecs = 1; + links[id].init = rt5650_spk_init; + links[id].ops = &sof_rt5682_ops; } else { max_98357a_dai_link(&links[id]); } @@ -890,6 +952,12 @@ static int sof_audio_probe(struct platform_device *pdev) /* Detect the headset codec variant */ if (acpi_dev_present("RTL5682", NULL, -1)) sof_rt5682_quirk |= SOF_RT5682S_HEADPHONE_CODEC_PRESENT; + else if (acpi_dev_present("10EC5650", NULL, -1)) { + sof_rt5682_quirk |= SOF_RT5650_HEADPHONE_CODEC_PRESENT; + + sof_audio_card_rt5682.name = devm_kstrdup(&pdev->dev, "rt5650", + GFP_KERNEL); + } if (soc_intel_is_byt() || soc_intel_is_cht()) { is_legacy_cpu = 1; @@ -1178,6 +1246,14 @@ static const struct platform_device_id board_ids[] = { SOF_RT5682_SSP_AMP(0) | SOF_RT5682_NUM_HDMIDEV(3)), }, + { + .name = "jsl_rt5650", + .driver_data = (kernel_ulong_t)(SOF_RT5682_MCLK_EN | + SOF_RT5682_MCLK_24MHZ | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(1)), + }, { } }; MODULE_DEVICE_TABLE(platform, board_ids); diff --git a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c index f5c7e1bbded0..36ea2bacbe7d 100644 --- a/sound/soc/intel/common/soc-acpi-intel-jsl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-jsl-match.c @@ -34,6 +34,11 @@ static const struct snd_soc_acpi_codecs mx98360a_spk = { .codecs = {"MX98360A"} }; +static struct snd_soc_acpi_codecs rt5650_spk = { + .num_codecs = 1, + .codecs = {"10EC5650"} +}; + static const struct snd_soc_acpi_codecs rt5682_rt5682s_hp = { .num_codecs = 2, .codecs = {"10EC5682", "RTL5682"}, @@ -98,6 +103,13 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_jsl_machines[] = { SND_SOC_ACPI_TPLG_INTEL_SSP_MSB | SND_SOC_ACPI_TPLG_INTEL_DMIC_NUMBER, }, + { + .id = "10EC5650", + .drv_name = "jsl_rt5650", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &rt5650_spk, + .sof_tplg_filename = "sof-jsl-rt5650-dual.tplg", + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_jsl_machines); -- 2.34.1

3 4