June 2022 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 0/2] ASOC: amd: acp: Add Machine driver changes for
by V sujith kumar Reddy 07 Jun '22

07 Jun '22

Add Machine driver support for nau8825, max98560 and rt5682s,rt1019 combination This patch set depends on: --gpio patch --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-1-Vsujithkumar.Reddy@amd.com/ --https://patchwork.kernel.org/project/alsa-devel/patch/20220516160619.17832-2-Vsujithkumar.Reddy@amd.com/ V sujith kumar Reddy (2): ASoC: amd: acp: Add support for nau8825 and max98360 card ASoC: amd: acp: Add support for rt5682s and rt1019 card with hs instance sound/soc/amd/acp-config.c | 30 ++++ sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/acp-mach-common.c | 228 +++++++++++++++++++++++++--- sound/soc/amd/acp/acp-mach.h | 3 + sound/soc/amd/acp/acp-sof-mach.c | 30 ++++ sound/soc/amd/mach-config.h | 1 + 6 files changed, 276 insertions(+), 17 deletions(-) -- 2.25.1

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[PATCH v3 00/17] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (16): ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ASoC: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add module parameter to control firmware load Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 7 + sound/pci/hda/Kconfig | 8 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 815 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 37 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 1 + sound/pci/hda/hda_component.h | 4 + sound/pci/hda/hda_cs_dsp_ctl.c | 302 ++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 27 +- sound/soc/codecs/cs35l41-lib.c | 82 +++- sound/soc/codecs/cs35l41.c | 71 +-- 14 files changed, 1324 insertions(+), 74 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

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[PATCH v2 00/26] *ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (25): ALSA: hda: cs35l41: Fix error in spi cs35l41 hda driver name ALSA: hda: cs35l41: Set Speaker Position for CLSA0100 Laptop ALSA: hda: cs35l41: Remove Set Channel Map api from binding ALSA: hda: cs35l41: Add Support for Interrupts ALSA: hda: cs35l41: Enable GPIO2 Interrupt for CLSA0100 laptops ASoC: cs35l41: Move cs35l41_set_cspl_mbox_cmd to shared code ASoC: cs35l41: Move cs35l41 fs errata into shared code ASoC: cs35l41: Move cs_dsp config struct into shared code ALSA: hda: cs35l41: Add Amp Name based on channel and index ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ASoC: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add kernel config to disable firmware autoload Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 84 +++ sound/pci/hda/Kconfig | 17 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 954 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 41 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 3 +- sound/pci/hda/hda_component.h | 6 +- sound/pci/hda/hda_cs_dsp_ctl.c | 424 ++++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 81 +-- sound/soc/codecs/cs35l41-lib.c | 193 +++++++ sound/soc/codecs/cs35l41.c | 186 +------ sound/soc/codecs/cs35l41.h | 18 - 15 files changed, 1797 insertions(+), 254 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.34.1

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[PATCH 00/26] ALSA: hda: cirrus: Add initial DSP support and firmware loading
by Vitaly Rodionov 07 Jun '22

07 Jun '22

The CS35L41 Amplifier contains a DSP, capable of running firmware. The firmware can run algorithms such as Speaker Protection, to ensure that playback at high gains do not harm the speakers. Adding support for CS35L41 firmware into the CS35L41 HDA driver also allows us to support several extra features, such as hiberation and interrupts. The chain adds support in stages: - General fixes to improve generalization and code re-use inside the CS35L41 HDA driver. - Add support for interrupts into the driver, which is required for complete support of the firmware. - Refactor ASoC CS35L41 code which deals with firmware to allow for code re-use inside the CS35L41 HDA driver. - Add support for loading firmware and tuning files from file system, and creating alsa controls to control it. - Support firmware load paths for different hardware systems. - Support suspend/resume in the driver when using firmware. The firmware supports hibernation, which allows the CS35L41 to drop into a low power mode during suspend. - Support the ability to unload firmware, swap and reload the firmware. This is to allow different firmware to run during calibration. The intended use-case is to load the firmware once on boot, and the driver autmatically tries to load the firmware after it binds to the HDA driver. This behaviour can be switched off using a kconfig, if desired. Stefan Binding (25): ALSA: hda: cs35l41: Fix error in spi cs35l41 hda driver name ALSA: hda: cs35l41: Set Speaker Position for CLSA0100 Laptop ALSA: hda: cs35l41: Remove Set Channel Map api from binding ALSA: hda: cs35l41: Add Support for Interrupts ALSA: hda: cs35l41: Enable GPIO2 Interrupt for CLSA0100 laptops ASoC: cs35l41: Move cs35l41_set_cspl_mbox_cmd to shared code ASoC: cs35l41: Move cs35l41 fs errata into shared code ASoC: cs35l41: Move cs_dsp config struct into shared code ALSA: hda: cs35l41: Add Amp Name based on channel and index ALSA: hda: hda_cs_dsp_ctl: Add Library to support CS_DSP ALSA controls ALSA: hda: hda_cs_dsp_ctl: Add apis to write the controls directly ALSA: hda: cs35l41: Save codec object inside component struct ALSA: hda: cs35l41: Save Subsystem ID inside CS35L41 Driver ALSA: hda: cs35l41: Support reading subsystem id from ACPI ALSA: hda: cs35l41: Support multiple load paths for firmware ALSA: hda: cs35l41: Support Speaker ID for laptops ASoC: cs35l41: Move cs35l41 exit hibernate function into shared code ALSA: cs35l41: Do not print error when waking from hibernation ASoC: cs35l41: Add common cs35l41 enter hibernate function ALSA: hda: cs35l41: Support Hibernation during Suspend ALSA: hda: cs35l41: Read Speaker Calibration data from UEFI variables ALSA: hda: hda_cs_dsp_ctl: Add fw id strings ALSA: hda: cs35l41: Add defaulted values into dsp bypass config sequence ALSA: hda: cs35l41: Support Firmware switching and reloading ALSA: hda: cs35l41: Add kernel config to disable firmware autoload Vitaly Rodionov (1): ALSA: hda: cs35l41: Add initial DSP support and firmware loading MAINTAINERS | 1 + include/sound/cs35l41.h | 84 +++ sound/pci/hda/Kconfig | 17 + sound/pci/hda/Makefile | 2 + sound/pci/hda/cs35l41_hda.c | 954 +++++++++++++++++++++++++++++++- sound/pci/hda/cs35l41_hda.h | 41 ++ sound/pci/hda/cs35l41_hda_i2c.c | 1 + sound/pci/hda/cs35l41_hda_spi.c | 3 +- sound/pci/hda/hda_component.h | 6 +- sound/pci/hda/hda_cs_dsp_ctl.c | 424 ++++++++++++++ sound/pci/hda/hda_cs_dsp_ctl.h | 40 ++ sound/pci/hda/patch_realtek.c | 81 +-- sound/soc/codecs/cs35l41-lib.c | 193 +++++++ sound/soc/codecs/cs35l41.c | 186 +------ sound/soc/codecs/cs35l41.h | 18 - 15 files changed, 1797 insertions(+), 254 deletions(-) create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.c create mode 100644 sound/pci/hda/hda_cs_dsp_ctl.h -- 2.32.0

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[PATCH] ASoC: es8328: Fix event generation for deemphasis control
by Mark Brown 07 Jun '22

07 Jun '22

Currently the put() method for the deemphasis control returns 0 when a new value is written to the control even if the value changed, meaning events are not generated. Fix this, skip the work of updating the value when it is unchanged and then return 1 after having done so. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es8328.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 3f00ead97006..dd53dfd87b04 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -161,13 +161,16 @@ static int es8328_put_deemph(struct snd_kcontrol *kcontrol, if (deemph > 1) return -EINVAL; + if (es8328->deemph == deemph) + return 0; + ret = es8328_set_deemph(component); if (ret < 0) return ret; es8328->deemph = deemph; - return 0; + return 1; } -- 2.30.2

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[PATCH] ASoC: wm8962: Fix suspend while playing music
by Adam Ford 07 Jun '22

07 Jun '22

If the audio CODEC is playing sound when the system is suspended, it can be left in a state which throws the following error: wm8962 3-001a: ASoC: error at soc_component_read_no_lock on wm8962.3-001a: -16 Once this error has occurred, the audio will not work again until rebooted. Fix this by configuring SET_SYSTEM_SLEEP_PM_OPS. Signed-off-by: Adam Ford <aford173(a)gmail.com> diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 34cd5a2a997c..5cca89364280 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3868,6 +3868,7 @@ static int wm8962_runtime_suspend(struct device *dev) #endif static const struct dev_pm_ops wm8962_pm = { + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, pm_runtime_force_resume) SET_RUNTIME_PM_OPS(wm8962_runtime_suspend, wm8962_runtime_resume, NULL) }; -- 2.25.1

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[PATCH] ASoC: ab8500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules
by Christophe JAILLET 07 Jun '22

07 Jun '22

The "Replace GPLv2 boilerplate/reference with SPDX" has left some empty "License terms" paragraphs. Remove them as well. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/codecs/ab8500-codec.c | 2 -- sound/soc/codecs/ab8500-codec.h | 2 -- 2 files changed, 4 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index aefafb0b7b97..cbd4a92cb06c 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -12,8 +12,6 @@ * Mikko Sarmanne <mikko.sarmanne(a)symbio.com>, * Jarmo K. Kuronen <jarmo.kuronen(a)symbio.com>, * for ST-Ericsson. - * - * License terms: */ #include <linux/kernel.h> diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h index 0ac87d0446c2..2a6f6409f1f8 100644 --- a/sound/soc/codecs/ab8500-codec.h +++ b/sound/soc/codecs/ab8500-codec.h @@ -11,8 +11,6 @@ * Mikko J. Lehto <mikko.lehto(a)symbio.com>, * Mikko Sarmanne <mikko.sarmanne(a)symbio.com>, * for ST-Ericsson. - * - * License terms: */ #ifndef AB8500_CODEC_REGISTERS_H -- 2.34.1

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[PATCH] ASoC: ux500: Remove some leftover from the "Replace GPLv2 boilerplate/reference with SPDX" rules
by Christophe JAILLET 07 Jun '22

07 Jun '22

The "Replace GPLv2 boilerplate/reference with SPDX" has left some empty "License terms" paragraphs. Remove them as well. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/ux500/mop500.c | 2 -- sound/soc/ux500/mop500_ab8500.c | 2 -- sound/soc/ux500/mop500_ab8500.h | 2 -- sound/soc/ux500/ux500_msp_dai.c | 2 -- sound/soc/ux500/ux500_msp_dai.h | 2 -- sound/soc/ux500/ux500_msp_i2s.c | 2 -- sound/soc/ux500/ux500_msp_i2s.h | 2 -- sound/soc/ux500/ux500_pcm.c | 2 -- sound/soc/ux500/ux500_pcm.h | 2 -- 9 files changed, 18 deletions(-) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 4f41bb0ab2b0..fdd55d772b8e 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -4,8 +4,6 @@ * * Author: Ola Lilja (ola.o.lilja(a)stericsson.com) * for ST-Ericsson. - * - * License terms: */ #include <asm/mach-types.h> diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c index 1ea1729984a9..e5e73a2bd9fe 100644 --- a/sound/soc/ux500/mop500_ab8500.c +++ b/sound/soc/ux500/mop500_ab8500.c @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Kristoffer Karlsson <kristoffer.karlsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #include <linux/module.h> diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h index 087ef246d87d..98de80a9cc4f 100644 --- a/sound/soc/ux500/mop500_ab8500.h +++ b/sound/soc/ux500/mop500_ab8500.h @@ -4,8 +4,6 @@ * * Author: Ola Lilja <ola.o.lilja(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #ifndef MOP500_AB8500_H diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 21052378a32e..56532b62faf3 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #include <linux/module.h> diff --git a/sound/soc/ux500/ux500_msp_dai.h b/sound/soc/ux500/ux500_msp_dai.h index fcd4b26f5d2d..30bf70838196 100644 --- a/sound/soc/ux500/ux500_msp_dai.h +++ b/sound/soc/ux500/ux500_msp_dai.h @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #ifndef UX500_msp_dai_H diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index fd0b88bb7921..d113411a19f8 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -6,8 +6,6 @@ * Roger Nilsson <roger.xr.nilsson(a)stericsson.com>, * Sandeep Kaushik <sandeep.kaushik(a)st.com> * for ST-Ericsson. - * - * License terms: */ #include <linux/module.h> diff --git a/sound/soc/ux500/ux500_msp_i2s.h b/sound/soc/ux500/ux500_msp_i2s.h index 756b3973af9a..d45b5e2831cc 100644 --- a/sound/soc/ux500/ux500_msp_i2s.h +++ b/sound/soc/ux500/ux500_msp_i2s.h @@ -4,8 +4,6 @@ * * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * for ST-Ericsson. - * - * License terms: */ diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 18191084b8b8..d3802e5ef196 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #include <asm/page.h> diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h index ff3ef7223db6..bd4348ebf9a1 100644 --- a/sound/soc/ux500/ux500_pcm.h +++ b/sound/soc/ux500/ux500_pcm.h @@ -5,8 +5,6 @@ * Author: Ola Lilja <ola.o.lilja(a)stericsson.com>, * Roger Nilsson <roger.xr.nilsson(a)stericsson.com> * for ST-Ericsson. - * - * License terms: */ #ifndef UX500_PCM_H #define UX500_PCM_H -- 2.34.1

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[PATCH] ASoC: simple-card-utils: rename asoc_simple_init_dai_link_params() to asoc_simple_init_for_codec2codec()
by Kuninori Morimoto 07 Jun '22

07 Jun '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 95cfc0a0aaf5752071 ("ASoC: simple-card: Add support for codec2codec DAI links") added the function asoc_simple_init_dai_link_params() to initialize dai_link "params". It is very straight naming, but difficult to noticed that it is for Codec2Codec support. Handling Codec2Codec is one of very tricky part on ALSA SoC, thus it is very important to clarify it. This patch renames the function name. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card-utils.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 539d7f081bd7..fa080f166345 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -513,7 +513,7 @@ static int asoc_simple_init_dai(struct snd_soc_dai *dai, return 0; } -static int asoc_simple_init_dai_link_params(struct snd_soc_pcm_runtime *rtd, +static int asoc_simple_init_for_codec2codec(struct snd_soc_pcm_runtime *rtd, struct simple_dai_props *dai_props) { struct snd_soc_dai_link *dai_link = rtd->dai_link; @@ -575,7 +575,7 @@ int asoc_simple_dai_init(struct snd_soc_pcm_runtime *rtd) return ret; } - ret = asoc_simple_init_dai_link_params(rtd, props); + ret = asoc_simple_init_for_codec2codec(rtd, props); if (ret < 0) return ret; -- 2.25.1

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[PATCH] ASoC: fsl_sai: Add support for i.MX8MN
by Shengjiu Wang 07 Jun '22

07 Jun '22

The SAI module on i.MX8MN is almost same as i.MX8MP, So reuse same soc data as i.MX8MP. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index dae16a14f177..59a91a6d91a9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1299,6 +1299,7 @@ static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,imx8mm-sai", .data = &fsl_sai_imx8mm_data }, { .compatible = "fsl,imx8mp-sai", .data = &fsl_sai_imx8mp_data }, { .compatible = "fsl,imx8ulp-sai", .data = &fsl_sai_imx8ulp_data }, + { .compatible = "fsl,imx8mn-sai", .data = &fsl_sai_imx8mp_data }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, fsl_sai_ids); -- 2.17.1

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