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

[PATCH] ALSA: scarlett2: Prevent leaking urb
by Chen Ni 21 May '24

21 May '24

In scarlett2_init_notify() if kmalloc() fails the allocated urb should be released. Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/usb/mixer_scarlett2.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer_scarlett2.c b/sound/usb/mixer_scarlett2.c index 1150cf104985..4cad6b2a0292 100644 --- a/sound/usb/mixer_scarlett2.c +++ b/sound/usb/mixer_scarlett2.c @@ -8637,9 +8637,10 @@ static int scarlett2_init_notify(struct usb_mixer_interface *mixer) return -ENOMEM; transfer_buffer = kmalloc(private->wMaxPacketSize, GFP_KERNEL); - if (!transfer_buffer) + if (!transfer_buffer) { + usb_free_urb(mixer->urb); return -ENOMEM; - + } usb_fill_int_urb(mixer->urb, dev, pipe, transfer_buffer, private->wMaxPacketSize, scarlett2_notify, mixer, private->bInterval); -- 2.25.1

2 1

Add UCM files for avs driver
by GitHub pull_request - opened 21 May '24

21 May '24

alsa-project/alsa-ucm-conf pull request #418 was opened from amadeuszslawinski-intel: Present in this pull request are UCM files for some of the devices available on market that we were able to test. They are meant to be used with avs driver and relevant topology files. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/418 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/418.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/2] Support Tegra I2S client format conversion
by Sameer Pujar 21 May '24

21 May '24

The AHUB HW modules are interconnected with CIF which are capable of supporting Channel and Sample bit format conversion. Due to this, the I2S Client can have different Channel and Sample bit from the hw_params() and this config is passed from CIF port of I2S DT node which can help to perform this conversion. - First change to split simple_fixup_sample_fmt to support returning sample format value - Second patch to support Tegra I2S client channel and sample format programming based on CIF port from DT node. Mohan Kumar (2): ASoC: simple-card-utils: Split simple_fixup_sample_fmt func ASoC: tegra: I2S client convert formats handling include/sound/simple_card_utils.h | 2 + sound/soc/generic/simple-card-utils.c | 26 +++++++--- sound/soc/tegra/tegra210_i2s.c | 71 +++++++++++++++++++++++++-- sound/soc/tegra/tegra210_i2s.h | 2 + 4 files changed, 91 insertions(+), 10 deletions(-) -- 2.45.1

2 3

[PATCH 1/2] ASoC: dt-bindings: fsl,mqs: Add i.MX95 platform support
by Shengjiu Wang 21 May '24

21 May '24

In order to support the MQS module on i.MX95, a new property "fsl,mqs-ctrl" needs to be added, as there are two MQS instances on the i.MX95 platform, the definition of bit positions in the control register is different. This new property is to distinguish these two instances. Without this property, the difference of platforms except the i.MX95 was handled by the driver itself. But this new property can also be used for previous platforms. The MQS only has one control register, the register may be in General Purpose Register memory space, or MQS its own memory space, or controlled by System Manager. The bit position in the register may be different for each platform, there are four parts (bits for module enablement, bits for reset, bits for oversampling ratio, bits for divider ratio). This new property includes all these things. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- .../devicetree/bindings/sound/fsl,mqs.yaml | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,mqs.yaml b/Documentation/devicetree/bindings/sound/fsl,mqs.yaml index 8b33353a80ca..a2129b7cb147 100644 --- a/Documentation/devicetree/bindings/sound/fsl,mqs.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,mqs.yaml @@ -23,6 +23,7 @@ properties: - fsl,imx8qm-mqs - fsl,imx8qxp-mqs - fsl,imx93-mqs + - fsl,imx95-mqs clocks: minItems: 1 @@ -45,6 +46,22 @@ properties: resets: maxItems: 1 + fsl,mqs-ctrl: + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 6 + maxItems: 6 + description: | + Contains the control register information, defined as, + Cell #1: register type + 0 - the register in owned register map + 1 - the register in general purpose register map + 2 - the register in control of system manager + Cell #2: offset of the control register from the syscon + Cell #3: shift bits for module enable bit + Cell #4: shift bits for reset bit + Cell #5: shift bits for oversampling ratio bit + Cell #6: shift bits for divider ratio control bit + required: - compatible - clocks -- 2.34.1

3 2

[PATCH v2] ASoC: intel: Constify struct snd_soc_ops
by Christophe JAILLET 21 May '24

21 May '24

Constifying "struct snd_soc_ops" moves some data to a read-only section, so increase overall security. This structure is also part of scripts/const_structs.checkpatch. As an example, on a x86_64, with allmodconfig: Before: text data bss dec hex filename 6315 3696 0 10011 271b sound/soc/intel/boards/ehl_rt5660.o After: text data bss dec hex filename 6379 3648 0 10027 272b sound/soc/intel/boards/ehl_rt5660.o Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- Compile tested only. I hope that it can be applied with this single patch because all files are in sound/soc/intel/boards/ Changes in v2: - Remove changes in sound/soc/intel/boards/sof_maxim_common.[ch]. An even better solution has been applied in commit 2bb765f05391 ("ASoC: Intel: maxim-common: change max98373 data to static") - synch with -next-20240513 v1: https://lore.kernel.org/all/242aef53b5b9533ae4cca78148622f5fe752b7ee.171545… --- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/ehl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 4 ++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +++--- sound/soc/intel/boards/kbl_rt5660.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 8 ++++---- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 6 +++--- sound/soc/intel/boards/sof_es8336.c | 2 +- sound/soc/intel/boards/sof_nau8825.c | 2 +- sound/soc/intel/boards/sof_realtek_common.c | 2 +- sound/soc/intel/boards/sof_rt5682.c | 2 +- sound/soc/intel/boards/sof_sdw_common.h | 2 +- sound/soc/intel/boards/sof_sdw_rt_amp.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- 14 files changed, 22 insertions(+), 22 deletions(-) diff --git a/sound/soc/intel/boards/bdw-rt5650.c b/sound/soc/intel/boards/bdw-rt5650.c index 3ae26f21458f..3c7cee03a02e 100644 --- a/sound/soc/intel/boards/bdw-rt5650.c +++ b/sound/soc/intel/boards/bdw-rt5650.c @@ -131,7 +131,7 @@ static int bdw_rt5650_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops bdw_rt5650_ops = { +static const struct snd_soc_ops bdw_rt5650_ops = { .hw_params = bdw_rt5650_hw_params, }; diff --git a/sound/soc/intel/boards/ehl_rt5660.c b/sound/soc/intel/boards/ehl_rt5660.c index 686e60321224..26289e8fdd87 100644 --- a/sound/soc/intel/boards/ehl_rt5660.c +++ b/sound/soc/intel/boards/ehl_rt5660.c @@ -132,7 +132,7 @@ static int rt5660_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops rt5660_ops = { +static const struct snd_soc_ops rt5660_ops = { .hw_params = rt5660_hw_params, }; diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 9dbc15f9d1c9..154f6a74ed15 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -354,7 +354,7 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -static struct snd_soc_ops kabylake_dmic_ops = { +static const struct snd_soc_ops kabylake_dmic_ops = { .startup = kabylake_dmic_startup, }; @@ -388,7 +388,7 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static struct snd_soc_ops skylake_refcap_ops = { +static const struct snd_soc_ops skylake_refcap_ops = { .startup = kabylake_refcap_startup, }; diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index e662da5af83b..02ed77a07e23 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -288,7 +288,7 @@ static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) return 0; } -static struct snd_soc_ops kabylake_ssp0_ops = { +static const struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, .trigger = kabylake_ssp0_trigger, }; @@ -535,7 +535,7 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -static struct snd_soc_ops kabylake_dmic_ops = { +static const struct snd_soc_ops kabylake_dmic_ops = { .startup = kabylake_dmic_startup, }; @@ -569,7 +569,7 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) } -static struct snd_soc_ops skylake_refcap_ops = { +static const struct snd_soc_ops skylake_refcap_ops = { .startup = kabylake_refcap_startup, }; diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 894d127c482a..66885cb36f24 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -277,7 +277,7 @@ static int kabylake_rt5660_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops kabylake_rt5660_ops = { +static const struct snd_soc_ops kabylake_rt5660_ops = { .hw_params = kabylake_rt5660_hw_params, }; diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index e16c42e81eca..9da89436a917 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -489,7 +489,7 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops kabylake_rt5663_ops = { +static const struct snd_soc_ops kabylake_rt5663_ops = { .hw_params = kabylake_rt5663_hw_params, }; @@ -539,7 +539,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops kabylake_ssp0_ops = { +static const struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, }; @@ -575,7 +575,7 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -static struct snd_soc_ops kabylake_dmic_ops = { +static const struct snd_soc_ops kabylake_dmic_ops = { .startup = kabylake_dmic_startup, }; @@ -609,7 +609,7 @@ static int kabylake_refcap_startup(struct snd_pcm_substream *substream) &constraints_16000); } -static struct snd_soc_ops skylake_refcap_ops = { +static const struct snd_soc_ops skylake_refcap_ops = { .startup = kabylake_refcap_startup, }; diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index a9501cd106ff..a32ce8f972f3 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -424,7 +424,7 @@ static int kabylake_rt5663_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops kabylake_rt5663_ops = { +static const struct snd_soc_ops kabylake_rt5663_ops = { .hw_params = kabylake_rt5663_hw_params, }; @@ -469,7 +469,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops kabylake_ssp0_ops = { +static const struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, }; @@ -508,7 +508,7 @@ static int kabylake_dmic_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } -static struct snd_soc_ops kabylake_dmic_ops = { +static const struct snd_soc_ops kabylake_dmic_ops = { .startup = kabylake_dmic_startup, }; diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index c1fcc156a575..2a88efaa6d26 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -371,7 +371,7 @@ static int sof_es8336_hw_params(struct snd_pcm_substream *substream, } /* machine stream operations */ -static struct snd_soc_ops sof_es8336_ops = { +static const struct snd_soc_ops sof_es8336_ops = { .hw_params = sof_es8336_hw_params, .trigger = sof_8336_trigger, }; diff --git a/sound/soc/intel/boards/sof_nau8825.c b/sound/soc/intel/boards/sof_nau8825.c index c08b4eef0bcb..bfe17acbc161 100644 --- a/sound/soc/intel/boards/sof_nau8825.c +++ b/sound/soc/intel/boards/sof_nau8825.c @@ -115,7 +115,7 @@ static int sof_nau8825_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops sof_nau8825_ops = { +static const struct snd_soc_ops sof_nau8825_ops = { .hw_params = sof_nau8825_hw_params, }; diff --git a/sound/soc/intel/boards/sof_realtek_common.c b/sound/soc/intel/boards/sof_realtek_common.c index dda346e0f737..f52e25083905 100644 --- a/sound/soc/intel/boards/sof_realtek_common.c +++ b/sound/soc/intel/boards/sof_realtek_common.c @@ -452,7 +452,7 @@ static int rt1015_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops rt1015_ops = { +static const struct snd_soc_ops rt1015_ops = { .hw_params = rt1015_hw_params, }; diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index 6fc6eb0c5172..23a40b913290 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -397,7 +397,7 @@ static int sof_rt5682_hw_params(struct snd_pcm_substream *substream, return ret; } -static struct snd_soc_ops sof_rt5682_ops = { +static const struct snd_soc_ops sof_rt5682_ops = { .hw_params = sof_rt5682_hw_params, }; diff --git a/sound/soc/intel/boards/sof_sdw_common.h b/sound/soc/intel/boards/sof_sdw_common.h index 3dfba6f6b95d..7dedac918065 100644 --- a/sound/soc/intel/boards/sof_sdw_common.h +++ b/sound/soc/intel/boards/sof_sdw_common.h @@ -169,7 +169,7 @@ int sof_sdw_rt_sdca_jack_init(struct snd_soc_card *card, int sof_sdw_rt_sdca_jack_exit(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); /* RT1308 I2S support */ -extern struct snd_soc_ops sof_sdw_rt1308_i2s_ops; +extern const struct snd_soc_ops sof_sdw_rt1308_i2s_ops; /* generic amp support */ int sof_sdw_rt_amp_init(struct snd_soc_card *card, diff --git a/sound/soc/intel/boards/sof_sdw_rt_amp.c b/sound/soc/intel/boards/sof_sdw_rt_amp.c index 797ea9ffa77a..d1c0f91ce589 100644 --- a/sound/soc/intel/boards/sof_sdw_rt_amp.c +++ b/sound/soc/intel/boards/sof_sdw_rt_amp.c @@ -233,7 +233,7 @@ static int rt1308_i2s_hw_params(struct snd_pcm_substream *substream, } /* machine stream operations */ -struct snd_soc_ops sof_sdw_rt1308_i2s_ops = { +const struct snd_soc_ops sof_sdw_rt1308_i2s_ops = { .hw_params = rt1308_i2s_hw_params, }; diff --git a/sound/soc/intel/boards/sof_wm8804.c b/sound/soc/intel/boards/sof_wm8804.c index 4cb0d463bf40..b2d02cc92a6a 100644 --- a/sound/soc/intel/boards/sof_wm8804.c +++ b/sound/soc/intel/boards/sof_wm8804.c @@ -148,7 +148,7 @@ static int sof_wm8804_hw_params(struct snd_pcm_substream *substream, } /* machine stream operations */ -static struct snd_soc_ops sof_wm8804_ops = { +static const struct snd_soc_ops sof_wm8804_ops = { .hw_params = sof_wm8804_hw_params, }; -- 2.45.0

2 1

[PATCH 1/6] ASoC: topology: Constify an argument of snd_soc_tplg_component_load()
by Christophe JAILLET 21 May '24

21 May '24

snd_soc_tplg_component_load() does not modify its "*ops" argument. It only read some values and stores it in "soc_tplg.ops". This argument and the ops field in "struct soc_tplg" can be made const. Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- include/sound/soc-topology.h | 2 +- sound/soc/soc-topology.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index f055c6917f6c..1eedd203ac29 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -178,7 +178,7 @@ static inline const void *snd_soc_tplg_get_data(struct snd_soc_tplg_hdr *hdr) /* Dynamic Object loading and removal for component drivers */ int snd_soc_tplg_component_load(struct snd_soc_component *comp, - struct snd_soc_tplg_ops *ops, const struct firmware *fw); + const struct snd_soc_tplg_ops *ops, const struct firmware *fw); int snd_soc_tplg_component_remove(struct snd_soc_component *comp); /* Binds event handlers to dynamic widgets */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 90ca37e008b3..b00ec01361c2 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -73,7 +73,7 @@ struct soc_tplg { int bytes_ext_ops_count; /* optional fw loading callbacks to component drivers */ - struct snd_soc_tplg_ops *ops; + const struct snd_soc_tplg_ops *ops; }; /* check we dont overflow the data for this control chunk */ @@ -2334,7 +2334,7 @@ static int soc_tplg_load(struct soc_tplg *tplg) /* load audio component topology from "firmware" file */ int snd_soc_tplg_component_load(struct snd_soc_component *comp, - struct snd_soc_tplg_ops *ops, const struct firmware *fw) + const struct snd_soc_tplg_ops *ops, const struct firmware *fw) { struct soc_tplg tplg; int ret; -- 2.45.0

4 12

[RESEND PATCH v5 1/3] ASoC: tas2781: Fix wrong loading calibrated data sequence
by Shenghao Ding 20 May '24

20 May '24

Remove declaration of unused API which load calibrated data in wrong sequence, changed the copyright year and correct file name in license header. Fixes: ef3bcde75d06 ("ASoc: tas2781: Add tas2781 driver") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v5: - correct changelog has no much relationship with the patch v4: - Use the the culprit of the bug itself as the fixes tag v3: - No changes. v2: - In the Subject, fixed --> Fix - Add Fixes tag - Changed the copyright year to 2024 in the related files - In tas2781-dsp.h, __TASDEVICE_DSP_H__ --> __TAS2781_DSP_H__ v1: - Download calibrated data after loading the new DSP config params - Remove tasdevice_prmg_calibdata_load, because it is unnecessary to load calibrated data after loading DSP program. --- include/sound/tas2781-dsp.h | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index ea9af2726a53..7fba7ea26a4b 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// Copyright (C) 2022 - 2024 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -13,8 +13,8 @@ // Author: Kevin Lu <kevin-lu(a)ti.com> // -#ifndef __TASDEVICE_DSP_H__ -#define __TASDEVICE_DSP_H__ +#ifndef __TAS2781_DSP_H__ +#define __TAS2781_DSP_H__ #define MAIN_ALL_DEVICES 0x0d #define MAIN_DEVICE_A 0x01 @@ -180,7 +180,6 @@ void tasdevice_calbin_remove(void *context); int tasdevice_select_tuningprm_cfg(void *context, int prm, int cfg_no, int rca_conf_no); int tasdevice_prmg_load(void *context, int prm_no); -int tasdevice_prmg_calibdata_load(void *context, int prm_no); void tasdevice_tuning_switch(void *context, int state); int tas2781_load_calibration(void *context, char *file_name, unsigned short i); -- 2.34.1

2 3

[PATCH v1] ASoC: tas2552: Add TX path for capturing AUDIO-OUT data
by Shenghao Ding 20 May '24

20 May '24

TAS2552 is a Smartamp with I/V sense data, add TX path to support capturing I/V data. Fixes: 38803ce7b53b ("ASoC: codecs: tas*: merge .digital_mute() into .mute_stream()") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - Changed the copyright year to 2024 in the related files. - Add tx device in tas2552_dai. - Add Stream Domain Widgets for I/V capture - Add INPUT widget - Add I/V capture route --- sound/soc/codecs/tas2552.c | 15 +++++++++++++-- 1 file changed, 13 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c index 40f5f27e74c0..9fd3642b98b3 100644 --- a/sound/soc/codecs/tas2552.c +++ b/sound/soc/codecs/tas2552.c @@ -2,7 +2,8 @@ /* * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier * - * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com + * Copyright (C) 2014 - 2024 Texas Instruments Incorporated - + * https://www.ti.com * * Author: Dan Murphy <dmurphy(a)ti.com> */ @@ -119,12 +120,14 @@ static const struct snd_soc_dapm_widget tas2552_dapm_widgets[] = &tas2552_input_mux_control), SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "DAC Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_OUT_DRV("ClassD", TAS2552_CFG_2, 7, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("PLL", TAS2552_CFG_2, 3, 0, NULL, 0), SND_SOC_DAPM_POST("Post Event", tas2552_post_event), - SND_SOC_DAPM_OUTPUT("OUT") + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC") }; static const struct snd_soc_dapm_route tas2552_audio_map[] = { @@ -134,6 +137,7 @@ static const struct snd_soc_dapm_route tas2552_audio_map[] = { {"ClassD", NULL, "Input selection"}, {"OUT", NULL, "ClassD"}, {"ClassD", NULL, "PLL"}, + {"ASI OUT", NULL, "DMIC"} }; #ifdef CONFIG_PM @@ -538,6 +542,13 @@ static struct snd_soc_dai_driver tas2552_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = TAS2552_FORMATS, }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, .ops = &tas2552_speaker_dai_ops, }, }; -- 2.34.1

3 3

[PATCH v5 0/1] Add tas2781 driver for SPI.
by Baojun Xu 20 May '24

20 May '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> Baojun Xu (1): ALSA: hda/tas2781: Add tas2781 hda driver based on SPI drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 12 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 148 ++ sound/pci/hda/tas2781_hda_spi.c | 1234 +++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2247 +++++++++++++++++ 8 files changed, 3671 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c -- 2.40.1

2 3

[PATCH v4 0/4] ASoC: qcom: x1e80100: Correct channel mapping
by Krzysztof Kozlowski 20 May '24

20 May '24

Hi, First patch is a build dependency. Changes in v4: - Rename audioreach_set_channel_mapping -> audioreach_set_default_channel_mapping - Link to v3: https://lore.kernel.org/r/20240509-asoc-x1e80100-4-channel-mapping-v3-0-6f8… Changes in v3: - Add missing Charles' Ack - Link to v2: https://lore.kernel.org/r/20240508-asoc-x1e80100-4-channel-mapping-v2-0-cce… Changes in v2: - Re-work most of the idea according to Srini comments: set channel mapping for backend DAIs, not frontend. - Patch #1: no changes - Patch #2 is entirely replaced - now channel mapping is implemented in q6apm-lpass-dais. - Patch #3: rework to new approach, but most of the code stays. - Patch #4: rework significantly, because only backend DAIs is now affected. - Link to v1: https://lore.kernel.org/r/20240507-asoc-x1e80100-4-channel-mapping-v1-0-b12… Description =========== X1E80100 CRD is the first board, which comes with four speakers, so we still keep fixing and adding missing pieces. The board has speaker arranged as left front+back and then right front+back. Using default channel mapping causes front right speaker to play left back stream. Adjust the channel maps for frontend DAIs to fix stereo and four-channel playback. Best regards, Krzysztof --- Krzysztof Kozlowski (4): ASoC: Constify channel mapping array arguments in set_channel_map() ASoC: qcom: q6apm-lpass-dais: Implement proper channel mapping ASoC: qcom: qdsp6: Set channel mapping instead of fixed defaults ASoC: qcom: x1e80100: Correct channel mapping include/sound/cs35l41.h | 4 ++-- include/sound/soc-dai.h | 8 ++++---- sound/soc/codecs/adau7118.c | 6 ++++-- sound/soc/codecs/cs35l41-lib.c | 4 ++-- sound/soc/codecs/cs35l41.c | 3 ++- sound/soc/codecs/max98504.c | 6 ++++-- sound/soc/codecs/wcd9335.c | 6 ++++-- sound/soc/codecs/wcd934x.c | 6 ++++-- sound/soc/qcom/qdsp6/audioreach.c | 30 +++++++----------------------- sound/soc/qcom/qdsp6/audioreach.h | 2 +- sound/soc/qcom/qdsp6/q6afe-dai.c | 16 ++++++++++------ sound/soc/qcom/qdsp6/q6apm-dai.c | 3 +++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 21 +++++++++++++-------- sound/soc/qcom/x1e80100.c | 18 ++++++++++++++++++ sound/soc/soc-dai.c | 4 ++-- 15 files changed, 80 insertions(+), 57 deletions(-) --- base-commit: 2b84edefcad14934796fad37b16512b6a2ca467e change-id: 20240507-asoc-x1e80100-4-channel-mapping-ea5f02b9e678 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

1 4