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

[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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[PATCH 0/4] firmware: cs_dsp: Add check to prevent overrunning the firmware file
by Richard Fitzgerald 01 Jul '24

01 Jul '24

This series fixes various missing length checks when processing variable-length data from the firmware file. These could have caused overrunning the end of firmware file buffer, or wild pointer accesses. Richard Fitzgerald (4): firmware: cs_dsp: Fix overflow checking of wmfw header firmware: cs_dsp: Return error if block header overflows file firmware: cs_dsp: Validate payload length before processing block firmware: cs_dsp: Prevent buffer overrun when processing V2 alg headers drivers/firmware/cirrus/cs_dsp.c | 223 ++++++++++++++++++++++--------- 1 file changed, 160 insertions(+), 63 deletions(-) -- 2.39.2

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[PATCH 1/2] ASoC: dt-bindings: fsl_rpmsg: Add compatible string for i.MX95
by Chancel Liu 01 Jul '24

01 Jul '24

Add compatible string for i.MX95 platform which supports audio function through rpmsg channel between Cortex-A and Cortex-M core. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml index 188f38baddec..3d5d435c765b 100644 --- a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -29,6 +29,7 @@ properties: - fsl,imx8mp-rpmsg-audio - fsl,imx8ulp-rpmsg-audio - fsl,imx93-rpmsg-audio + - fsl,imx95-rpmsg-audio clocks: items: -- 2.43.0

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[PATCH 00/10] Add support for non-interleaved mode in qmc_audio
by Herve Codina 01 Jul '24

01 Jul '24

The qmc_audio driver supports only audio in interleaved mode. Non-interleaved mode can be easily supported using several QMC channel per DAI. In that case, data related to ch0 are sent to (received from) the first QMC channel, data related to ch1 use the next QMC channel and so on up to the last channel. In terms of constraints and settings, the interleaved and non-interleaved modes are slightly different. In interleaved mode: - The sample size should fit in the number of time-slots available for the QMC channel. - The number of audio channels should fit in the number of time-slots (taking into account the sample size) available for the QMC channel. In non-interleaved mode: - The number of audio channels is the number of available QMC channels. - Each QMC channel should have the same number of time-slots. - The sample size equals the number of time-slots of one QMC channel. This series add support for the non-interleaved mode in the qmc_audio driver and is composed of the following parts: - Patches 1 and 2: Fix some issues in the qmc_audio - Patches 3 to 6: Prepare qmc_audio for the non-interleaved mode - Patches 7 and 8: Extend the QMC driver API - Patches 9 and 10: The support for non-interleaved mode itself Best regards, Hervé Herve Codina (10): ASoC: fsl: fsl_qmc_audio: Check devm_kasprintf() returned value ASoC: fsl: fsl_qmc_audio: Fix issues detected by checkpatch ASoC: fsl: fsl_qmc_audio: Split channel buffer and PCM pointer handling ASoC: fsl: fsl_qmc_audio: Identify the QMC channel involved in completion routines ASoC: fsl: fsl_qmc_audio: Introduce qmc_audio_pcm_{read,write}_submit() ASoC: fsl: fsl_qmc_audio: Introduce qmc_dai_constraints_interleaved() soc: fsl: cpm1: qmc: Introduce functions to get a channel from a phandle list soc: fsl: cpm1: qmc: Introduce qmc_chan_count_phandles() dt-bindings: sound: fsl,qmc-audio: Add support for multiple QMC channels per DAI ASoC: fsl: fsl_qmc_audio: Add support for non-interleaved mode. .../bindings/sound/fsl,qmc-audio.yaml | 41 +- drivers/soc/fsl/qe/qmc.c | 32 +- include/soc/fsl/qe/qmc.h | 27 +- sound/soc/fsl/fsl_qmc_audio.c | 590 +++++++++++++----- 4 files changed, 505 insertions(+), 185 deletions(-) -- 2.45.0

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