May 2024 - Alsa-devel - mailman.alsa-project.org

Re: [PATCH] ASoC: codecs: ES8326: solve hp and button detect issue
by Mark Brown 15 May '24

15 May '24

On Wed, 15 May 2024 14:25:17 +0800, Zhang Yi wrote: > We got an error report about headphone type detection and button detection. > We fixed the headphone type detection error by adjusting the condition > of setting es8326->hp to 0.And we fixed the button detection error by > adjusting micbias and vref. > > Applied to https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next Thanks! [1/1] ASoC: codecs: ES8326: solve hp and button detect issue commit: 714f5df027b085c19c32af6f08a959bf35b9fb7c All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark

1 0

Ucm2 device sorting / ordering
by GitHub issues - edited 15 May '24

15 May '24

alsa-project/alsa-lib issue #397 was edited from beatboxa: The sort order that devices are probed and ultimately listed in my OS sound settings is reversed--though consistent--from the order that they appear (top to bottom) in the UCM configuration files. And they also do not appear consistent with configured device priority. In other words, if the UCM configuration file hypothetically lists the following, from top to bottom: 1. Device 1 - priority 300 2. Device 2 - priority 100 3. Device 3 - priority 200 These devices will be probed and listed in the OS in the following order: 1. Device 3 2. Device 2 3. Device 1 I am currently creating a UCM config file for a device with 32 discreet output channels and 30 discreet input channels (and both stereo and mono device versions of many of these), so the sort order is crucial for user experience in a list this long. I originally tried to reorder and rename the devices in the ucm configuration file to make it easier to follow and list correctly in my OS, but then I ran afoul of the alsa-lib rules for naming and ordering. Related: - https://github.com/alsa-project/alsa-ucm-conf/pull/416#issuecomment-2111101… - https://github.com/alsa-project/alsa-lib/issues/245#issuecomment-1168910073 Issue URL : https://github.com/alsa-project/alsa-lib/issues/397 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Ucm2 device sorting / ordering
by GitHub issues - opened 15 May '24

15 May '24

alsa-project/alsa-lib issue #397 was opened from beatboxa: The sort order that devices are probed and ultimately listed in my OS sound settings is reversed--though consisten--from the order that they appear (top to bottom) in the UCM configuration files. And they also do not appear consistent with configured device priority. In other words, if the UCM configuration file hypothetically lists the following, from top to bottom: 1. Device 1 - priority 300 2. Device 2 - priority 100 3. Device 3 - priority 200 These devices will be probed and listed in the OS in the following order: 1. Device 3 2. Device 2 3. Device 1 I am currently creating a UCM config file for a device with 32 discreet output channels and 30 discreet input channels (and both stereo and mono device versions of many of these), so the sort order is crucial for user experience in a list this long. Related: - https://github.com/alsa-project/alsa-ucm-conf/pull/416#issuecomment-2111101… - https://github.com/alsa-project/alsa-lib/issues/245#issuecomment-1168910073 Issue URL : https://github.com/alsa-project/alsa-lib/issues/397 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v15 00/16] Add audio support in v4l2 framework
by Shengjiu Wang 15 May '24

15 May '24

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 v15: - update MAINTAINERS for imx-asrc.c and vim2m-audio.c changes in v14: - document the reservation of 'AUXX' fourcc format. - add v4l2_audfmt_to_fourcc() definition. changes in v13 - change 'pixelformat' to 'audioformat' in dev-audio-mem2mem.rst - add more description for clock drift in ext-ctrls-audio-m2m.rst - Add "media: v4l2-ctrls: add support for fraction_bits" from Hans to avoid build issue for kernel test robot changes in v12 - minor changes according to comments - drop min_buffers_needed = 1 and V4L2_CTRL_FLAG_UPDATE flag - drop bus_info changes in v11 - add add-fixed-point-test-controls in vivid. - add v4l2_ctrl_fp_compose() helper function for min and max 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 Hans Verkuil (1): media: v4l2-ctrls: add support for fraction_bits Shengjiu Wang (15): 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: vivid: add fixed point test controls 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 | 59 + .../userspace-api/media/v4l/pixfmt-audio.rst | 100 ++ .../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/v4l/vidioc-queryctrl.rst | 11 +- .../media/videodev2.h.rst.exceptions | 3 + MAINTAINERS | 17 + .../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 | 1256 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 10 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/vim2m-audio.c | 793 +++++++++++ drivers/media/test-drivers/vivid/vivid-core.h | 2 + .../media/test-drivers/vivid/vivid-ctrls.c | 26 + drivers/media/v4l2-core/v4l2-compat-ioctl32.c | 9 + drivers/media/v4l2-core/v4l2-ctrls-api.c | 1 + drivers/media/v4l2-core/v4l2-ctrls-core.c | 93 +- 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-ctrls.h | 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 | 50 +- 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 +- 43 files changed, 3145 insertions(+), 27 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

10 45

[PATCHv4 9/9] ASoC: dt-bindings: fsl-asoc-card: add compatible for generic codec
by Elinor Montmasson 15 May '24

15 May '24

Add documentation about new dts bindings following new support for compatible "fsl,imx-audio-generic". Some CPU DAI don't require a real audio codec. The new compatible "fsl,imx-audio-generic" allows using the driver with codec drivers SPDIF DIT and SPDIF DIR as dummy codecs. It also allows using not pre-configured audio codecs which don't require specific control through a codec driver. The new dts properties give the possibility to set some parameters about the CPU DAI usually set through the codec configuration. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- .../bindings/sound/fsl-asoc-card.yaml | 96 ++++++++++++++++++- 1 file changed, 92 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index 9922664d5ccc..332d8bf96e06 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -23,6 +23,16 @@ description: and PCM DAI formats. However, it'll be also possible to support those non AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long as the driver has been properly upgraded. + To use CPU DAIs that do not require a codec such as an S/PDIF controller, + or to use a DAI to output or capture raw I2S/TDM data, you can + use the compatible "fsl,imx-audio-generic". + +definitions: + imx-audio-generic-dependency: + properties: + compatible: + contains: + const: fsl,imx-audio-generic maintainers: - Shengjiu Wang <shengjiu.wang(a)nxp.com> @@ -81,6 +91,7 @@ properties: - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 - fsl,imx-audio-wm8958 + - fsl,imx-audio-generic model: $ref: /schemas/types.yaml#/definitions/string @@ -93,8 +104,14 @@ properties: need to add ASRC support via DPCM. audio-codec: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of an audio codec + $ref: /schemas/types.yaml#/definitions/phandle-array + description: | + The phandle of an audio codec. + If using the "fsl,imx-audio-generic" compatible, give instead a pair of + phandles with the spdif_transmitter first (driver SPDIF DIT) and the + spdif_receiver second (driver SPDIF DIR). + items: + maxItems: 1 audio-cpu: $ref: /schemas/types.yaml#/definitions/phandle @@ -150,8 +167,8 @@ properties: description: dai-link uses bit clock inversion. mclk-id: - $ref: /schemas/types.yaml#/definitions/uint32 - description: main clock id, specific for each card configuration. + $ref: /schemas/types.yaml#/definitions/uint32-array + description: Main clock id for each codec, specific for each card configuration. mux-int-port: $ref: /schemas/types.yaml#/definitions/uint32 @@ -167,10 +184,68 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of an CPU DAI controller + # Properties relevant only with "fsl,imx-audio-generic" compatible + dai-tdm-slot-width: + description: See tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-num: + description: See tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + clocks: + description: | + The CPU DAI system clock, used to retrieve + the CPU DAI system clock frequency with the generic codec. + maxItems: 1 + + clock-names: + items: + - const: cpu_sysclk + + cpu-system-clock-direction-out: + description: | + Specifies cpu system clock direction as 'out' on initialization. + If not set, direction is 'in'. + $ref: /schemas/types.yaml#/definitions/flag + +dependencies: + dai-tdm-slot-width: + $ref: "#/definitions/imx-audio-generic-dependency" + dai-tdm-slot-num: + $ref: "#/definitions/imx-audio-generic-dependency" + clocks: + $ref: "#/definitions/imx-audio-generic-dependency" + cpu-system-clock-direction-out: + $ref: "#/definitions/imx-audio-generic-dependency" + required: - compatible - model +allOf: + - if: + $ref: "#/definitions/imx-audio-generic-dependency" + then: + properties: + audio-codec: + items: + - description: SPDIF DIT phandle + - description: SPDIF DIR phandle + mclk-id: + maxItems: 1 + items: + minItems: 1 + maxItems: 2 + else: + properties: + audio-codec: + maxItems: 1 + mclk-id: + maxItems: 1 + items: + maxItems: 1 + unevaluatedProperties: false examples: @@ -195,3 +270,16 @@ examples: "AIN2L", "Line In Jack", "AIN2R", "Line In Jack"; }; + + - | + #include <dt-bindings/clock/imx8mn-clock.h> + sound-spdif-asrc { + compatible = "fsl,imx-audio-generic"; + model = "spdif-asrc-audio"; + audio-cpu = <&spdif>; + audio-asrc = <&easrc>; + audio-codec = <&spdifdit>, <&spdifdir>; + clocks = <&clk IMX8MN_CLK_SAI5_ROOT>; + clock-names = "cpu_sysclk"; + cpu-system-clock-direction-out; + }; -- 2.34.1

1 0

[PATCHv4 8/9] ASoC: fsl-asoc-card: add DT property "cpu-system-clock-direction-out"
by Elinor Montmasson 15 May '24

15 May '24

Add new optional DT property "cpu-system-clock-direction-out" to set sysclk direction as "out" for the CPU DAI when using the generic codec. It is set for both Tx and Rx. If not set, the direction is "in". The way the direction value is used is up to the CPU DAI driver implementation. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index c7fc9c16f761..f3fc2b29c92f 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -760,6 +760,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_freq[RX] = priv->cpu_priv.sysclk_freq[TX]; clk_put(cpu_sysclk); } + priv->cpu_priv.sysclk_dir[TX] = + of_property_read_bool(np, "cpu-system-clock-direction-out") ? + SND_SOC_CLOCK_OUT : SND_SOC_CLOCK_IN; + priv->cpu_priv.sysclk_dir[RX] = priv->cpu_priv.sysclk_dir[TX]; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; -- 2.34.1

1 0

[PATCHv4 7/9] ASoC: fsl-asoc-card: add DT clock "cpu_sysclk" with generic codec
by Elinor Montmasson 15 May '24

15 May '24

Add an optional DT clock "cpu_sysclk" to get the CPU DAI system-clock frequency when using the generic codec. It is set for both Tx and Rx. The way the frequency value is used is up to the CPU DAI driver implementation. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 9aca8ad15372..c7fc9c16f761 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -754,6 +754,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) snd_soc_of_parse_tdm_slot(np, NULL, NULL, &priv->cpu_priv.slot_num, &priv->cpu_priv.slot_width); + struct clk *cpu_sysclk = clk_get(&pdev->dev, "cpu_sysclk"); + if (!IS_ERR(cpu_sysclk)) { + priv->cpu_priv.sysclk_freq[TX] = clk_get_rate(cpu_sysclk); + priv->cpu_priv.sysclk_freq[RX] = priv->cpu_priv.sysclk_freq[TX]; + clk_put(cpu_sysclk); + } } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; -- 2.34.1

1 0

[PATCHv4 6/9] ASoC: fsl-asoc-card: add use of devicetree TDM slot properties
by Elinor Montmasson 15 May '24

15 May '24

Add use of optional TDM slot properties "dai-tdm-slot-num" and "dai-tdm-slot-width" through snd_soc_of_parse_tdm_slot(). They allow setting a custom TDM slot width in bits and number of slots for the CPU DAI when using the generic codec. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 82ed7f4e81a1..9aca8ad15372 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -751,6 +751,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[0].num_codecs = 2; priv->dai_link[2].num_codecs = 2; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + snd_soc_of_parse_tdm_slot(np, NULL, NULL, + &priv->cpu_priv.slot_num, + &priv->cpu_priv.slot_width); } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; -- 2.34.1

1 0

[PATCHv4 5/9] ASoC: fsl-asoc-card: set generic codec as clock provider
by Elinor Montmasson 15 May '24

15 May '24

The default dai format defined by DAI_FMT_BASE doesn't set if the codec is consumer or provider of the bit and frame clocks. S/PDIF DIR usually converts audio signal to an asynchronous I2S/PCM stream, and doesn't consume a bit or frame clock. As S/PDIF DIR and DIT are used as codecs for the generic use case, set codecs as provider of both bit and frame clocks by default. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> Co-authored-by: Philip-Dylan Gleonec <philip-dylan.gleonec(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index a4ecc9093558..82ed7f4e81a1 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -750,6 +750,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) generic_codec_dai_names[1] = "dir-hifi"; priv->dai_link[0].num_codecs = 2; priv->dai_link[2].num_codecs = 2; + priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; -- 2.34.1

1 0

[PATCHv4 4/9] ASoC: fsl-asoc-card: add new compatible for a generic codec use case
by Elinor Montmasson 15 May '24

15 May '24

Add the new compatible "fsl,imx-audio-generic" for a generic codec use case. It allows using the fsl-asoc-card driver with the spdif_receiver and spdif_transmitter codec drivers used as dummy codecs. It can be used for cases where there is no real codec or codecs which do not require declaring controls. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> Co-authored-by: Philip-Dylan Gleonec <philip-dylan.gleonec(a)savoirfairelinux.com> --- sound/soc/fsl/fsl-asoc-card.c | 29 ++++++++++++++++++++++++++--- 1 file changed, 26 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 620a25eb068a..a4ecc9093558 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -567,6 +567,7 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct device *codec_dev[2] = { NULL, NULL }; + const char *generic_codec_dai_names[2]; const char *codec_dai_name; const char *codec_dev_name[2]; u32 asrc_fmt = 0; @@ -744,6 +745,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv[0].fll_id = WM8904_CLK_FLL; priv->codec_priv[0].pll_id = WM8904_FLL_MCLK; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-generic")) { + generic_codec_dai_names[0] = "dit-hifi"; + generic_codec_dai_names[1] = "dir-hifi"; + priv->dai_link[0].num_codecs = 2; + priv->dai_link[2].num_codecs = 2; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; @@ -798,6 +804,12 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) ret = -EPROBE_DEFER; goto asrc_fail; } + if (of_device_is_compatible(np, "fsl,imx-audio-generic") + && !codec_dev[1]) { + dev_dbg(&pdev->dev, "failed to find second codec device\n"); + ret = -EPROBE_DEFER; + goto asrc_fail; + } /* Common settings for corresponding Freescale CPU DAI driver */ if (of_node_name_eq(cpu_np, "ssi")) { @@ -855,11 +867,21 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpus->of_node = cpu_np; - priv->dai_link[0].codecs[0].dai_name = codec_dai_name; - if (!fsl_asoc_card_is_ac97(priv)) + if (of_device_is_compatible(np, "fsl,imx-audio-generic")) { + priv->dai_link[0].codecs[0].dai_name = + generic_codec_dai_names[0]; + priv->dai_link[0].codecs[1].dai_name = + generic_codec_dai_names[1]; + } else { + priv->dai_link[0].codecs[0].dai_name = codec_dai_name; + } + + if (!fsl_asoc_card_is_ac97(priv)) { priv->dai_link[0].codecs[0].of_node = codec_np[0]; - else { + if (of_device_is_compatible(np, "fsl,imx-audio-generic")) + priv->dai_link[0].codecs[1].of_node = codec_np[1]; + } else { u32 idx; ret = of_property_read_u32(cpu_np, "cell-index", &idx); @@ -990,6 +1012,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8958", }, { .compatible = "fsl,imx-audio-nau8822", }, { .compatible = "fsl,imx-audio-wm8904", }, + { .compatible = "fsl,imx-audio-generic", }, {} }; MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); -- 2.34.1

1 0