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

Fix incorrect usages of `strerror`
by GitHub pull_request - opened 21 Nov '23

21 Nov '23

alsa-project/alsa-utils pull request #245 was opened from i-Pear: `strerror` takes the `errno` directly as its argument, negating it will result in an "Unknown error". This fixes such usages across multiple modules. Request URL : https://github.com/alsa-project/alsa-utils/pull/245 Patch URL : https://github.com/alsa-project/alsa-utils/pull/245.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

No sound over HDMI using USB Type-C port replicator with Philips TV
by Paul Menzel 21 Nov '23

21 Nov '23

Dear Linux folks, Connecting a USB Type-C port replicator [1] to the only USB Type-C port of the Dell XPS 13 9360 with Debian sid/unstable and Debian’s Linux kernel 6.10.5, and then connecting a Philips 40PFL5206H/12 TV device to the HDMI port, GNOME shows the HDMI sound device, but selecting it as output device, the TV does not play any sound. (Video display works.) Using the same USB Type-C port replicator with the Google Chromebook Lenovo IdeaPad Duet, video and sound over HDMI work. Using the same USB Type-C port replicator with the same Dell XPS 13 9360, but a Samsung TV, video and sound over HDMI also work. Please found the output of `/sbin/alsa-info` from *alsa-utils* 1.2.10-1 attached. Kind regards, Paul [1]: https://lmp-adapter.com/product/lmp-usb-c-mini-dock/?attribute_pa_color=sil… "LMP USB-C mini Dock (P/N 15954)"

1 0

[PATCH v10 00/14] Add audio support in v4l2 framework
by Shengjiu Wang 21 Nov '23

21 Nov '23

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v10 - remove FIXED_POINT type - change code base on media: v4l2-ctrls: add support for fraction_bits - fix issue reported by kernel test robot - remove module_alias changes in v9: - add MEDIA_ENT_F_PROC_AUDIO_RESAMPLER. - add MEDIA_INTF_T_V4L_AUDIO - add media controller support - refine the vim2m-audio to support 8k<->16k conversion. changes in v8: - refine V4L2_CAP_AUDIO_M2M to be 0x00000008 - update doc for FIXED_POINT - address comments for imx-asrc changes in v7: - add acked-by from Mark - separate commit for fixed point, m2m audio class, audio rate controls - use INTEGER_MENU for rate, FIXED_POINT for rate offset - remove used fmts - address other comments for Hans changes in v6: - use m2m_prepare/m2m_unprepare/m2m_start/m2m_stop to replace m2m_start_part_one/m2m_stop_part_one, m2m_start_part_two/m2m_stop_part_two. - change V4L2_CTRL_TYPE_ASRC_RATE to V4L2_CTRL_TYPE_FIXED_POINT - fix warning by kernel test rebot - remove some unused format V4L2_AUDIO_FMT_XX - Get SNDRV_PCM_FORMAT from V4L2_AUDIO_FMT in driver. - rename audm2m to viaudm2m. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Shengjiu Wang (14): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Define audio sample format fourcc type media: uapi: Add V4L2_CTRL_CLASS_M2M_AUDIO media: uapi: Add audio rate controls support media: uapi: Declare interface types for Audio media: uapi: Add an entity type for audio resampler media: imx-asrc: Add memory to memory driver media: vim2m-audio: add virtual driver for audio memory to memory .../media/mediactl/media-types.rst | 11 + .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-audio-m2m.rst | 41 + .../userspace-api/media/v4l/pixfmt-audio.rst | 87 ++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/videodev2.h.rst.exceptions | 3 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 13 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1264 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 11 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 799 +++++++++++ drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 9 + drivers/media/v4l2-core/v4l2-ctrls-defs.c | 10 + drivers/media/v4l2-core/v4l2-dev.c | 21 + drivers/media/v4l2-core/v4l2-ioctl.c | 66 + drivers/media/v4l2-core/v4l2-mem2mem.c | 13 +- include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 60 + include/uapi/linux/media.h | 2 + include/uapi/linux/v4l2-controls.h | 9 + include/uapi/linux/videodev2.h | 41 + sound/soc/fsl/fsl_asrc.c | 144 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 233 +++ sound/soc/fsl/fsl_easrc.h | 6 +- 36 files changed, 2977 insertions(+), 7 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-audio-m2m.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/vim2m-audio.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

3 19

[RESEND v2 0/3] Add audio device nodes for i.MX93 platform
by Chancel Liu 21 Nov '23

21 Nov '23

changes in v2 - Move MQS device node under "/" root node - Convert fsl,mqs.txt to yaml Tests pass based on broonie-sound/for-next. Chancel Liu (3): arm64: dts: imx93: Add audio device nodes arm64: dts: imx93: Configure clock rate for audio PLL ASoC: dt-bindings: fsl,mqs: Convert format to json-schema .../devicetree/bindings/sound/fsl,mqs.txt | 36 ------ .../devicetree/bindings/sound/fsl,mqs.yaml | 113 ++++++++++++++++++ arch/arm64/boot/dts/freescale/imx93.dtsi | 89 ++++++++++++++ 3 files changed, 202 insertions(+), 36 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.txt create mode 100644 Documentation/devicetree/bindings/sound/fsl,mqs.yaml -- 2.42.0

2 5

[PATCH v4 1/2] ASoC: dt-bindings: sound-card-common: List sound widgets ignoring system suspend
by Chancel Liu 20 Nov '23

20 Nov '23

Add a property to list audio sound widgets which are marked ignoring system suspend. Paths between these endpoints are still active over suspend of the main application processor that the current operating system is running. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- .../devicetree/bindings/sound/sound-card-common.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sound-card-common.yaml b/Documentation/devicetree/bindings/sound/sound-card-common.yaml index 3a941177f684..721950f65748 100644 --- a/Documentation/devicetree/bindings/sound/sound-card-common.yaml +++ b/Documentation/devicetree/bindings/sound/sound-card-common.yaml @@ -17,6 +17,13 @@ properties: pair of strings, the first being the connection's sink, the second being the connection's source. + ignore-suspend-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of audio sound widgets which are marked ignoring system suspend. + Paths between these endpoints are still active over suspend of the main + application processor that the current operating system is running. + model: $ref: /schemas/types.yaml#/definitions/string description: User specified audio sound card name -- 2.25.1

3 4

[PATCH v3] ASoC: cs43130: Allow driver to work without IRQ connection
by Maciej Strozek 20 Nov '23

20 Nov '23

Add a polling mechanism that will keep the driver operational even in absence of physical IRQ connection. If IRQ line is detected, the driver will continue working as usual, in case of missing IRQ line it will fallback to the polling mechanism introduced in this change. This will support users which choose not to connect an IRQ line as it is not critical to part's operation. Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- V2 -> V3: Amended changelog message and subject line V1 -> V2: Add changelog message sound/soc/codecs/cs43130.c | 56 +++++++++++++++++++++++++++++++------- sound/soc/codecs/cs43130.h | 1 + 2 files changed, 47 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/cs43130.c b/sound/soc/codecs/cs43130.c index fd39328579fb..1e7c32eedc7b 100644 --- a/sound/soc/codecs/cs43130.c +++ b/sound/soc/codecs/cs43130.c @@ -326,6 +326,43 @@ static int cs43130_set_pll(struct snd_soc_component *component, int pll_id, int return ret; } +static int cs43130_wait_for_completion(struct cs43130_private *cs43130, struct completion *to_poll, + int time) +{ + int stickies, offset, flag; + int ret = 0; + + if (cs43130->has_irq_line) { + ret = wait_for_completion_timeout(to_poll, msecs_to_jiffies(time)); + } else { + if (to_poll == &cs43130->xtal_rdy) { + offset = 0; + flag = CS43130_XTAL_RDY_INT; + } else if (to_poll == &cs43130->pll_rdy) { + offset = 0; + flag = CS43130_PLL_RDY_INT; + } else if (to_poll == &cs43130->hpload_evt) { + offset = 3; + flag = CS43130_HPLOAD_NO_DC_INT | CS43130_HPLOAD_UNPLUG_INT | + CS43130_HPLOAD_OOR_INT | CS43130_HPLOAD_AC_INT | + CS43130_HPLOAD_DC_INT | CS43130_HPLOAD_ON_INT | + CS43130_HPLOAD_OFF_INT; + } else { + return 0; + } + + ret = regmap_read_poll_timeout(cs43130->regmap, CS43130_INT_STATUS_1 + offset, + stickies, (stickies & flag), + 1000, time * 1000); + + /* + * Return 0 for an timeout/error to be consistent with wait_for_completion_timeout + */ + ret = !ret; + } + return ret; +} + static int cs43130_change_clksrc(struct snd_soc_component *component, enum cs43130_mclk_src_sel src) { @@ -364,8 +401,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); @@ -400,8 +436,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_XTAL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_XTAL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->xtal_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->xtal_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_XTAL_RDY_INT_MASK, 1 << CS43130_XTAL_RDY_INT_SHIFT); @@ -416,8 +451,7 @@ static int cs43130_change_clksrc(struct snd_soc_component *component, CS43130_PLL_RDY_INT_MASK, 0); regmap_update_bits(cs43130->regmap, CS43130_PWDN_CTL, CS43130_PDN_PLL_MASK, 0); - ret = wait_for_completion_timeout(&cs43130->pll_rdy, - msecs_to_jiffies(100)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->pll_rdy, 100); regmap_update_bits(cs43130->regmap, CS43130_INT_MASK_1, CS43130_PLL_RDY_INT_MASK, 1 << CS43130_PLL_RDY_INT_SHIFT); @@ -2040,8 +2074,8 @@ static int cs43130_hpload_proc(struct cs43130_private *cs43130, regmap_multi_reg_write(cs43130->regmap, seq, seq_size); - ret = wait_for_completion_timeout(&cs43130->hpload_evt, - msecs_to_jiffies(1000)); + ret = cs43130_wait_for_completion(cs43130, &cs43130->hpload_evt, 1000); + regmap_read(cs43130->regmap, CS43130_INT_MASK_4, &msk); if (!ret) { dev_err(cs43130->dev, "Timeout waiting for HPLOAD interrupt\n"); @@ -2545,8 +2579,10 @@ static int cs43130_i2c_probe(struct i2c_client *client) IRQF_ONESHOT | IRQF_TRIGGER_LOW, "cs43130", cs43130); if (ret != 0) { - dev_err(cs43130->dev, "Failed to request IRQ: %d\n", ret); - goto err; + dev_dbg(cs43130->dev, "Failed to request IRQ: %d, will poll instead\n", ret); + cs43130->has_irq_line = 0; + } else { + cs43130->has_irq_line = 1; } cs43130->mclk_int_src = CS43130_MCLK_SRC_RCO; diff --git a/sound/soc/codecs/cs43130.h b/sound/soc/codecs/cs43130.h index d3f595bbd3ba..dbdb5b262f1b 100644 --- a/sound/soc/codecs/cs43130.h +++ b/sound/soc/codecs/cs43130.h @@ -508,6 +508,7 @@ struct cs43130_private { struct gpio_desc *reset_gpio; unsigned int dev_id; /* codec device ID */ int xtal_ibias; + bool has_irq_line; /* shared by both DAIs */ struct mutex clk_mutex; -- 2.34.1

3 7

[PATCH] ASoC: nau8810: Fix incorrect type in assignment and cast to restricted __be16
by David Lin 20 Nov '23

20 Nov '23

This issue is reproduced when W=1 build in compiler gcc-12. The following are sparse warnings: sound/soc/codecs/nau8810.c:183:25: sparse: warning: incorrect type in assignment sound/soc/codecs/nau8810.c:183:25: sparse: expected int sound/soc/codecs/nau8810.c:183:25: sparse: got restricted __be16 sound/soc/codecs/nau8810.c:219:25: sparse: warning: cast to restricted __be16 sound/soc/codecs/nau8810.c:219:25: sparse: warning: cast to restricted __be16 sound/soc/codecs/nau8810.c:219:25: sparse: warning: cast to restricted __be16 sound/soc/codecs/nau8810.c:219:25: sparse: warning: cast to restricted __be16 This issue is not still actively checked by kernel test robot. Actually, it is same with nau8822's sparse warnings issue. Signed-off-by: David Lin <CTLIN0(a)nuvoton.com> --- sound/soc/codecs/nau8810.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/nau8810.c b/sound/soc/codecs/nau8810.c index 47f000cd4d99..97a54059474c 100644 --- a/sound/soc/codecs/nau8810.c +++ b/sound/soc/codecs/nau8810.c @@ -169,6 +169,7 @@ static int nau8810_eq_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *params = (void *)kcontrol->private_value; int i, reg, reg_val; u16 *val; + __be16 tmp; val = (u16 *)ucontrol->value.bytes.data; reg = NAU8810_REG_EQ1; @@ -177,8 +178,8 @@ static int nau8810_eq_get(struct snd_kcontrol *kcontrol, /* conversion of 16-bit integers between native CPU format * and big endian format */ - reg_val = cpu_to_be16(reg_val); - memcpy(val + i, &reg_val, sizeof(reg_val)); + tmp = cpu_to_be16(reg_val); + memcpy(val + i, &tmp, sizeof(tmp)); } return 0; @@ -201,6 +202,7 @@ static int nau8810_eq_put(struct snd_kcontrol *kcontrol, void *data; u16 *val, value; int i, reg, ret; + __be16 *tmp; data = kmemdup(ucontrol->value.bytes.data, params->max, GFP_KERNEL | GFP_DMA); @@ -213,7 +215,8 @@ static int nau8810_eq_put(struct snd_kcontrol *kcontrol, /* conversion of 16-bit integers between native CPU format * and big endian format */ - value = be16_to_cpu(*(val + i)); + tmp = (__be16 *)(val + i); + value = be16_to_cpup(tmp); ret = regmap_write(nau8810->regmap, reg + i, value); if (ret) { dev_err(component->dev, "EQ configuration fail, register: %x ret: %d\n", -- 2.25.1

2 1

[PATCH] ASoC: fsl_mqs: Remove duplicate linux/of.h header
by Lucas Tanure 20 Nov '23

20 Nov '23

Remove linux/of.h as is included more than once. Reported by make includecheck. Signed-off-by: Lucas Tanure <tanure(a)linux.com> --- sound/soc/fsl/fsl_mqs.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index f2d74ec05cdf..86704ba5f6f0 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -10,7 +10,6 @@ #include <linux/moduleparam.h> #include <linux/mfd/syscon.h> #include <linux/mfd/syscon/imx6q-iomuxc-gpr.h> -#include <linux/of.h> #include <linux/pm_runtime.h> #include <linux/of.h> #include <linux/pm.h> -- 2.42.1

3 2

[PATCH] ASoC: fsl_sai: Fix no frame sync clock issue on i.MX8MP
by Shengjiu Wang 20 Nov '23

20 Nov '23

On i.MX8MP, when the TERE and FSD_MSTR enabled before configuring the word width, there will be no frame sync clock issue, because old word width impact the generation of frame sync. TERE enabled earlier only for i.MX8MP case for the hardware limitation, So need to disable FSD_MSTR before configuring word width, then enable FSD_MSTR bit for this specific case. Fixes: 3e4a82612998 ("ASoC: fsl_sai: MCLK bind with TX/RX enable bit") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 21 +++++++++++++++++++++ 1 file changed, 21 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 79e7c6b98a75..32bbe5056a63 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -673,6 +673,20 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, FSL_SAI_CR3_TRCE_MASK, FSL_SAI_CR3_TRCE((dl_cfg[dl_cfg_idx].mask[tx] & trce_mask))); + /* + * When the TERE and FSD_MSTR enabled before configuring the word width + * There will be no frame sync clock issue, because word width impact + * the generation of frame sync clock. + * + * TERE enabled earlier only for i.MX8MP case for the hardware limitation, + * We need to disable FSD_MSTR before configuring word width, then enable + * FSD_MSTR bit for this specific case. + */ + if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output && + !sai->is_consumer_mode) + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), + FSL_SAI_CR4_FSD_MSTR, 0); + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK | FSL_SAI_CR4_CHMOD_MASK, @@ -680,6 +694,13 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | FSL_SAI_CR5_FBT_MASK, val_cr5); + + /* Enable FSD_MSTR after configuring word width */ + if (sai->soc_data->mclk_with_tere && sai->mclk_direction_output && + !sai->is_consumer_mode) + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx, ofs), + FSL_SAI_CR4_FSD_MSTR, FSL_SAI_CR4_FSD_MSTR); + regmap_write(sai->regmap, FSL_SAI_xMR(tx), ~0UL - ((1 << min(channels, slots)) - 1)); -- 2.34.1

2 1

[PATCHv2 00/10] ASoC: fsl-asoc-card: compatibility integration of a generic codec use case for use with S/PDIF controller
by Elinor Montmasson 20 Nov '23

20 Nov '23

Hello, This is the v2 of the series of patch aiming to make the machine driver fsl-asoc-card compatible with use cases where there is no real codec driver. It proposes to use the spdif_receiver and spdif_transmitter drivers instead of the dummy codec. This is a first step in using the S/PDIF controller with the ASRC. The five first patches add compatibility with the pair of codecs spdif_receiver and spdif_transmitter with a new compatible, "fsl,imx-audio-generic". Codec parameters are set with default values. The remaining patches add configuration options for the device tree. They configure the CPU DAI when using "fsl,imx-audio-generic". These are usually hard-coded in fsl-asoc-card for each audio codec. Because the generic codec could be used with other CPU DAI than the S/PDIF controller, setting these parameters could be required. This series of patch was successfully built for arm64 and x86 on top of the latest for-next branch of the ASoC git tree on the 24/10. These modifications have also been tested on an i.MX8MN evaluation board, with a linux kernel RT v6.1.26-rt8. We also have a few questions : * We named the compatible "fsl,imx-audio-generic" as, for the moment, it could work with any CPU DAI, even if it uses S/PDIF codecs. Is it preferable to keep these modifications specific to the S/PDIF, thus specifying "spdif" in the compatible ? Or is it okay to keep the actual name of the compatible, and the generic purpose of this compatible ? * Part of the dai_fmt variable hold information on the codec provider or consumer status for bit/frame clocks. In patch 03/10, as we add compatibility for multiple codecs, we make the test about bit/frame clock provider only check with codec[0]. That way we assure compatibility with all already existing compatibles. As it was never intended before to have multiple codecs for this test, is there a better way to handle it ? Should we make this test check if any codec is clock provider ? Or should we let codec[0] be the default possibility ? That way, future compatibles that could encounter this specific case with multi-codecs should adapt the test for their needs. Best regards, Elinor Montmasson Changelog: v1 -> v2: * replace use of the dummy codec by the pair of codecs spdif_receiver / spdif_transmitter * adapt how dai links codecs are used to take into account the possibility for multiple codecs per link * change compatible name * adapt driver to be able to register two codecs given in the device tree * v1 patch series at: https://lore.kernel.org/alsa-devel/20230901144550.520072-1-elinor.montmasso… Elinor Montmasson (10): ASoC: fsl-asoc-card: add support for dai links with multiple codecs ASoC: fsl-asoc-card: add second dai link component for codecs ASoC: fsl-asoc-card: add compatibility to use 2 codecs from device tree ASoC: fsl-asoc-card: add new compatible for a generic codec use case ASoC: fsl-asoc-card: set generic codec as clock provider ASoC: fsl-asoc-card: add dts property "cpu-slot-width" ASoC: fsl-asoc-card: add dts property "cpu-slot-num" ASoC: fsl-asoc-card: add dts properties "cpu-sysclk-freq" ASoC: fsl-asoc-card: add dts properties "cpu-sysclk-dir-out" Documentation: fsl-asoc-card: add documentation for generic codec case .../bindings/sound/fsl-asoc-card.txt | 26 +++- sound/soc/fsl/fsl-asoc-card.c | 128 ++++++++++++------ 2 files changed, 114 insertions(+), 40 deletions(-) -- 2.25.1

2 11