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

imx8mn: Trying to get EASRC to work
by Fabio Estevam 16 Jun '23

16 Jun '23

Hi, I am working on an imx8mn-based board with an ES9080 codec and I am trying to get EASCR to work. (The ES9080 driver is not upstreamed yet). Audio playback works fine via simple-audio-card and here are my changes to get EASRC to work: - sound-ess { - compatible = "simple-audio-card"; - simple-audio-card,name = "ess-audio"; - simple-audio-card,format = "left_j"; - simple-audio-card,frame-master = <&s2cpu>; - simple-audio-card,bitclock-master = <&s2cpu>; - - s2cpu: simple-audio-card,cpu { - sound-dai = <&sai2>; - dai-tdm-slot-num = <8>; - dai-tdm-slot-width = <32>; - }; - - simple-audio-card,codec { - sound-dai = <&es9080>; - }; + sound-es9080 { + compatible = "fsl,imx-audio-es9080"; + model = "es9080-with-easrc"; + audio-cpu = <&sai2>; + audio-codec = <&es9080>; + audio-asrc = <&easrc>; }; }; diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 8d14b5593658..27b8f2c0cd65 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -624,6 +624,14 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) codec_dai_name = "sgtl5000"; priv->codec_priv.mclk_id = SGTL5000_SYSCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-es9080")) { + codec_dai_name = "es9080"; + priv->dai_fmt |= SND_SOC_DAIFMT_CBC_CFC; + priv->dai_link[1].dpcm_capture = 0; + priv->dai_link[2].dpcm_capture = 0; + priv->cpu_priv.slot_width = 32; + priv->card.dapm_routes = audio_map_tx; + priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-tlv320aic32x4")) { codec_dai_name = "tlv320aic32x4-hifi"; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; @@ -902,6 +910,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-tlv320aic32x4", }, { .compatible = "fsl,imx-audio-tlv320aic31xx", }, { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-es9080", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, The es9080-with-easrc card is registered, but after running 'speaker-test' no audio output is present. Is there a way to use the EASRC driver with the simple-audio-card without touching fsl-asoc-card.c? That would be preferable. Would you have any suggestions as to what may be missing to get EASRC to work? Thanks, Fabio Estevam

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[PATCH AUTOSEL 6.1 20/33] ASoC: Intel: avs: Account for UID of ACPI device
by Sasha Levin 16 Jun '23

16 Jun '23

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 836855100b87b4dd7a82546131779dc255c18b67 ] Configurations with multiple codecs attached to the platform are supported but only if each from the set is different. Add new field representing the 'Unique ID' so that codecs that share Vendor and Part IDs can be differentiated and thus enabling support for such configurations. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Link: https://lore.kernel.org/r/20230519201711.4073845-6-amadeuszx.slawinski@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc-acpi.h | 1 + sound/soc/intel/avs/board_selection.c | 2 +- 2 files changed, 2 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-acpi.h b/include/sound/soc-acpi.h index b38fd25c57295..528279056b3ab 100644 --- a/include/sound/soc-acpi.h +++ b/include/sound/soc-acpi.h @@ -170,6 +170,7 @@ struct snd_soc_acpi_link_adr { /* Descriptor for SST ASoC machine driver */ struct snd_soc_acpi_mach { u8 id[ACPI_ID_LEN]; + const char *uid; const struct snd_soc_acpi_codecs *comp_ids; const u32 link_mask; const struct snd_soc_acpi_link_adr *links; diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 87f9c18be238d..87353b4b0cd73 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -394,7 +394,7 @@ static int avs_register_i2s_boards(struct avs_dev *adev) } for (mach = boards->machs; mach->id[0]; mach++) { - if (!acpi_dev_present(mach->id, NULL, -1)) + if (!acpi_dev_present(mach->id, mach->uid, -1)) continue; if (mach->machine_quirk) -- 2.39.2

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[PATCH AUTOSEL 5.10 13/21] ASoC: soc-pcm: test if a BE can be prepared
by Sasha Levin 16 Jun '23

16 Jun '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> [ Upstream commit e123036be377ddf628226a7c6d4f9af5efd113d3 ] In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Link: https://lore.kernel.org/r/20230517185731.487124-1-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- include/sound/soc-dpcm.h | 4 ++++ sound/soc/soc-pcm.c | 20 ++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 0f6c50b17bba8..bd8795198a7d6 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -121,6 +121,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); +/* can this BE perform prepare */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + /* is the current PCM operation for this FE ? */ int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fb874f924bbe3..e52c030bd17a2 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2332,6 +2332,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; + if (!snd_soc_dpcm_can_be_prepared(fe, be, stream)) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && @@ -2972,3 +2975,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); -- 2.39.2

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[PATCH AUTOSEL 6.1 28/57] ASoC: SOF: sof-client-probes: fix pm_runtime imbalance in error handling
by Sasha Levin 16 Jun '23

16 Jun '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit bc424273c74c1565c459c8f2a6ed95caee368d0a ] When an error occurs, we need to make sure the device can pm_runtime suspend instead of keeping it active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com Reviewed-by: Daniel Baluta <daniel.baluta(a)nxp.com Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com Link: https://lore.kernel.org/r/20230512103315.8921-4-peter.ujfalusi@linux.intel.… Signed-off-by: Mark Brown <broonie(a)kernel.org Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/sof-client-probes.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/sof-client-probes.c b/sound/soc/sof/sof-client-probes.c index ddeabbb5580e1..1e521a9e34d27 100644 --- a/sound/soc/sof/sof-client-probes.c +++ b/sound/soc/sof/sof-client-probes.c @@ -441,12 +441,7 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to, ret = sof_probes_points_info(cdev, &desc, &num_desc); if (ret < 0) - goto exit; - - pm_runtime_mark_last_busy(dev); - err = pm_runtime_put_autosuspend(dev); - if (err < 0) - dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err); + goto pm_error; for (i = 0; i < num_desc; i++) { offset = strlen(buf); @@ -464,6 +459,13 @@ static ssize_t sof_probes_dfs_points_read(struct file *file, char __user *to, ret = simple_read_from_buffer(to, count, ppos, buf, strlen(buf)); kfree(desc); + +pm_error: + pm_runtime_mark_last_busy(dev); + err = pm_runtime_put_autosuspend(dev); + if (err < 0) + dev_err_ratelimited(dev, "debugfs read failed to idle %d\n", err); + exit: kfree(buf); return ret; -- 2.39.2

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topology: nhlt: intel: support m/n divider and xtal/cardinal/pll clock source
by GitHub pull_request - edited 16 Jun '23

16 Jun '23

alsa-project/alsa-utils pull request #220 was edited from brentlu: Current alsatplg supports only cardinal clock source without m/n divider. Code in SOF ssp driver is ported here to implement the divider function. I tested the code on ADL brya device with following setting: MCLK: 19.2MHz BCLK: 3.072Mz (m 24 n 25) Request URL : https://github.com/alsa-project/alsa-utils/pull/220 Patch URL : https://github.com/alsa-project/alsa-utils/pull/220.patch Repository URL: https://github.com/alsa-project/alsa-utils

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topology: nhlt: intel: support m/n divider and xtal/cardinal/pll clock source
by GitHub pull_request - opened 16 Jun '23

16 Jun '23

alsa-project/alsa-utils pull request #220 was opened from brentlu: Current alsatplg supports only cardinal clock source without m/n divider. Code in SOF ssp driver is ported here to implement the clock source selection and divider function. I tested the code on ADL brya device with following setting: MCLK: 19.2MHz BCLK: 3.072Mz (m 24 n 25) Request URL : https://github.com/alsa-project/alsa-utils/pull/220 Patch URL : https://github.com/alsa-project/alsa-utils/pull/220.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v2 1/1] ASoC: dt-bindings: tlv320aic32x4: convert to DT schema format
by Alexander Stein 16 Jun '23

16 Jun '23

Convert the binding to DT schema format. Since commit 514b044cba667 ("ASoC: tlv320aic32x4: Model PLL in CCF") clocks & clock-names = "mclk" is mandatory, it has been added to required properties as well. '#sound-dai-cells' is added for reference from simple-audio-card. Signed-off-by: Alexander Stein <alexander.stein(a)ew.tq-group.com> --- Changes in v2: * Rebased onto for-next of [1] * Add vendor prefix 'ti' to file name * Simplify 'reg' property * Removed description from 'reset-gpios' * Added reference to dai-common.yaml * Fixed node name in example There is no in-tree user for aic32x4-gpio-func, so I just converted the bindings to YAML and skipped introducing a dt-binding header for defining the constants. [1] https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/log/ .../bindings/sound/ti,tlv320aic32x4.yaml | 101 ++++++++++++++++++ .../bindings/sound/tlv320aic32x4.txt | 42 -------- 2 files changed, 101 insertions(+), 42 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt diff --git a/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml new file mode 100644 index 0000000000000..a7cc9aa344684 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tlv320aic32x4.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ti,tlv320aic32x4.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320AIC32x4 Stereo Audio codec + +maintainers: + - Alexander Stein <alexander.stein(a)ew.tq-group.com> + +description: | + The TLV320AIC32x4 audio codec can be accessed using I2C or SPI + +properties: + compatible: + enum: + - ti,tas2505 + - ti,tlv320aic32x4 + - ti,tlv320aic32x6 + + reg: + maxItems: 1 + + clocks: + items: + - description: Master clock + + clock-names: + items: + - const: mclk + + av-supply: + description: Analog core power supply + + dv-supply: + description: Digital core power supply + + iov-supply: + description: Digital IO power supply + + ldoin-supply: + description: LDO power supply + + reset-gpios: + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + aic32x4-gpio-func: + description: | + GPIO function configuration for pins MFP1-MFP5. + Types are defined in include/sound/tlv320aic32x4.h + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 5 + maxItems: 5 + +required: + - compatible + - reg + - clocks + - clock-names + - iov-supply + +allOf: + - $ref: dai-common.yaml# + - if: + not: + required: + - ldoin-supply + then: + required: + - av-supply + - dv-supply + +additionalProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c { + #address-cells = <1>; + #size-cells = <0>; + audio-codec@18 { + compatible = "ti,tlv320aic32x4"; + reg = <0x18>; + iov-supply = <&reg_3v3>; + ldoin-supply = <&reg_3v3>; + clocks = <&clks 201>; + clock-names = "mclk"; + aic32x4-gpio-func= < + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ + 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ + 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */ + >; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt deleted file mode 100644 index 0b4e21bde5bc1..0000000000000 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ /dev/null @@ -1,42 +0,0 @@ -Texas Instruments - tlv320aic32x4 Codec module - -The tlv320aic32x4 serial control bus communicates through I2C protocols - -Required properties: - - compatible - "string" - One of: - "ti,tlv320aic32x4" TLV320AIC3204 - "ti,tlv320aic32x6" TLV320AIC3206, TLV320AIC3256 - "ti,tas2505" TAS2505, TAS2521 - - reg: I2C slave address - - *-supply: Required supply regulators are: - "iov" - digital IO power supply - "ldoin" - LDO power supply - "dv" - Digital core power supply - "av" - Analog core power supply - If you supply ldoin, dv and av are optional. Otherwise they are required - See regulator/regulator.txt for more information about the detailed binding - format. - -Optional properties: - - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt - - clocks/clock-names: Clock named 'mclk' for the master clock of the codec. - See clock/clock-bindings.txt for information about the detailed format. - - aic32x4-gpio-func - <array of 5 int> - - Types are defined in include/sound/tlv320aic32x4.h - - -Example: - -codec: tlv320aic32x4@18 { - compatible = "ti,tlv320aic32x4"; - reg = <0x18>; - clocks = <&clks 201>; - clock-names = "mclk"; - aic32x4-gpio-func= < - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0x04 /* MFP3 AIC32X4_MFP3_GPIO_ENABLED */ - 0xff /* AIC32X4_MFPX_DEFAULT_VALUE */ - 0x08 /* MFP5 AIC32X4_MFP5_GPIO_INPUT */ - >; -}; -- 2.34.1

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[PATCH] ASoC: codecs: wm0010: Add MODULE_FIRMWARE macros
by Juerg Haefliger 16 Jun '23

16 Jun '23

The module loads firmware so add MODULE_FIRMWARE macros to provide that information via modinfo. Signed-off-by: Juerg Haefliger <juerg.haefliger(a)canonical.com> --- sound/soc/codecs/wm0010.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 034a4e858c7e..1d4259433f47 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -994,3 +994,6 @@ module_spi_driver(wm0010_spi_driver); MODULE_DESCRIPTION("ASoC WM0010 driver"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); MODULE_LICENSE("GPL"); + +MODULE_FIRMWARE("wm0010.dfw"); +MODULE_FIRMWARE("wm0010_stage2.bin"); -- 2.37.2

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[PATCH] ASoC: rt5677: Add MODULE_FIRMWARE macro
by Juerg Haefliger 16 Jun '23

16 Jun '23

The module loads firmware so add a MODULE_FIRMWARE macro to provide that information via modinfo. Signed-off-by: Juerg Haefliger <juerg.haefliger(a)canonical.com> --- sound/soc/codecs/rt5677.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5677.c b/sound/soc/codecs/rt5677.c index 3bf019b3f700..c8fb98e3406f 100644 --- a/sound/soc/codecs/rt5677.c +++ b/sound/soc/codecs/rt5677.c @@ -5712,3 +5712,5 @@ module_i2c_driver(rt5677_i2c_driver); MODULE_DESCRIPTION("ASoC RT5677 driver"); MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); MODULE_LICENSE("GPL v2"); + +MODULE_FIRMWARE("rt5677_elf_vad"); -- 2.37.2

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[PATCH 0/8] ASoC: SOF: misc updates for 6.5
by Pierre-Louis Bossart 16 Jun '23

16 Jun '23

Couple of improvements on virtual_widget support, firmware trace free, IPC payload dump, duplicated code in suspend and MeteorLake primary code support. Bard Liao (2): ASoC: SOF: sof-audio: add is_virtual_widget helper ASoC: SOF: sof-audio: test virtual widget in sof_walk_widgets_in_order Peter Ujfalusi (5): ASoC: SOF: core: Free the firmware trace before calling snd_sof_shutdown() ASoC: SOF: Add new sof_debug flag to request message payload dump ASoC: SOF: ipc3: Dump IPC message payload ASoC: SOF: ipc4: Switch to use the sof_debug:bit11 to dump message payload ASoC: SOF: pm: Remove duplicated code in sof_suspend Rander Wang (1): ASoC: SOF: Intel: mtl: setup primary core info on MeteorLake platform sound/soc/sof/core.c | 4 +++- sound/soc/sof/intel/mtl.c | 19 ++++++++++++---- sound/soc/sof/ipc3.c | 39 ++++++++++++++++++++++++++++++++ sound/soc/sof/ipc4.c | 46 ++++++++++++++++++-------------------- sound/soc/sof/pm.c | 12 ++++------ sound/soc/sof/sof-audio.c | 47 ++++++++++++++++++++++++++------------- sound/soc/sof/sof-priv.h | 3 +++ 7 files changed, 117 insertions(+), 53 deletions(-) -- 2.37.2

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