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

[PATCH alsa-ucm-conf v2 1/2] sof-soundwire: Add missing match for cs42l43 speakers
by Charles Keepax 08 Jul '24

08 Jul '24

The cs42l43 reports as cs42l43-spk in the speaker case, the regex in sof-soundwire needs updated to recognise that. Fixes: 035d9206cffd ("sof-soundwire: Add basic support for cs42l43's speaker") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- No change since v1. Thanks, Charles ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 31492e3..2f61d11 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -21,7 +21,7 @@ Define { DefineRegex { SpeakerCodec { - Regex " spk:([a-z0-9]+(-sdca)?)" + Regex " spk:([a-z0-9]+((-sdca)|(-spk))?)" String "${CardComponents}" } SpeakerChannels { -- 2.39.2

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[PATCH v6 5/7] ASoC: dt-bindings: update fsl-asoc-card bindings after imx-spdif merge
by Elinor Montmasson 08 Jul '24

08 Jul '24

The S/PDIF audio card support with compatible "fsl,imx-audio-spdif" was merged from imx-spdif into the fsl-asoc-card driver. It makes possible to use an S/PDIF with an ASRC. This merge introduces new DT bindings to use with compatible "fsl,imx-audio-spdif" to follow the way fsl-asoc-card works: * the "spdif-controller" property from imx-spdif is named "audio-cpu" in fsl-asoc-card. * fsl-asoc-card uses codecs explicitly declared in DT with "audio-codec". With an SPDIF, codec drivers spdif_transmitter and spdif_receiver should be used. Driver imx-spdif used instead the dummy codec and a pair of boolean properties, "spdif-in" and "spdif-out". In an upcoming commit, in-tree DTs will be modified to follow these new properties: * Property "spdif-controller" will be renamed "audio-cpu". * spdif_transmitter and spdif_receiver nodes will be declared and linked to the fsl-asoc-card node with the property "audio-codec". To keep backward compatibility with other DTs, support for "spdif-controller", "spdif-in" and "spdif-out" properties is kept. However, it is recommended to use the new properties if possible. It is better to declare transmitter and/or receiver in DT than using the dummy codec. DTs using compatible "fsl,imx-audio-spdif" are still supported, and fsl-asoc-card will behave the same as imx-spdif for these DTs. Signed-off-by: Elinor Montmasson <elinor.montmasson(a)savoirfairelinux.com> --- .../bindings/sound/fsl,imx-audio-spdif.yaml | 66 ------------------- .../bindings/sound/fsl-asoc-card.yaml | 53 +++++++++++++-- 2 files changed, 49 insertions(+), 70 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml diff --git a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml b/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml deleted file mode 100644 index 5fc543d02ecb..000000000000 --- a/Documentation/devicetree/bindings/sound/fsl,imx-audio-spdif.yaml +++ /dev/null @@ -1,66 +0,0 @@ -# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) -%YAML 1.2 ---- -$id: http://devicetree.org/schemas/sound/fsl,imx-audio-spdif.yaml# -$schema: http://devicetree.org/meta-schemas/core.yaml# - -title: Freescale i.MX audio complex with S/PDIF transceiver - -maintainers: - - Shengjiu Wang <shengjiu.wang(a)nxp.com> - -properties: - compatible: - oneOf: - - items: - - enum: - - fsl,imx-sabreauto-spdif - - fsl,imx6sx-sdb-spdif - - const: fsl,imx-audio-spdif - - enum: - - fsl,imx-audio-spdif - - model: - $ref: /schemas/types.yaml#/definitions/string - description: User specified audio sound card name - - spdif-controller: - $ref: /schemas/types.yaml#/definitions/phandle - description: The phandle of the i.MX S/PDIF controller - - spdif-out: - type: boolean - description: - If present, the transmitting function of S/PDIF will be enabled, - indicating there's a physical S/PDIF out connector or jack on the - board or it's connecting to some other IP block, such as an HDMI - encoder or display-controller. - - spdif-in: - type: boolean - description: - If present, the receiving function of S/PDIF will be enabled, - indicating there is a physical S/PDIF in connector/jack on the board. - -required: - - compatible - - model - - spdif-controller - -anyOf: - - required: - - spdif-in - - required: - - spdif-out - -additionalProperties: false - -examples: - - | - sound-spdif { - compatible = "fsl,imx-audio-spdif"; - model = "imx-spdif"; - spdif-controller = <&spdif>; - spdif-out; - spdif-in; - }; diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml index 9922664d5ccc..92aa47ec72c7 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.yaml @@ -65,6 +65,11 @@ properties: - fsl,imx-audio-sgtl5000 - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 + - items: + - enum: + - fsl,imx-sabreauto-spdif + - fsl,imx6sx-sdb-spdif + - const: fsl,imx-audio-spdif - items: - enum: - fsl,imx-audio-ac97 @@ -81,6 +86,7 @@ properties: - fsl,imx-audio-wm8960 - fsl,imx-audio-wm8962 - fsl,imx-audio-wm8958 + - fsl,imx-audio-spdif model: $ref: /schemas/types.yaml#/definitions/string @@ -93,8 +99,15 @@ 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. + With "fsl,imx-audio-spdif", either SPDIF audio codec spdif_transmitter, + spdif_receiver or both. + minItems: 1 + maxItems: 2 + items: + maxItems: 1 audio-cpu: $ref: /schemas/types.yaml#/definitions/phandle @@ -150,8 +163,10 @@ 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. + minItems: 1 + maxItems: 2 mux-int-port: $ref: /schemas/types.yaml#/definitions/uint32 @@ -167,6 +182,27 @@ properties: $ref: /schemas/types.yaml#/definitions/phandle description: The phandle of an CPU DAI controller + spdif-controller: + $ref: /schemas/types.yaml#/definitions/phandle + deprecated: true + description: The phandle of an S/PDIF CPU DAI controller. + + spdif-out: + type: boolean + deprecated: true + description: | + If present, the transmitting function of S/PDIF will be enabled, + indicating there's a physical S/PDIF out connector or jack on the + board or it's connecting to some other IP block, such as an HDMI + encoder or display-controller. + + spdif-in: + type: boolean + deprecated: true + description: | + If present, the receiving function of S/PDIF will be enabled, + indicating there is a physical S/PDIF in connector/jack on the board. + required: - compatible - model @@ -195,3 +231,12 @@ examples: "AIN2L", "Line In Jack", "AIN2R", "Line In Jack"; }; + + - | + sound-spdif-asrc { + compatible = "fsl,imx-audio-spdif"; + model = "spdif-asrc-audio"; + audio-cpu = <&spdif>; + audio-asrc = <&easrc>; + audio-codec = <&spdifdit>, <&spdifdir>; + }; -- 2.34.1

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[alsa-devel] unsupported variant of snd_hda_intel
by Alexandre 04 Jul '24

04 Jul '24

Hello, I have an Acer Aspire 4736G laptop. The soundcard that is shipped in this laptop is intel-hda based as shown (actually it's a realtek): cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 46 Basic sound output is correct but few feature are missing: - internal mic seems to be found but do not works. - sound output stop working when trying to send it using HDMI output - heads do not work when plugged neither when forcing pulseaudio to use it. I have an ubuntu 10.10 which is shipped with alsa 1.0.23. I have googled around and tried loading the snd-hda-intel with several parameters () as found in HD-Audio-Models.txt to no avail. cat /proc/asound/pcm 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 00-01: ALC888 Digital : ALC888 Digital : playback 1 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 Where can I begin to help implement the support for this sound card? I have very skills in developement in general but am ready to try if it is not to hard (I supose I can use previous code for similar models).. I just need a shepherd. Thanks

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[PATCH v2] slimbus: Fix struct and documentation alignment in stream.c
by Amit Vadhavana 03 Jul '24

03 Jul '24

The placement of the `segdist_codes` array documentation was corrected to conform with kernel documentation guidelines. The `@segdist_codes` was placed incorrectly within the struct `segdist_code` documentation block, which led to a potential misinterpretation of the code structure. The `segdist_codes` array documentation was moved outside the struct block, and a separate comment block was provided for it. This change ensures that clarity and proper alignment with kernel documentation standards are maintained. A kernel-doc warning was addressed: ./drivers/slimbus/stream.c:49: warning: Excess struct member 'segdist_codes' description in 'segdist_code' Signed-off-by: Amit Vadhavana <av2082000(a)gmail.com> --- Changes in v2: - Removed `static const` keyword from `segdist_code` structure declaration. drivers/slimbus/stream.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/drivers/slimbus/stream.c b/drivers/slimbus/stream.c index 1d6b38657917..863ab3075d7e 100644 --- a/drivers/slimbus/stream.c +++ b/drivers/slimbus/stream.c @@ -18,15 +18,17 @@ * and the first slot of the next consecutive Segment. * @segdist_code: Segment Distribution Code SD[11:0] * @seg_offset_mask: Segment offset mask in SD[11:0] - * @segdist_codes: List of all possible Segmet Distribution codes. */ -static const struct segdist_code { +struct segdist_code { int ratem; int seg_interval; int segdist_code; u32 seg_offset_mask; -} segdist_codes[] = { +}; + +/* segdist_codes - List of all possible Segment Distribution codes. */ +static const struct segdist_code segdist_codes[] = { {1, 1536, 0x200, 0xdff}, {2, 768, 0x100, 0xcff}, {4, 384, 0x080, 0xc7f}, -- 2.25.1

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[PATCH] ASoC: fsl_xcvr: Improve suspend/resume flow in fsl_xcvr_trigger()
by Chancel Liu 03 Jul '24

03 Jul '24

In the current flow all interrupts are disabled in runtime suspend phase. However interrupts enablement only exists in fsl_xcvr_prepare(). After resume fsl_xcvr_prepare() may not be called so it will cause all interrupts still disabled even if resume from suspend. Interrupts should be explictily enabled after resume. Also, DPATH reset setting only exists in fsl_xcvr_prepare(). After resume from suspend DPATH should be reset otherwise there'll be channel swap issue. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- sound/soc/fsl/fsl_xcvr.c | 43 +++++++++++++++++++++------------------- 1 file changed, 23 insertions(+), 20 deletions(-) diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index 337da46a2f90..bf9a4e90978e 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -529,16 +529,6 @@ static int fsl_xcvr_prepare(struct snd_pcm_substream *substream, break; } - ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, - FSL_XCVR_IRQ_EARC_ALL, FSL_XCVR_IRQ_EARC_ALL); - if (ret < 0) { - dev_err(dai->dev, "Error while setting IER0: %d\n", ret); - return ret; - } - - /* set DPATH RESET */ - m_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); - v_ctl |= FSL_XCVR_EXT_CTRL_DPTH_RESET(tx); ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, m_ctl, v_ctl); if (ret < 0) { dev_err(dai->dev, "Error while setting EXT_CTRL: %d\n", ret); @@ -679,6 +669,15 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* set DPATH RESET */ + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, + FSL_XCVR_EXT_CTRL_DPTH_RESET(tx), + FSL_XCVR_EXT_CTRL_DPTH_RESET(tx)); + if (ret < 0) { + dev_err(dai->dev, "Failed to set DPATH RESET: %d\n", ret); + return ret; + } + if (tx) { switch (xcvr->mode) { case FSL_XCVR_MODE_EARC: @@ -711,6 +710,13 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, + FSL_XCVR_IRQ_EARC_ALL, FSL_XCVR_IRQ_EARC_ALL); + if (ret < 0) { + dev_err(dai->dev, "Error while setting IER0: %d\n", ret); + return ret; + } + /* clear DPATH RESET */ ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, FSL_XCVR_EXT_CTRL_DPTH_RESET(tx), @@ -733,6 +739,13 @@ static int fsl_xcvr_trigger(struct snd_pcm_substream *substream, int cmd, return ret; } + ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, + FSL_XCVR_IRQ_EARC_ALL, 0); + if (ret < 0) { + dev_err(dai->dev, "Failed to clear IER0: %d\n", ret); + return ret; + } + if (tx) { switch (xcvr->mode) { case FSL_XCVR_MODE_SPDIF: @@ -1411,16 +1424,6 @@ static int fsl_xcvr_runtime_suspend(struct device *dev) struct fsl_xcvr *xcvr = dev_get_drvdata(dev); int ret; - /* - * Clear interrupts, when streams starts or resumes after - * suspend, interrupts are enabled in prepare(), so no need - * to enable interrupts in resume(). - */ - ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_IER0, - FSL_XCVR_IRQ_EARC_ALL, 0); - if (ret < 0) - dev_err(dev, "Failed to clear IER0: %d\n", ret); - if (!xcvr->soc_data->spdif_only) { /* Assert M0+ reset */ ret = regmap_update_bits(xcvr->regmap, FSL_XCVR_EXT_CTRL, -- 2.43.0

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[PATCH v2 0/8] iio: adc: dfsdm: add scaling support
by Olivier Moysan 03 Jul '24

03 Jul '24

The aim of this serie is to add scaling support to STM32 DFSDM peripheral in the analog context. The DFSDM currently operates as a consumer of IIO channels provided by a generic SD modulator. As previously discussed in RFC [1], this topology is not suitable for implementing scaling. This series brings the integration of the DFSDM driver with the new IIO backend framework [2], enabling the DFSDM IIO device to offer scaling feature based on reference voltage data obtained from the IIO SD modulator backend. This generic SD modulator backend takes the place of the former SD modulator, used with legacy implementation. The DFSDM driver has been updated to adopt the generic ADC channel binding [3]. The reasons for this include: - Reducing the use of proprietary properties - Simplifying the coexistence of legacy and new backend bindings - Prepare the support of the MDF peripheral on STM32MP25 SoC Backward compatibility is maintained through legacy support. This series extends the backend framework with the following APIs: - iio_backend_read_raw: This API is intented to retrieve the voltage information from the backend. It is based on IIO framework read_raw API. - iio_backend_disable / iio_backend_enable: backend enable/disable to be used for PM management - devm_iio_backend_fwnode_get Intended for parsing DT subnodes to allow generic channel binding support, as generic channel DT nodes are not populated as devices. [1]: https://lore.kernel.org/lkml/20200204101008.11411-1-olivier.moysan@st.com/ [2]: https://lore.kernel.org/all/20240206-iio-backend-v9-0-df66d159c000@analog.c… [3]: devicetree/bindings/iio/adc/adc.yaml Changes in v2: - Update enable/disable backend API - Rename devm_iio_backend_subnode_get(), as devm_iio_backend_fwnode_get() - Update iio_backend_read_raw() prototype to fully match IIO framework read_raw callback prototype. - Change st,adc-channel-type property name and type in DFSDM binding - Remove sd-backend and rename ads1201 compatibles in SD binding Conor, in this v2, I left the SD modulator driver & binding unchanged, regarding the naming issue you raised previously. The problem here is that we have two versions of the generic sigma delta modulator driver: one for legacy support and a new one to support new binding. Maybe an alternate, is to rename former sd modulator as "legacy" or something similar. I will address this point in a v3 if necessary. Olivier Moysan (8): iio: add read raw service to iio backend framework iio: add enable and disable services to iio backend framework iio: add child nodes support in iio backend framework dt-bindings: iio: dfsdm: move to backend framework dt-bindings: iio: add sigma delta modulator backend iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add sd modulator generic iio backend iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sd-modulator-backend.yaml | 39 +++ .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 157 ++++++++- drivers/iio/adc/Kconfig | 11 + drivers/iio/adc/Makefile | 1 + drivers/iio/adc/sd_adc_backend.c | 117 +++++++ drivers/iio/adc/stm32-dfsdm-adc.c | 302 +++++++++++++++--- drivers/iio/industrialio-backend.c | 108 +++++-- include/linux/iio/backend.h | 10 +- 8 files changed, 679 insertions(+), 66 deletions(-) create mode 100644 Documentation/devicetree/bindings/iio/adc/sd-modulator-backend.yaml create mode 100644 drivers/iio/adc/sd_adc_backend.c base-commit: 2dfa1b7bfc07e58acb9f9eaa8c871f37189dbfee -- 2.25.1

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[PATCH v1] ASoc: tas2781: Add name_prefix as the prefix name of DSP firmwares and calibrated data files
by Shenghao Ding 03 Jul '24

03 Jul '24

Add name_prefix as the prefix name of DSP firmwares and calibrated data files which stored speaker calibrated impedance. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 19 +++++++++++++++---- 1 file changed, 15 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 4d1a0d836e77..cc765d45c6b5 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -394,8 +394,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * failing to load DSP firmware is NOT an error. */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; - scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", - tas_priv->dev_name); + if (tas_priv->name_prefix) + scnprintf(tas_priv->rca_binaryname, 64, "%s-%s_coef.bin", + tas_priv->name_prefix, tas_priv->dev_name); + else + scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", + tas_priv->dev_name); ret = tasdevice_dsp_parser(tas_priv); if (ret) { dev_err(tas_priv->dev, "dspfw load %s error\n", @@ -418,8 +422,15 @@ static void tasdevice_fw_ready(const struct firmware *fmw, * calibrated data inside algo. */ for (i = 0; i < tas_priv->ndev; i++) { - scnprintf(tas_priv->cal_binaryname[i], 64, "%s_cal_0x%02x.bin", - tas_priv->dev_name, tas_priv->tasdevice[i].dev_addr); + if (tas_priv->name_prefix) + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s-%s_cal_0x%02x.bin", tas_priv->name_prefix, + tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); + else + scnprintf(tas_priv->cal_binaryname[i], 64, + "%s_cal_0x%02x.bin", tas_priv->dev_name, + tas_priv->tasdevice[i].dev_addr); ret = tas2781_load_calibration(tas_priv, tas_priv->cal_binaryname[i], i); if (ret != 0) -- 2.34.1

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[PATCH 1/2] ASoC: codecs: lpass-rx-macro: Fix using NULL pointer in probe() dev_err
by Krzysztof Kozlowski 02 Jul '24

02 Jul '24

The 'rx->dev' is assigned closer to the end of the probe() function, so the dev_err() must not use it - it is still NULL at this point. Instead there is already a local 'dev' variable. Fixes: dbacef05898d ("ASoC: codec: lpass-rx-macro: prepare driver to accomdate new codec versions") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/lpass-rx-macro.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 08308a63d1b7..8d4a61f7debc 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -3843,7 +3843,7 @@ static int rx_macro_probe(struct platform_device *pdev) rx_2_5_defaults, sizeof(rx_2_5_defaults)); break; default: - dev_err(rx->dev, "Unsupported Codec version (%d)\n", rx->codec_version); + dev_err(dev, "Unsupported Codec version (%d)\n", rx->codec_version); ret = -EINVAL; goto err; } -- 2.43.0

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[PATCH] drivers:soundwire: qcom: cleanup port maask calculations
by Srinivas Kandagatla 02 Jul '24

02 Jul '24

Cleanup the port map calculations, existing masks of having separate masks for in and out ports is not really required. Having a single mask for all the ports in the controller is simple and cuts of some unnecessary code. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 22 ++++++---------------- 1 file changed, 6 insertions(+), 16 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index ce5cf3ecceb5..aed57002fd0e 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -197,8 +197,7 @@ struct qcom_swrm_ctrl { int num_dout_ports; int cols_index; int rows_index; - unsigned long dout_port_mask; - unsigned long din_port_mask; + unsigned long port_mask; u32 intr_mask; u8 rcmd_id; u8 wcmd_id; @@ -1146,11 +1145,7 @@ static void qcom_swrm_stream_free_ports(struct qcom_swrm_ctrl *ctrl, mutex_lock(&ctrl->port_lock); list_for_each_entry(m_rt, &stream->master_list, stream_node) { - if (m_rt->direction == SDW_DATA_DIR_RX) - port_mask = &ctrl->dout_port_mask; - else - port_mask = &ctrl->din_port_mask; - + port_mask = &ctrl->port_mask; list_for_each_entry(p_rt, &m_rt->port_list, port_node) clear_bit(p_rt->num, port_mask); } @@ -1195,13 +1190,9 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, if (ctrl->bus.id != m_rt->bus->id) continue; - if (m_rt->direction == SDW_DATA_DIR_RX) { - maxport = ctrl->num_dout_ports; - port_mask = &ctrl->dout_port_mask; - } else { - maxport = ctrl->num_din_ports; - port_mask = &ctrl->din_port_mask; - } + port_mask = &ctrl->port_mask; + maxport = ctrl->num_dout_ports + ctrl->num_din_ports; + list_for_each_entry(s_rt, &m_rt->slave_rt_list, m_rt_node) { slave = s_rt->slave; @@ -1401,8 +1392,7 @@ static int qcom_swrm_get_port_config(struct qcom_swrm_ctrl *ctrl) return -EINVAL; /* Valid port numbers are from 1-14, so mask out port 0 explicitly */ - set_bit(0, &ctrl->dout_port_mask); - set_bit(0, &ctrl->din_port_mask); + set_bit(0, &ctrl->port_mask); ret = of_property_read_u8_array(np, "qcom,ports-offset1", off1, nports); --- base-commit: 1613e604df0cd359cf2a7fbd9be7a0bcfacfabd0 change-id: 20240618-soundwire-port-map-008f9dfae0f9 Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

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[Patch v4 00/10] Add audio support for LPC32XX CPUs
by Piotr Wojtaszczyk 02 Jul '24

02 Jul '24

This pach set is to bring back audio to machines with a LPC32XX CPU. The legacy LPC32XX SoC used to have audio spport in linux 2.6.27. The support was dropped due to lack of interest from mainaeners. Piotr Wojtaszczyk (10): dt-bindings: dma: pl08x: Add dma-cells description dt-bindings: dma: Add lpc32xx DMA mux binding ASoC: dt-bindings: lpc32xx: Add lpc32xx i2s DT binding ARM: dts: lpc32xx: Add missing dma and i2s properties clk: lpc32xx: initialize regmap using parent syscon dmaengine: Add dma router for pl08x in LPC32XX SoC ARM: lpc32xx: Remove pl08x platform data in favor for device tree mtd: rawnand: lpx32xx: Request DMA channels using DT entries ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs i2x: pnx: Use threaded irq to fix warning from del_timer_sync() .../devicetree/bindings/dma/arm-pl08x.yaml | 7 + .../bindings/dma/nxp,lpc3220-dmamux.yaml | 56 +++ .../bindings/sound/nxp,lpc3220-i2s.yaml | 73 ++++ MAINTAINERS | 21 + arch/arm/boot/dts/nxp/lpc/lpc32xx.dtsi | 53 ++- arch/arm/mach-lpc32xx/phy3250.c | 54 --- drivers/clk/Kconfig | 1 + drivers/clk/nxp/clk-lpc32xx.c | 10 +- drivers/dma/Kconfig | 9 + drivers/dma/Makefile | 1 + drivers/dma/lpc32xx-dmamux.c | 195 +++++++++ drivers/i2c/busses/i2c-pnx.c | 4 +- drivers/mtd/nand/raw/lpc32xx_mlc.c | 10 +- drivers/mtd/nand/raw/lpc32xx_slc.c | 10 +- sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/lpc3xxx-i2s.c | 376 ++++++++++++++++++ sound/soc/fsl/lpc3xxx-i2s.h | 79 ++++ sound/soc/fsl/lpc3xxx-pcm.c | 73 ++++ 19 files changed, 954 insertions(+), 87 deletions(-) create mode 100644 Documentation/devicetree/bindings/dma/nxp,lpc3220-dmamux.yaml create mode 100644 Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml create mode 100644 drivers/dma/lpc32xx-dmamux.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.c create mode 100644 sound/soc/fsl/lpc3xxx-i2s.h create mode 100644 sound/soc/fsl/lpc3xxx-pcm.c -- 2.25.1

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