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

[PATCH 0/2] ASoC: cs35l56: Set correct upper volume limit
by Richard Fitzgerald 03 Jul '24

03 Jul '24

These two commits set the upper limit of the Speaker Volume control to +12dB instead of +100dB. This should have been a simple 1-line change to the #define in the header file, but only the HDA cs35l56 driver is using this define. The ASoC cs35l56 driver was using hardcoded numbers instead of the header defines. So the first commit changes the ASoC driver to use the #defined constants. The second commit corrects the value of the constant. Richard Fitzgerald (2): ASoC: cs35l56: Use header defines for Speaker Volume control definition ASoC: cs35l56: Limit Speaker Volume to +12dB maximum include/sound/cs35l56.h | 2 +- sound/soc/codecs/cs35l56.c | 6 +++++- 2 files changed, 6 insertions(+), 2 deletions(-) -- 2.39.2

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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 v3 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 channel attributes from the backend. It is based on IIO framework read_raw API. - iio_backend_read_scale / iio_backend_read_offset Dedicated APIs to read scale and offset channel values. - 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 Changes in v3: - Add iio_backend_read_scale and iio_backend_read_offset - Remove sd-backend compatible. Adapt existing sd modulator driver and binding to support backend topology. - Misc corrections in DFSDM binding & minor changes in DFSDM driver Olivier Moysan (8): iio: add read scale and offset services 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 backend support to sd modulator iio: adc: stm32-dfsdm: adopt generic channels bindings iio: add iio backend support to sd modulator iio: adc: stm32-dfsdm: add scaling support to dfsdm .../iio/adc/sigma-delta-modulator.yaml | 24 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 153 ++++++++- drivers/iio/adc/Kconfig | 2 + drivers/iio/adc/sd_adc_modulator.c | 92 +++++- drivers/iio/adc/stm32-dfsdm-adc.c | 298 +++++++++++++++--- drivers/iio/industrialio-backend.c | 111 ++++++- include/linux/iio/backend.h | 11 +- 7 files changed, 626 insertions(+), 65 deletions(-) base-commit: 529d2e1900642eba6df28307e26e19793e227546 -- 2.25.1

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[PATCH] ASoC: tas2781: Use of_property_read_reg()
by Rob Herring (Arm) 03 Jul '24

03 Jul '24

Replace the open-coded parsing of "reg" with of_property_read_reg(). The #ifdef is also easily replaced with IS_ENABLED(). Signed-off-by: Rob Herring (Arm) <robh(a)kernel.org> --- sound/soc/codecs/tas2781-i2c.c | 34 +++++++++++----------------------- 1 file changed, 11 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9350972dfefe..746e6964bfb3 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -21,6 +21,7 @@ #include <linux/interrupt.h> #include <linux/module.h> #include <linux/of.h> +#include <linux/of_address.h> #include <linux/of_gpio.h> #include <linux/of_irq.h> #include <linux/regmap.h> @@ -622,33 +623,20 @@ static void tasdevice_parse_dt(struct tasdevice_priv *tas_priv) tas_priv->irq_info.irq_gpio = acpi_dev_gpio_irq_get(ACPI_COMPANION(&client->dev), 0); - } else { + } else if (IS_ENABLED(CONFIG_OF)) { struct device_node *np = tas_priv->dev->of_node; -#ifdef CONFIG_OF - const __be32 *reg, *reg_end; - int len, sw, aw; - - aw = of_n_addr_cells(np); - sw = of_n_size_cells(np); - if (sw == 0) { - reg = (const __be32 *)of_get_property(np, - "reg", &len); - reg_end = reg + len/sizeof(*reg); - ndev = 0; - do { - dev_addrs[ndev] = of_read_number(reg, aw); - reg += aw; - ndev++; - } while (reg < reg_end); - } else { - ndev = 1; - dev_addrs[0] = client->addr; + u64 addr; + + for (i = 0; i < TASDEVICE_MAX_CHANNELS; i++) { + if (of_property_read_reg(np, i, &addr, NULL)) + break; + dev_addrs[ndev++] = addr; } -#else + + tas_priv->irq_info.irq_gpio = of_irq_get(np, 0); + } else { ndev = 1; dev_addrs[0] = client->addr; -#endif - tas_priv->irq_info.irq_gpio = of_irq_get(np, 0); } tas_priv->ndev = ndev; for (i = 0; i < ndev; i++) -- 2.43.0

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[PATCH 0/9] soundwire: simplify code with cleanup.h
by Krzysztof Kozlowski 03 Jul '24

03 Jul '24

Allocate the memory with scoped/cleanup.h to reduce error handling (simpler error paths) and make the code a bit smaller. Best regards, Krzysztof --- Krzysztof Kozlowski (9): soundwire: amd: simplify return path in hw_params soundwire: amd: simplify with cleanup.h soundwire: amd_init: simplify with cleanup.h soundwire: intel: simplify return path in hw_params soundwire: intel: simplify with cleanup.h soundwire: intel_ace2x: simplify return path in hw_params soundwire: intel_ace2x: simplify with cleanup.h soundwire: cadence: simplify with cleanup.h soundwire: debugfs: simplify with cleanup.h drivers/soundwire/amd_init.c | 7 +++---- drivers/soundwire/amd_manager.c | 13 +++++-------- drivers/soundwire/cadence_master.c | 5 ++--- drivers/soundwire/debugfs.c | 7 ++----- drivers/soundwire/intel.c | 25 +++++++++---------------- drivers/soundwire/intel_ace2x.c | 22 ++++++++-------------- 6 files changed, 29 insertions(+), 50 deletions(-) --- base-commit: 246cf92bafedeea46cb16964d16478d6b68f8e5a change-id: 20240703-soundwire-cleanup-h-a29941fb3dd4 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

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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 v3 0/2] Add master clock handling for nau8824
by Maxim Kochetkov 02 Jul '24

02 Jul '24

nau8824 has external MCLK pin. So add enable/disable external clock management. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- Changes in v3: - move bindings before their users - use devm_clk_get_optional() - let clock API care about NULL ptr Changes in v2: - move the devm_clk_get() to the nau8824_read_device() Maxim Kochetkov (2): ASoC: dt-bindings: nau8824: Add master clock handling ASoC: codecs: nau8824: Add master clock handling .../bindings/sound/nuvoton,nau8824.yaml | 8 +++++++ sound/soc/codecs/nau8824.c | 21 +++++++++++++++++-- sound/soc/codecs/nau8824.h | 1 + 3 files changed, 28 insertions(+), 2 deletions(-) -- 2.45.2

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[PATCH] firmware: cs_dsp: Don't allow writes to read-only controls
by Richard Fitzgerald 02 Jul '24

02 Jul '24

Add a check to cs_dsp_coeff_write_ctrl() to abort if the control is not writeable. The cs_dsp code originated as an ASoC driver (wm_adsp) where all controls were exported as ALSA controls. It relied on ALSA to enforce the read-only permission. Now that the code has been separated from ALSA/ASoC it must perform its own permission check. This isn't currently causing any problems so there shouldn't be any need to backport this. If the client of cs_dsp exposes the control as an ALSA control, it should set permissions on that ALSA control to protect it. The few uses of cs_dsp_coeff_write_ctrl() inside drivers are for writable controls. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- drivers/firmware/cirrus/cs_dsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index 16484ab9b09d..be8747d227db 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -796,6 +796,9 @@ int cs_dsp_coeff_write_ctrl(struct cs_dsp_coeff_ctl *ctl, lockdep_assert_held(&ctl->dsp->pwr_lock); + if (ctl->flags && !(ctl->flags & WMFW_CTL_FLAG_WRITEABLE)) + return -EPERM; + if (len + off * sizeof(u32) > ctl->len) return -EINVAL; -- 2.39.2

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[PATCH v2 0/6] ASoC: codecs: wsa88xx: add support for static port mapping.
by srinivas.kandagatla＠linaro.org 02 Jul '24

02 Jul '24

Existing way of allocating soundwire master ports on Qualcommm platforms is dynamic, and in linear order starting from 1 to MAX_PORTS. This will work as long as soundwire device ports are 1:1 mapped linearly. However on most Qcom SoCs like SM8550, SM8650, x1e80100, these are NOT mapped in that order. The result of this is that only one speaker among the pair of speakers is always silent, With recent changes for WSA codec to support codec versions and along with these patches we are able to get all speakers working on these SoCs. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes in v2: - used dev_dbg instead of dev_info - Link to v1: https://lore.kernel.org/r/20240626-port-map-v1-0-bd8987d2b332@linaro.org --- Manikantan R (1): ASoC: dt-bindings: wsa883x: Document port mapping property Srinivas Kandagatla (5): ASoC: codecs: wsa883x: parse port-mapping information ASoC: dt-bindings: wsa8840: Document port mapping property ASoC: codecs: wsa884x: parse port-mapping information arm64: dts: x1e80100-crd: fix wsa soundwire port mapping arm64: dts: x1e80100-qcp: fix wsa soundwire port mapping Documentation/devicetree/bindings/sound/qcom,wsa883x.yaml | 8 ++++++++ Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml | 8 ++++++++ arch/arm64/boot/dts/qcom/x1e80100-crd.dts | 4 ++++ arch/arm64/boot/dts/qcom/x1e80100-qcp.dts | 2 ++ sound/soc/codecs/wsa883x.c | 8 ++++++++ sound/soc/codecs/wsa884x.c | 8 ++++++++ 6 files changed, 38 insertions(+) --- base-commit: 9935be184a55dd84fc3275094f2df095491f6ea1 change-id: 20240626-port-map-ef50c3304d4a Best regards, -- Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org>

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