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[PATCH] ASoC: dt-bindings: davinci-mcasp: Fix interrupt properties
by Miquel Raynal 07 Oct '24

07 Oct '24

Combinations of "tx" alone, "rx" alone and "tx", "rx" together are supposedly valid (see link below), which is not the case today as "rx" alone is not accepted by the current binding. Let's rework the two interrupt properties to expose all correct possibilities. Cc: Péter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/linux-sound/20241003102552.2c11840e@xps-13/T/#m277f… Fixes: 8be90641a0bb ("ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema") Signed-off-by: Miquel Raynal <miquel.raynal(a)bootlin.com> --- Hello Mark, This patch applies on top of the one you already took, but if you prefer you can squash them together, I don't mind. Cheers, Miquèl --- .../bindings/sound/davinci-mcasp-audio.yaml | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml index ab3206ffa4af..beef193aaaeb 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -102,21 +102,21 @@ properties: default: 2 interrupts: - oneOf: - - minItems: 1 - items: - - description: TX interrupt - - description: RX interrupt - - items: - - description: common/combined interrupt + minItems: 1 + maxItems: 2 interrupt-names: oneOf: - - minItems: 1 + - description: TX interrupt + const: tx + - description: RX interrupt + const: rx + - description: TX and RX interrupts items: - const: tx - const: rx - - const: common + - description: Common/combined interrupt + const: common fck_parent: $ref: /schemas/types.yaml#/definitions/string -- 2.43.0

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Re: [v3?] ASoC: qcom: Fix NULL Dereference in asoc_qcom_lpass_cpu_platform_probe()
by Markus Elfring 07 Oct '24

07 Oct '24

… > --- > v2: Fix "From" tag. > v3: Format tags to Fixes/Cc/Signed-off-by. > --- … * How do you think about to reconsider the version numbers a bit more? * Would you like to mention the repeated adjustment of the patch subject? * Can a duplicate marker line be replaced by a blank line? See also: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/Doc… Is the email address “linux-kernel(a)vger.kernel.xn--org-9o0a still relevant here? Regards, Markus

2 1

Re: [PATCH v3] ASoC: qcom: Fix NULL Dereference in asoc_qcom_lpass_cpu_platform_probe()
by Markus Elfring 07 Oct '24

07 Oct '24

> A devm_kzalloc() in asoc_qcom_lpass_cpu_platform_probe() could call? > possibly return NULL pointer. NULL Pointer Dereference may be Can the term “null pointer dereference” be applied for the final commit message (including the summary phrase)? > triggerred without addtional check. triggered? additional? Regards, Markus

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[PATCH] ASoC: qcom: Fix NULL Dereference in asoc_qcom_lpass_cpu_platform_probe()
by Gax-c 07 Oct '24

07 Oct '24

A devm_kzalloc() in asoc_qcom_lpass_cpu_platform_probe() could possibly return NULL pointer. NULL Pointer Dereference may be triggerred without addtional check. Add a NULL check for the returned pointer. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Zichen Xie <zichenxie0106(a)gmail.com> Cc: stable(a)vger.kernel.org --- sound/soc/qcom/lpass-cpu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 5a47f661e0c6..242bc16da36d 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1242,6 +1242,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) /* Allocation for i2sctl regmap fields */ drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl), GFP_KERNEL); + if (!drvdata->i2sctl) + return -ENOMEM; /* Initialize bitfields for dai I2SCTL register */ ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl, -- 2.25.1

5 9

[PATCH v2 0/7] qrb4210-rb2: add HDMI audio playback support
by Alexey Klimov 06 Oct '24

06 Oct '24

Rebased on top of -master, tested. Changes since v1: -- removed handling of MI2S clock in sm2450_snd_shutdown(): setting clock rate and disabling it causes audio delay on playback start; -- removed empty sound { } node from sm6115.dtsi as suggested by Krzysztof; -- moved lpi_i2s2_active pins description to qrb423310 board-specific file as suggested by Dmitry Baryshkov; -- moved q6asmdai DAIs to apr soc node as suggested by Konrad Dybcio; -- lpass_tlmm is not disabled; -- lpass_tlmm node moved to sm4250.dtsi; -- kept MultiMedia DAIs as is, without them the sound card driver doesn't initialise; -- added some reviewed-by tags. This series still keeps "qcom,qrb4210-rb2-sndcard" for sm8250 soundcard. As per off the list discussion with Srini it was suggested to have it since in future it may be required to add clocks, workarounds, quirks, model-specific things based on this compatible. The same as for RB5 compatible in sm8250 snd driver. This focuses on HDMI audio playback only hence there are no soundwire and dmic pins, for instance. The work to enable playback via wcd+wsa8815 amplifier is in progress (it works) and one of the routes is to merge such two patchsets together. Link to prev series: https://lore.kernel.org/linux-sound/20240628010715.438471-1-alexey.klimov@l… Alexey Klimov (7): ASoC: dt-bindings: qcom,sm8250: add qrb4210-rb2-sndcard ASoC: qcom: sm8250: add qrb4210-rb2-sndcard compatible string ASoC: qcom: sm8250: add handling of secondary MI2S clock arm64: dts: qcom: sm6115: add apr and its services arm64: dts: qcom: sm4250: add LPASS LPI pin controller arm64: dts: qcom: qrb4210-rb2: add description of lpi_i2s2 pins arm64: dts: qcom: qrb4210-rb2: add HDMI audio playback support .../bindings/sound/qcom,sm8250.yaml | 1 + arch/arm64/boot/dts/qcom/qrb4210-rb2.dts | 91 +++++++++++++++++++ arch/arm64/boot/dts/qcom/sm4250.dtsi | 16 ++++ arch/arm64/boot/dts/qcom/sm6115.dtsi | 72 +++++++++++++++ sound/soc/qcom/sm8250.c | 9 ++ 5 files changed, 189 insertions(+) -- 2.45.2

5 15

Ubuntu 24.04 microphone is not working USB-Audio: ALC4080 Z-590E
by GitHub issues - edited 05 Oct '24

05 Oct '24

alsa-project/alsa-ucm-conf issue #447 was edited from jiutian12: My rear microphone is not working on Ubuntu 24.04 (nor on Ubuntu 22), but it can be used normally on Windows 10. Can hear sound After updating the master branch, the microphone still doesn't work, but the response is the same as the sound system Initial feedback on the issue, the following is my hardware information, I am not sure if it is complete http://alsa-project.org/db/?f=2558f03ffd8c55e18f8208a806f3e2336d9fa49c Kernel: k6.8.0-45-generic USB-Audio: ALC4080 ROG STRIX Z590-E This command can be recorded, but the sound is very low： arecord -D plughw:1,0 -c 1 -r 16000 -f S16_LE output.wav Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/447 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Ubuntu 24.04 microphone is not working USB-Audio: ALC4080 Z-590E
by GitHub issues - edited 05 Oct '24

05 Oct '24

alsa-project/alsa-ucm-conf issue #447 was edited from jiutian12: My rear microphone is not working on Ubuntu 24.04 (nor on Ubuntu 22), but it can be used normally on Windows 10. Can hear sound After updating the master branch, the microphone still doesn't work, but the response is the same as the sound system Initial feedback on the issue, the following is my hardware information, I am not sure if it is complete http://alsa-project.org/db/?f=2558f03ffd8c55e18f8208a806f3e2336d9fa49c Kernel: k6.8.0-45-generic USB-Audio: ALC4080 ROG STRIX Z590-E Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/447 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Ubuntu 24.04 microphone is not working USB-Audio: ALC4080 Z-590p
by GitHub issues - edited 05 Oct '24

05 Oct '24

alsa-project/alsa-ucm-conf issue #447 was edited from jiutian12: My rear microphone is not working on Ubuntu 24.04 (nor on Ubuntu 22), but it can be used normally on Windows 10. Can hear sound After updating the master branch, the microphone still doesn't work, but the response is the same as the sound system Initial feedback on the issue, the following is my hardware information, I am not sure if it is complete http://alsa-project.org/db/?f=2558f03ffd8c55e18f8208a806f3e2336d9fa49c Kernel: k6.8.0-45-generic USB-Audio: ALC4080 ROG STRIX Z590-E Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/447 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Ubuntu 24.04 microphone is not working USB-Audio: ALC4080 Z-590p
by GitHub issues - opened 05 Oct '24

05 Oct '24

alsa-project/alsa-ucm-conf issue #447 was opened from jiutian12: My rear microphone is not working on Ubuntu 24.04 (nor on Ubuntu 22), but it can be used normally on Windows 10. Can hear sound After updating the master branch, the microphone still doesn't work, but the response is the same as the sound system Initial feedback on the issue, the following is my hardware information, I am not sure if it is complete http://alsa-project.org/db/?f=2558f03ffd8c55e18f8208a806f3e2336d9fa49c Kernel: k6.8.0-45-generic USB-Audio: ALC4080 asus Z-590p Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/447 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param()
by (학생) 장인규 (컴퓨터공학과) 05 Oct '24

05 Oct '24

From 791716bf359b8540c519810848fd1f8006d7c3c5 Mon Sep 17 00:00:00 2001 From: Ingyu Jang <ingyujang25(a)unist.ac.kr> Date: Thu, 26 Sep 2024 19:40:04 +0900 Subject: [PATCH] sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param() The aio_src_set_param() function did not previously check the return values of regmap_write() and regmap_update_bits(). If these functions fail, it could lead to silent failures when configuring the sample rate converter (SRC), causing improper behavior in audio processing without any indication of an error. This patch modifies aio_src_set_param to check the return values of regmap_write() and regmap_update_bits(). If either function returns an error, the error code is propagated back to the caller to ensure proper error handling. This change aligns with the existing error-handling behavior in functions like uniphier_aio_prepare(), where a failure in a sub-function should result in an immediate return of the error. Signed-off-by: Ingyu Jang <ingyujang25(a)unist.ac.kr> --- sound/soc/uniphier/aio-core.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 0eba60758134..2c4e8b873253 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -921,16 +921,19 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, { struct regmap *r = sub->aio->chip->regmap; u32 v; + int ret; if (sub->swm->dir != PORT_DIR_OUTPUT) return 0; - regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), + ret = regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), OPORTMXSRC1CTR_THMODE_SRC | OPORTMXSRC1CTR_SRCPATH_CALC | OPORTMXSRC1CTR_SYNC_ASYNC | OPORTMXSRC1CTR_FSIIPSEL_INNER | OPORTMXSRC1CTR_FSISEL_ACLK); + if (ret) + return ret; switch (params_rate(params)) { default: @@ -951,12 +954,18 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, break; } - regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), + + ret = regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), v | OPORTMXRATE_I_ACLKSRC_APLL | OPORTMXRATE_I_LRCKSTP_STOP); - regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), + if (ret) + return ret; + + ret = regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), OPORTMXRATE_I_LRCKSTP_MASK, OPORTMXRATE_I_LRCKSTP_START); + if (ret) + return ret; return 0; } -- 2.34.1

2 2

[PATCH] Fix unsigned int compared against 0
by Advait Dhamorikar 05 Oct '24

05 Oct '24

An unsigned value held by offset can never be negative, so this test will always evaluate the same way and is therefore redundant. Signed-off-by: Advait Dhamorikar <advaitdhamorikar(a)gmail.com> --- sound/soc/codecs/tlv320adc3xxx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index 7073b9d1cda8..868e8a91e05b 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -961,7 +961,7 @@ static int adc3xxx_gpio_request(struct gpio_chip *chip, unsigned int offset) if (offset >= ADC3XXX_GPIOS_MAX) return -EINVAL; - if (offset >= 0 && offset < ADC3XXX_GPIO_PINS) { + if (offset < ADC3XXX_GPIO_PINS) { /* GPIO1 is offset 0, GPIO2 is offset 1 */ /* We check here that the GPIO pins are either not configured * in the DT, or that they purposely are set as outputs. -- 2.34.1

2 2

[PATCH] ASoC: fsl: fsl_qmc_audio: Remove the logging when parsing channels
by Herve Codina 05 Oct '24

05 Oct '24

On each channel parsing, a log message is issued. This log message is not needed and become annoying when many channels are used (up to 64 channel supported). Simply remove this unneeded log message. Signed-off-by: Herve Codina <herve.codina(a)bootlin.com> --- sound/soc/fsl/fsl_qmc_audio.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c index 8668abd35208..e257b8adafe0 100644 --- a/sound/soc/fsl/fsl_qmc_audio.c +++ b/sound/soc/fsl/fsl_qmc_audio.c @@ -838,8 +838,6 @@ static int qmc_audio_dai_parse(struct qmc_audio *qmc_audio, struct device_node * qmc_dai->id, i, ret); return ret; } - dev_info(qmc_audio->dev, "dai %d QMC channel %d mode %d, nb_tx_ts %u, nb_rx_ts %u\n", - qmc_dai->id, i, info.mode, info.nb_tx_ts, info.nb_rx_ts); if (info.mode != QMC_TRANSPARENT) { dev_err(qmc_audio->dev, "dai %d QMC chan %d mode %d is not QMC_TRANSPARENT\n", -- 2.46.1

2 1

[PATCH AUTOSEL 6.6 34/58] soundwire: cadence: re-check Peripheral status with delayed_work
by Sasha Levin 04 Oct '24

04 Oct '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit f8c35d61ba01afa76846905c67862cdace7f66b0 ] The SoundWire peripheral enumeration is entirely based on interrupts, more specifically sticky bits tracking state changes. This patch adds a defensive programming check on the actual status reported in PING frames. If for some reason an interrupt was lost or delayed, the delayed work would detect a peripheral change of status after the bus starts. The 100ms defined for the delay is not completely arbitrary, if a Peripheral didn't join the bus within that delay then probably the hardware link is broken, and conversely if the detection didn't happen because of software issues the 100ms is still acceptable in terms of user experience. The overhead of the one-shot workqueue is minimal, and the mutual exclusion ensures that the interrupt and delayed work cannot update the status concurrently. Reviewed-by: Liam Girdwood <liam.r.girdwood(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20240805114921.88007-1-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 39 ++++++++++++++++++++++++++-- drivers/soundwire/cadence_master.h | 5 ++++ drivers/soundwire/intel.h | 2 ++ drivers/soundwire/intel_auxdevice.c | 1 + drivers/soundwire/intel_bus_common.c | 11 ++++++++ 5 files changed, 56 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 3e7cf04aaf2a6..e69982dbd449b 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -891,8 +891,14 @@ static int cdns_update_slave_status(struct sdw_cdns *cdns, } } - if (is_slave) - return sdw_handle_slave_status(&cdns->bus, status); + if (is_slave) { + int ret; + + mutex_lock(&cdns->status_update_lock); + ret = sdw_handle_slave_status(&cdns->bus, status); + mutex_unlock(&cdns->status_update_lock); + return ret; + } return 0; } @@ -989,6 +995,31 @@ irqreturn_t sdw_cdns_irq(int irq, void *dev_id) } EXPORT_SYMBOL(sdw_cdns_irq); +static void cdns_check_attached_status_dwork(struct work_struct *work) +{ + struct sdw_cdns *cdns = + container_of(work, struct sdw_cdns, attach_dwork.work); + enum sdw_slave_status status[SDW_MAX_DEVICES + 1]; + u32 val; + int ret; + int i; + + val = cdns_readl(cdns, CDNS_MCP_SLAVE_STAT); + + for (i = 0; i <= SDW_MAX_DEVICES; i++) { + status[i] = val & 0x3; + if (status[i]) + dev_dbg(cdns->dev, "Peripheral %d status: %d\n", i, status[i]); + val >>= 2; + } + + mutex_lock(&cdns->status_update_lock); + ret = sdw_handle_slave_status(&cdns->bus, status); + mutex_unlock(&cdns->status_update_lock); + if (ret < 0) + dev_err(cdns->dev, "%s: sdw_handle_slave_status failed: %d\n", __func__, ret); +} + /** * cdns_update_slave_status_work - update slave status in a work since we will need to handle * other interrupts eg. CDNS_MCP_INT_RX_WL during the update slave @@ -1745,7 +1776,11 @@ int sdw_cdns_probe(struct sdw_cdns *cdns) init_completion(&cdns->tx_complete); cdns->bus.port_ops = &cdns_port_ops; + mutex_init(&cdns->status_update_lock); + INIT_WORK(&cdns->work, cdns_update_slave_status_work); + INIT_DELAYED_WORK(&cdns->attach_dwork, cdns_check_attached_status_dwork); + return 0; } EXPORT_SYMBOL(sdw_cdns_probe); diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index bc84435e420f5..e1d7969ba48ae 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -117,6 +117,8 @@ struct sdw_cdns_dai_runtime { * @link_up: Link status * @msg_count: Messages sent on bus * @dai_runtime_array: runtime context for each allocated DAI. + * @status_update_lock: protect concurrency between interrupt-based and delayed work + * status update */ struct sdw_cdns { struct device *dev; @@ -148,10 +150,13 @@ struct sdw_cdns { bool interrupt_enabled; struct work_struct work; + struct delayed_work attach_dwork; struct list_head list; struct sdw_cdns_dai_runtime **dai_runtime_array; + + struct mutex status_update_lock; /* add mutual exclusion to sdw_handle_slave_status() */ }; #define bus_to_cdns(_bus) container_of(_bus, struct sdw_cdns, bus) diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index 511932c55216c..bb6b1df2d2c20 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -91,6 +91,8 @@ static inline void intel_writew(void __iomem *base, int offset, u16 value) #define INTEL_MASTER_RESET_ITERATIONS 10 +#define SDW_INTEL_DELAYED_ENUMERATION_MS 100 + #define SDW_INTEL_CHECK_OPS(sdw, cb) ((sdw) && (sdw)->link_res && (sdw)->link_res->hw_ops && \ (sdw)->link_res->hw_ops->cb) #define SDW_INTEL_OPS(sdw, cb) ((sdw)->link_res->hw_ops->cb) diff --git a/drivers/soundwire/intel_auxdevice.c b/drivers/soundwire/intel_auxdevice.c index 93698532deac4..bdfff78ac2f81 100644 --- a/drivers/soundwire/intel_auxdevice.c +++ b/drivers/soundwire/intel_auxdevice.c @@ -398,6 +398,7 @@ static void intel_link_remove(struct auxiliary_device *auxdev) */ if (!bus->prop.hw_disabled) { sdw_intel_debugfs_exit(sdw); + cancel_delayed_work_sync(&cdns->attach_dwork); sdw_cdns_enable_interrupt(cdns, false); } sdw_bus_master_delete(bus); diff --git a/drivers/soundwire/intel_bus_common.c b/drivers/soundwire/intel_bus_common.c index 179aa0d85951b..db9cf211671a3 100644 --- a/drivers/soundwire/intel_bus_common.c +++ b/drivers/soundwire/intel_bus_common.c @@ -60,6 +60,9 @@ int intel_start_bus(struct sdw_intel *sdw) sdw_cdns_check_self_clearing_bits(cdns, __func__, true, INTEL_MASTER_RESET_ITERATIONS); + schedule_delayed_work(&cdns->attach_dwork, + msecs_to_jiffies(SDW_INTEL_DELAYED_ENUMERATION_MS)); + return 0; } @@ -151,6 +154,9 @@ int intel_start_bus_after_reset(struct sdw_intel *sdw) } sdw_cdns_check_self_clearing_bits(cdns, __func__, true, INTEL_MASTER_RESET_ITERATIONS); + schedule_delayed_work(&cdns->attach_dwork, + msecs_to_jiffies(SDW_INTEL_DELAYED_ENUMERATION_MS)); + return 0; } @@ -184,6 +190,9 @@ int intel_start_bus_after_clock_stop(struct sdw_intel *sdw) sdw_cdns_check_self_clearing_bits(cdns, __func__, true, INTEL_MASTER_RESET_ITERATIONS); + schedule_delayed_work(&cdns->attach_dwork, + msecs_to_jiffies(SDW_INTEL_DELAYED_ENUMERATION_MS)); + return 0; } @@ -194,6 +203,8 @@ int intel_stop_bus(struct sdw_intel *sdw, bool clock_stop) bool wake_enable = false; int ret; + cancel_delayed_work_sync(&cdns->attach_dwork); + if (clock_stop) { ret = sdw_cdns_clock_stop(cdns, true); if (ret < 0) -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 20/58] soundwire: intel_bus_common: enable interrupts before exiting reset
by Sasha Levin 04 Oct '24

04 Oct '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 5aedb8d8336b0a0421b58ca27d1b572aa6695b5b ] The existing code enables the Cadence IP interrupts after the bus reset sequence. The problem with this sequence is that it might be pre-empted, giving SoundWire devices time to sync and report as ATTACHED before the interrupts are enabled. In that case, the Cadence IP will not detect a state change and will not throw an interrupt to proceed with the enumeration of a Device0. In our overnight stress tests, we observed that a slight sub-millisecond delay in enabling interrupts after the reset was correlated with detection failures. This problem is more prevalent on the LunarLake silicon, likely due to SOC integration changes, but it was observed on earlier generations as well. This patch reverts the sequence, with the interrupts enabled before performing the bus reset. This removes the race condition and makes sure the Cadence IP is able to detect the presence of a Device0 in all cases. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20240805115003.88035-1-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/intel_bus_common.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/drivers/soundwire/intel_bus_common.c b/drivers/soundwire/intel_bus_common.c index e5ac3cc7cb79b..179aa0d85951b 100644 --- a/drivers/soundwire/intel_bus_common.c +++ b/drivers/soundwire/intel_bus_common.c @@ -45,15 +45,15 @@ int intel_start_bus(struct sdw_intel *sdw) return ret; } - ret = sdw_cdns_exit_reset(cdns); + ret = sdw_cdns_enable_interrupt(cdns, true); if (ret < 0) { - dev_err(dev, "%s: unable to exit bus reset sequence: %d\n", __func__, ret); + dev_err(dev, "%s: cannot enable interrupts: %d\n", __func__, ret); return ret; } - ret = sdw_cdns_enable_interrupt(cdns, true); + ret = sdw_cdns_exit_reset(cdns); if (ret < 0) { - dev_err(dev, "%s: cannot enable interrupts: %d\n", __func__, ret); + dev_err(dev, "%s: unable to exit bus reset sequence: %d\n", __func__, ret); return ret; } @@ -136,15 +136,15 @@ int intel_start_bus_after_reset(struct sdw_intel *sdw) return ret; } - ret = sdw_cdns_exit_reset(cdns); + ret = sdw_cdns_enable_interrupt(cdns, true); if (ret < 0) { - dev_err(dev, "unable to exit bus reset sequence during resume\n"); + dev_err(dev, "cannot enable interrupts during resume\n"); return ret; } - ret = sdw_cdns_enable_interrupt(cdns, true); + ret = sdw_cdns_exit_reset(cdns); if (ret < 0) { - dev_err(dev, "cannot enable interrupts during resume\n"); + dev_err(dev, "unable to exit bus reset sequence during resume\n"); return ret; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 39/70] soundwire: cadence: re-check Peripheral status with delayed_work
by Sasha Levin 04 Oct '24

04 Oct '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit f8c35d61ba01afa76846905c67862cdace7f66b0 ] The SoundWire peripheral enumeration is entirely based on interrupts, more specifically sticky bits tracking state changes. This patch adds a defensive programming check on the actual status reported in PING frames. If for some reason an interrupt was lost or delayed, the delayed work would detect a peripheral change of status after the bus starts. The 100ms defined for the delay is not completely arbitrary, if a Peripheral didn't join the bus within that delay then probably the hardware link is broken, and conversely if the detection didn't happen because of software issues the 100ms is still acceptable in terms of user experience. The overhead of the one-shot workqueue is minimal, and the mutual exclusion ensures that the interrupt and delayed work cannot update the status concurrently. Reviewed-by: Liam Girdwood <liam.r.girdwood(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20240805114921.88007-1-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 39 ++++++++++++++++++++++++++-- drivers/soundwire/cadence_master.h | 5 ++++ drivers/soundwire/intel.h | 2 ++ drivers/soundwire/intel_auxdevice.c | 1 + drivers/soundwire/intel_bus_common.c | 11 ++++++++ 5 files changed, 56 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index 74da99034dab5..ec08e26cef829 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -891,8 +891,14 @@ static int cdns_update_slave_status(struct sdw_cdns *cdns, } } - if (is_slave) - return sdw_handle_slave_status(&cdns->bus, status); + if (is_slave) { + int ret; + + mutex_lock(&cdns->status_update_lock); + ret = sdw_handle_slave_status(&cdns->bus, status); + mutex_unlock(&cdns->status_update_lock); + return ret; + } return 0; } @@ -989,6 +995,31 @@ irqreturn_t sdw_cdns_irq(int irq, void *dev_id) } EXPORT_SYMBOL(sdw_cdns_irq); +static void cdns_check_attached_status_dwork(struct work_struct *work) +{ + struct sdw_cdns *cdns = + container_of(work, struct sdw_cdns, attach_dwork.work); + enum sdw_slave_status status[SDW_MAX_DEVICES + 1]; + u32 val; + int ret; + int i; + + val = cdns_readl(cdns, CDNS_MCP_SLAVE_STAT); + + for (i = 0; i <= SDW_MAX_DEVICES; i++) { + status[i] = val & 0x3; + if (status[i]) + dev_dbg(cdns->dev, "Peripheral %d status: %d\n", i, status[i]); + val >>= 2; + } + + mutex_lock(&cdns->status_update_lock); + ret = sdw_handle_slave_status(&cdns->bus, status); + mutex_unlock(&cdns->status_update_lock); + if (ret < 0) + dev_err(cdns->dev, "%s: sdw_handle_slave_status failed: %d\n", __func__, ret); +} + /** * cdns_update_slave_status_work - update slave status in a work since we will need to handle * other interrupts eg. CDNS_MCP_INT_RX_WL during the update slave @@ -1741,7 +1772,11 @@ int sdw_cdns_probe(struct sdw_cdns *cdns) init_completion(&cdns->tx_complete); cdns->bus.port_ops = &cdns_port_ops; + mutex_init(&cdns->status_update_lock); + INIT_WORK(&cdns->work, cdns_update_slave_status_work); + INIT_DELAYED_WORK(&cdns->attach_dwork, cdns_check_attached_status_dwork); + return 0; } EXPORT_SYMBOL(sdw_cdns_probe); diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index bc84435e420f5..e1d7969ba48ae 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -117,6 +117,8 @@ struct sdw_cdns_dai_runtime { * @link_up: Link status * @msg_count: Messages sent on bus * @dai_runtime_array: runtime context for each allocated DAI. + * @status_update_lock: protect concurrency between interrupt-based and delayed work + * status update */ struct sdw_cdns { struct device *dev; @@ -148,10 +150,13 @@ struct sdw_cdns { bool interrupt_enabled; struct work_struct work; + struct delayed_work attach_dwork; struct list_head list; struct sdw_cdns_dai_runtime **dai_runtime_array; + + struct mutex status_update_lock; /* add mutual exclusion to sdw_handle_slave_status() */ }; #define bus_to_cdns(_bus) container_of(_bus, struct sdw_cdns, bus) diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index b68e74c294e70..c502dfdfe8abf 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -98,6 +98,8 @@ static inline void intel_writew(void __iomem *base, int offset, u16 value) #define INTEL_MASTER_RESET_ITERATIONS 10 +#define SDW_INTEL_DELAYED_ENUMERATION_MS 100 + #define SDW_INTEL_CHECK_OPS(sdw, cb) ((sdw) && (sdw)->link_res && (sdw)->link_res->hw_ops && \ (sdw)->link_res->hw_ops->cb) #define SDW_INTEL_OPS(sdw, cb) ((sdw)->link_res->hw_ops->cb) diff --git a/drivers/soundwire/intel_auxdevice.c b/drivers/soundwire/intel_auxdevice.c index 18517121cc898..58f6b4f33cabd 100644 --- a/drivers/soundwire/intel_auxdevice.c +++ b/drivers/soundwire/intel_auxdevice.c @@ -433,6 +433,7 @@ static void intel_link_remove(struct auxiliary_device *auxdev) */ if (!bus->prop.hw_disabled) { sdw_intel_debugfs_exit(sdw); + cancel_delayed_work_sync(&cdns->attach_dwork); sdw_cdns_enable_interrupt(cdns, false); } sdw_bus_master_delete(bus); diff --git a/drivers/soundwire/intel_bus_common.c b/drivers/soundwire/intel_bus_common.c index 179aa0d85951b..db9cf211671a3 100644 --- a/drivers/soundwire/intel_bus_common.c +++ b/drivers/soundwire/intel_bus_common.c @@ -60,6 +60,9 @@ int intel_start_bus(struct sdw_intel *sdw) sdw_cdns_check_self_clearing_bits(cdns, __func__, true, INTEL_MASTER_RESET_ITERATIONS); + schedule_delayed_work(&cdns->attach_dwork, + msecs_to_jiffies(SDW_INTEL_DELAYED_ENUMERATION_MS)); + return 0; } @@ -151,6 +154,9 @@ int intel_start_bus_after_reset(struct sdw_intel *sdw) } sdw_cdns_check_self_clearing_bits(cdns, __func__, true, INTEL_MASTER_RESET_ITERATIONS); + schedule_delayed_work(&cdns->attach_dwork, + msecs_to_jiffies(SDW_INTEL_DELAYED_ENUMERATION_MS)); + return 0; } @@ -184,6 +190,9 @@ int intel_start_bus_after_clock_stop(struct sdw_intel *sdw) sdw_cdns_check_self_clearing_bits(cdns, __func__, true, INTEL_MASTER_RESET_ITERATIONS); + schedule_delayed_work(&cdns->attach_dwork, + msecs_to_jiffies(SDW_INTEL_DELAYED_ENUMERATION_MS)); + return 0; } @@ -194,6 +203,8 @@ int intel_stop_bus(struct sdw_intel *sdw, bool clock_stop) bool wake_enable = false; int ret; + cancel_delayed_work_sync(&cdns->attach_dwork); + if (clock_stop) { ret = sdw_cdns_clock_stop(cdns, true); if (ret < 0) -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 22/70] soundwire: intel_bus_common: enable interrupts before exiting reset
by Sasha Levin 04 Oct '24

04 Oct '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 5aedb8d8336b0a0421b58ca27d1b572aa6695b5b ] The existing code enables the Cadence IP interrupts after the bus reset sequence. The problem with this sequence is that it might be pre-empted, giving SoundWire devices time to sync and report as ATTACHED before the interrupts are enabled. In that case, the Cadence IP will not detect a state change and will not throw an interrupt to proceed with the enumeration of a Device0. In our overnight stress tests, we observed that a slight sub-millisecond delay in enabling interrupts after the reset was correlated with detection failures. This problem is more prevalent on the LunarLake silicon, likely due to SOC integration changes, but it was observed on earlier generations as well. This patch reverts the sequence, with the interrupts enabled before performing the bus reset. This removes the race condition and makes sure the Cadence IP is able to detect the presence of a Device0 in all cases. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20240805115003.88035-1-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/intel_bus_common.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/drivers/soundwire/intel_bus_common.c b/drivers/soundwire/intel_bus_common.c index e5ac3cc7cb79b..179aa0d85951b 100644 --- a/drivers/soundwire/intel_bus_common.c +++ b/drivers/soundwire/intel_bus_common.c @@ -45,15 +45,15 @@ int intel_start_bus(struct sdw_intel *sdw) return ret; } - ret = sdw_cdns_exit_reset(cdns); + ret = sdw_cdns_enable_interrupt(cdns, true); if (ret < 0) { - dev_err(dev, "%s: unable to exit bus reset sequence: %d\n", __func__, ret); + dev_err(dev, "%s: cannot enable interrupts: %d\n", __func__, ret); return ret; } - ret = sdw_cdns_enable_interrupt(cdns, true); + ret = sdw_cdns_exit_reset(cdns); if (ret < 0) { - dev_err(dev, "%s: cannot enable interrupts: %d\n", __func__, ret); + dev_err(dev, "%s: unable to exit bus reset sequence: %d\n", __func__, ret); return ret; } @@ -136,15 +136,15 @@ int intel_start_bus_after_reset(struct sdw_intel *sdw) return ret; } - ret = sdw_cdns_exit_reset(cdns); + ret = sdw_cdns_enable_interrupt(cdns, true); if (ret < 0) { - dev_err(dev, "unable to exit bus reset sequence during resume\n"); + dev_err(dev, "cannot enable interrupts during resume\n"); return ret; } - ret = sdw_cdns_enable_interrupt(cdns, true); + ret = sdw_cdns_exit_reset(cdns); if (ret < 0) { - dev_err(dev, "cannot enable interrupts during resume\n"); + dev_err(dev, "unable to exit bus reset sequence during resume\n"); return ret; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 42/76] soundwire: cadence: re-check Peripheral status with delayed_work
by Sasha Levin 04 Oct '24

04 Oct '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit f8c35d61ba01afa76846905c67862cdace7f66b0 ] The SoundWire peripheral enumeration is entirely based on interrupts, more specifically sticky bits tracking state changes. This patch adds a defensive programming check on the actual status reported in PING frames. If for some reason an interrupt was lost or delayed, the delayed work would detect a peripheral change of status after the bus starts. The 100ms defined for the delay is not completely arbitrary, if a Peripheral didn't join the bus within that delay then probably the hardware link is broken, and conversely if the detection didn't happen because of software issues the 100ms is still acceptable in terms of user experience. The overhead of the one-shot workqueue is minimal, and the mutual exclusion ensures that the interrupt and delayed work cannot update the status concurrently. Reviewed-by: Liam Girdwood <liam.r.girdwood(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20240805114921.88007-1-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/cadence_master.c | 39 ++++++++++++++++++++++++++-- drivers/soundwire/cadence_master.h | 5 ++++ drivers/soundwire/intel.h | 2 ++ drivers/soundwire/intel_auxdevice.c | 1 + drivers/soundwire/intel_bus_common.c | 11 ++++++++ 5 files changed, 56 insertions(+), 2 deletions(-) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index e0683a5975d10..05652e983539b 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -890,8 +890,14 @@ static int cdns_update_slave_status(struct sdw_cdns *cdns, } } - if (is_slave) - return sdw_handle_slave_status(&cdns->bus, status); + if (is_slave) { + int ret; + + mutex_lock(&cdns->status_update_lock); + ret = sdw_handle_slave_status(&cdns->bus, status); + mutex_unlock(&cdns->status_update_lock); + return ret; + } return 0; } @@ -988,6 +994,31 @@ irqreturn_t sdw_cdns_irq(int irq, void *dev_id) } EXPORT_SYMBOL(sdw_cdns_irq); +static void cdns_check_attached_status_dwork(struct work_struct *work) +{ + struct sdw_cdns *cdns = + container_of(work, struct sdw_cdns, attach_dwork.work); + enum sdw_slave_status status[SDW_MAX_DEVICES + 1]; + u32 val; + int ret; + int i; + + val = cdns_readl(cdns, CDNS_MCP_SLAVE_STAT); + + for (i = 0; i <= SDW_MAX_DEVICES; i++) { + status[i] = val & 0x3; + if (status[i]) + dev_dbg(cdns->dev, "Peripheral %d status: %d\n", i, status[i]); + val >>= 2; + } + + mutex_lock(&cdns->status_update_lock); + ret = sdw_handle_slave_status(&cdns->bus, status); + mutex_unlock(&cdns->status_update_lock); + if (ret < 0) + dev_err(cdns->dev, "%s: sdw_handle_slave_status failed: %d\n", __func__, ret); +} + /** * cdns_update_slave_status_work - update slave status in a work since we will need to handle * other interrupts eg. CDNS_MCP_INT_RX_WL during the update slave @@ -1740,7 +1771,11 @@ int sdw_cdns_probe(struct sdw_cdns *cdns) init_completion(&cdns->tx_complete); cdns->bus.port_ops = &cdns_port_ops; + mutex_init(&cdns->status_update_lock); + INIT_WORK(&cdns->work, cdns_update_slave_status_work); + INIT_DELAYED_WORK(&cdns->attach_dwork, cdns_check_attached_status_dwork); + return 0; } EXPORT_SYMBOL(sdw_cdns_probe); diff --git a/drivers/soundwire/cadence_master.h b/drivers/soundwire/cadence_master.h index bc84435e420f5..e1d7969ba48ae 100644 --- a/drivers/soundwire/cadence_master.h +++ b/drivers/soundwire/cadence_master.h @@ -117,6 +117,8 @@ struct sdw_cdns_dai_runtime { * @link_up: Link status * @msg_count: Messages sent on bus * @dai_runtime_array: runtime context for each allocated DAI. + * @status_update_lock: protect concurrency between interrupt-based and delayed work + * status update */ struct sdw_cdns { struct device *dev; @@ -148,10 +150,13 @@ struct sdw_cdns { bool interrupt_enabled; struct work_struct work; + struct delayed_work attach_dwork; struct list_head list; struct sdw_cdns_dai_runtime **dai_runtime_array; + + struct mutex status_update_lock; /* add mutual exclusion to sdw_handle_slave_status() */ }; #define bus_to_cdns(_bus) container_of(_bus, struct sdw_cdns, bus) diff --git a/drivers/soundwire/intel.h b/drivers/soundwire/intel.h index 68838e843b543..f4235f5991c37 100644 --- a/drivers/soundwire/intel.h +++ b/drivers/soundwire/intel.h @@ -103,6 +103,8 @@ static inline void intel_writew(void __iomem *base, int offset, u16 value) #define INTEL_MASTER_RESET_ITERATIONS 10 +#define SDW_INTEL_DELAYED_ENUMERATION_MS 100 + #define SDW_INTEL_CHECK_OPS(sdw, cb) ((sdw) && (sdw)->link_res && (sdw)->link_res->hw_ops && \ (sdw)->link_res->hw_ops->cb) #define SDW_INTEL_OPS(sdw, cb) ((sdw)->link_res->hw_ops->cb) diff --git a/drivers/soundwire/intel_auxdevice.c b/drivers/soundwire/intel_auxdevice.c index 8807e01cbf7c7..dff49c5ce5b30 100644 --- a/drivers/soundwire/intel_auxdevice.c +++ b/drivers/soundwire/intel_auxdevice.c @@ -475,6 +475,7 @@ static void intel_link_remove(struct auxiliary_device *auxdev) */ if (!bus->prop.hw_disabled) { sdw_intel_debugfs_exit(sdw); + cancel_delayed_work_sync(&cdns->attach_dwork); sdw_cdns_enable_interrupt(cdns, false); } sdw_bus_master_delete(bus); diff --git a/drivers/soundwire/intel_bus_common.c b/drivers/soundwire/intel_bus_common.c index a75e5d7d87154..d3ff6c65b64c3 100644 --- a/drivers/soundwire/intel_bus_common.c +++ b/drivers/soundwire/intel_bus_common.c @@ -60,6 +60,9 @@ int intel_start_bus(struct sdw_intel *sdw) sdw_cdns_check_self_clearing_bits(cdns, __func__, true, INTEL_MASTER_RESET_ITERATIONS); + schedule_delayed_work(&cdns->attach_dwork, + msecs_to_jiffies(SDW_INTEL_DELAYED_ENUMERATION_MS)); + return 0; } @@ -151,6 +154,9 @@ int intel_start_bus_after_reset(struct sdw_intel *sdw) } sdw_cdns_check_self_clearing_bits(cdns, __func__, true, INTEL_MASTER_RESET_ITERATIONS); + schedule_delayed_work(&cdns->attach_dwork, + msecs_to_jiffies(SDW_INTEL_DELAYED_ENUMERATION_MS)); + return 0; } @@ -184,6 +190,9 @@ int intel_start_bus_after_clock_stop(struct sdw_intel *sdw) sdw_cdns_check_self_clearing_bits(cdns, __func__, true, INTEL_MASTER_RESET_ITERATIONS); + schedule_delayed_work(&cdns->attach_dwork, + msecs_to_jiffies(SDW_INTEL_DELAYED_ENUMERATION_MS)); + return 0; } @@ -194,6 +203,8 @@ int intel_stop_bus(struct sdw_intel *sdw, bool clock_stop) bool wake_enable = false; int ret; + cancel_delayed_work_sync(&cdns->attach_dwork); + if (clock_stop) { ret = sdw_cdns_clock_stop(cdns, true); if (ret < 0) -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 23/76] soundwire: intel_bus_common: enable interrupts before exiting reset
by Sasha Levin 04 Oct '24

04 Oct '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 5aedb8d8336b0a0421b58ca27d1b572aa6695b5b ] The existing code enables the Cadence IP interrupts after the bus reset sequence. The problem with this sequence is that it might be pre-empted, giving SoundWire devices time to sync and report as ATTACHED before the interrupts are enabled. In that case, the Cadence IP will not detect a state change and will not throw an interrupt to proceed with the enumeration of a Device0. In our overnight stress tests, we observed that a slight sub-millisecond delay in enabling interrupts after the reset was correlated with detection failures. This problem is more prevalent on the LunarLake silicon, likely due to SOC integration changes, but it was observed on earlier generations as well. This patch reverts the sequence, with the interrupts enabled before performing the bus reset. This removes the race condition and makes sure the Cadence IP is able to detect the presence of a Device0 in all cases. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20240805115003.88035-1-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/intel_bus_common.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/drivers/soundwire/intel_bus_common.c b/drivers/soundwire/intel_bus_common.c index df944e11b9caa..a75e5d7d87154 100644 --- a/drivers/soundwire/intel_bus_common.c +++ b/drivers/soundwire/intel_bus_common.c @@ -45,15 +45,15 @@ int intel_start_bus(struct sdw_intel *sdw) return ret; } - ret = sdw_cdns_exit_reset(cdns); + ret = sdw_cdns_enable_interrupt(cdns, true); if (ret < 0) { - dev_err(dev, "%s: unable to exit bus reset sequence: %d\n", __func__, ret); + dev_err(dev, "%s: cannot enable interrupts: %d\n", __func__, ret); return ret; } - ret = sdw_cdns_enable_interrupt(cdns, true); + ret = sdw_cdns_exit_reset(cdns); if (ret < 0) { - dev_err(dev, "%s: cannot enable interrupts: %d\n", __func__, ret); + dev_err(dev, "%s: unable to exit bus reset sequence: %d\n", __func__, ret); return ret; } @@ -136,15 +136,15 @@ int intel_start_bus_after_reset(struct sdw_intel *sdw) return ret; } - ret = sdw_cdns_exit_reset(cdns); + ret = sdw_cdns_enable_interrupt(cdns, true); if (ret < 0) { - dev_err(dev, "unable to exit bus reset sequence during resume\n"); + dev_err(dev, "cannot enable interrupts during resume\n"); return ret; } - ret = sdw_cdns_enable_interrupt(cdns, true); + ret = sdw_cdns_exit_reset(cdns); if (ret < 0) { - dev_err(dev, "cannot enable interrupts during resume\n"); + dev_err(dev, "unable to exit bus reset sequence during resume\n"); return ret; } -- 2.43.0

1 0

[PATCH v2] ALSA: hda/realtek: Fix the push button function for the ALC257
by Oder Chiou 04 Oct '24

04 Oct '24

The headset push button cannot work properly in case of the ALC257. This patch reverted the previous commit to correct the side effect. Fixes: ef9718b3d54e ("ALSA: hda/realtek: Fix noise from speakers on Lenovo IdeaPad 3 15IAU7") Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f787ff4182d4..86db1a4ec24c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -587,6 +587,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x10ec0257: case 0x19e58326: case 0x10ec0283: case 0x10ec0285: -- 2.34.1

3 2

[PATCH] ASoC: codecs: wcd9335: remove unnecessary MODULE_ALIAS()
by Masahiro Yamada 03 Oct '24

03 Oct '24

Since commit b4b818305578 ("slimbus: generate MODULE_ALIAS() from MODULE_DEVICE_TABLE()"), modpost automatically generates MODULE_ALIAS() from MODULE_DEVICE_TABLE(slim, ). Signed-off-by: Masahiro Yamada <masahiroy(a)kernel.org> --- sound/soc/codecs/wcd9335.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 373a31ddccb2..a2521e16c099 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -5177,4 +5177,3 @@ static struct slim_driver wcd9335_slim_driver = { module_slim_driver(wcd9335_slim_driver); MODULE_DESCRIPTION("WCD9335 slim driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("slim:217:1a0:*"); -- 2.43.0

2 1

[PATCH] ASoC: dt-bindings: davinci-mcasp: Fix interrupts property
by Miquel Raynal 03 Oct '24

03 Oct '24

My understanding of the interrupts property is that it can either be: 1/ - TX 2/ - TX - RX 3/ - Common/combined. There are very little chances that either: - TX - Common/combined or even - TX - RX - Common/combined could be a thing. Looking at the interrupt-names definition (which uses oneOf instead of anyOf), it makes indeed little sense to use anyOf in the interrupts definition. I believe this is just a mistake, hence let's fix it. Fixes: 8be90641a0bb ("ASoC: dt-bindings: davinci-mcasp: convert McASP bindings to yaml schema") Signed-off-by: Miquel Raynal <miquel.raynal(a)bootlin.com> --- --- .../devicetree/bindings/sound/davinci-mcasp-audio.yaml | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml index 7735e08d35ba..ab3206ffa4af 100644 --- a/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml +++ b/Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml @@ -102,7 +102,7 @@ properties: default: 2 interrupts: - anyOf: + oneOf: - minItems: 1 items: - description: TX interrupt -- 2.43.0

4 7

[PATCH v1] soundwire: Correct some typos in comments
by Shen Lichuan 03 Oct '24

03 Oct '24

Fixed some confusing spelling errors that were currently identified, the details are as follows: -in the code comments: drivers/soundwire/bus.c:115: enumuration==> enumeration drivers/soundwire/qcom.c:1176: wil ==> will drivers/soundwire/sysfs_slave.c:218: highligh ==> highlight Signed-off-by: Shen Lichuan <shenlichuan(a)vivo.com> --- drivers/soundwire/bus.c | 2 +- drivers/soundwire/qcom.c | 2 +- drivers/soundwire/sysfs_slave.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 263ca32f0c5c..d1dc62c34f1c 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -112,7 +112,7 @@ int sdw_bus_master_add(struct sdw_bus *bus, struct device *parent, /* Set higher order bits */ *bus->assigned = ~GENMASK(SDW_BROADCAST_DEV_NUM, SDW_ENUM_DEV_NUM); - /* Set enumuration device number and broadcast device number */ + /* Set enumeration device number and broadcast device number */ set_bit(SDW_ENUM_DEV_NUM, bus->assigned); set_bit(SDW_BROADCAST_DEV_NUM, bus->assigned); diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index aed57002fd0e..2b403b14066c 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1173,7 +1173,7 @@ static int qcom_swrm_stream_alloc_ports(struct qcom_swrm_ctrl *ctrl, else sconfig.direction = SDW_DATA_DIR_RX; - /* hw parameters wil be ignored as we only support PDM */ + /* hw parameters will be ignored as we only support PDM */ sconfig.ch_count = 1; sconfig.frame_rate = params_rate(params); sconfig.type = stream->type; diff --git a/drivers/soundwire/sysfs_slave.c b/drivers/soundwire/sysfs_slave.c index f4259710dd0f..c5c22d1708ec 100644 --- a/drivers/soundwire/sysfs_slave.c +++ b/drivers/soundwire/sysfs_slave.c @@ -215,7 +215,7 @@ const struct attribute_group *sdw_attr_groups[] = { /* * the status is shown in capital letters for UNATTACHED and RESERVED - * on purpose, to highligh users to the fact that these status values + * on purpose, to highlight users to the fact that these status values * are not expected. */ static const char *const slave_status[] = { -- 2.17.1

2 1

[PATCH 0/4] refactor amd soundwire manager driver for acp 6.3 platform
by Vijendar Mukunda 03 Oct '24

03 Oct '24

This patch series refactors the amd soundwire manager driver to support acp 6.3 platform based on acp pci revision id. This patch series prepared on top of asoc tree for-next branch. This patch series should be applied at one go. @Vinod: Please help to review soundwire patches and provide ack-by for this patch series. Vijendar Mukunda (4): soundwire: amd: pass acp pci revision id as resource data soundwire: amd: refactor existing code for acp 6.3 platform ASoC: SOF: amd: pass acp_rev as soundwire resource data ASoC: amd: ps: pass acp pci revision id as soundwire resource data drivers/soundwire/amd_init.c | 1 + drivers/soundwire/amd_manager.c | 99 ++++++++++++++++++++----------- drivers/soundwire/amd_manager.h | 16 ++--- include/linux/soundwire/sdw_amd.h | 6 ++ sound/soc/amd/ps/acp63.h | 2 + sound/soc/amd/ps/pci-ps.c | 2 + sound/soc/sof/amd/acp.c | 1 + 7 files changed, 84 insertions(+), 43 deletions(-) -- 2.34.1

3 7

[GIT PULL] ASoC fixes for v6.12-rc1
by Mark Brown 03 Oct '24

03 Oct '24

The following changes since commit 9852d85ec9d492ebef56dc5f229416c925758edc: Linux 6.12-rc1 (2024-09-29 15:06:19 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc1 for you to fetch changes up to eb1df4bbf53c29babf68fcb36e80d9c62a345257: qrb4210-rb2: add HDMI audio playback support (2024-10-02 18:38:11 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.12 A bunch of fixes here that came in during the merge window and the first week of release, plus some new quirks and device IDs. There's nothing major here, it's a bit bigger than it might've been due to there being no fixes sent during the merge window due to your vacation. ---------------------------------------------------------------- Alexey Klimov (3): ASoC: codecs: lpass-rx-macro: add missing CDC_RX_BCL_VBAT_RF_PROC2 to default regs values ASoC: dt-bindings: qcom,sm8250: add qrb4210-rb2-sndcard ASoC: qcom: sm8250: add qrb4210-rb2-sndcard compatible string Andrei Simion (1): ASoC: atmel: mchp-pdmc: Skip ALSA restoration if substream runtime is uninitialized Bard Liao (1): ASoC: Intel: soc-acpi-intel-rpl-match: add missing empty item Charles Han (1): ASoC: intel: sof_sdw: Add check devm_kasprintf() returned value Charles Keepax (1): ASoC: Intel: soc-acpi: arl: Fix some missing empty terminators David Lawrence Glanzman (1): ASoC: amd: yc: Add quirk for HP Dragonfly pro one Hui Wang (1): ASoC: imx-card: Set card.owner to avoid a warning calltrace if SND=m Mark Brown (3): Merge existing fixes from asoc/for-6.12 into new branch ASoC: Intel: soc-acpi: Fix missing empty terminators qrb4210-rb2: add HDMI audio playback support Miquel Raynal (1): ASoC: dt-bindings: davinci-mcasp: Fix interrupts property Ricardo Rivera-Matos (1): ASoC: cs35l45: Corrects cs35l45_get_clk_freq_id function data type Shengjiu Wang (1): ASoC: fsl_sai: Enable 'FIFO continue on error' FCONT bit Tang Bin (1): ASoC: topology: Fix incorrect addressing assignments Vijendar Mukunda (1): ASoC: amd: acp: don't set card long_name Wolfram Sang (1): ASoC: dt-bindings: renesas,rsnd: correct reg-names for R-Car Gen1 .../devicetree/bindings/sound/davinci-mcasp-audio.yaml | 2 +- Documentation/devicetree/bindings/sound/qcom,sm8250.yaml | 1 + Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 2 +- sound/soc/amd/acp/acp-sdw-sof-mach.c | 5 ----- sound/soc/amd/yc/acp6x-mach.c | 7 +++++++ sound/soc/atmel/mchp-pdmc.c | 3 +++ sound/soc/codecs/cs35l45-tables.c | 2 +- sound/soc/codecs/cs35l45.h | 2 +- sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/fsl/fsl_sai.c | 5 ++++- sound/soc/fsl/fsl_sai.h | 1 + sound/soc/fsl/imx-card.c | 1 + sound/soc/intel/boards/sof_sdw.c | 12 ++++++++++++ sound/soc/intel/common/soc-acpi-intel-arl-match.c | 2 ++ sound/soc/intel/common/soc-acpi-intel-rpl-match.c | 1 + sound/soc/qcom/sm8250.c | 1 + sound/soc/soc-topology.c | 4 ++-- 17 files changed, 40 insertions(+), 13 deletions(-)

2 1

[PATCH] ASoC: intel: sof_sdw: Add check devm_kasprintf() returned value
by Charles Han 02 Oct '24

02 Oct '24

devm_kasprintf() can return a NULL pointer on failure but this returned value is not checked. Fixes: b359760d95ee ("ASoC: intel: sof_sdw: Add simple DAI link creation helper") Signed-off-by: Charles Han <hanchunchao(a)inspur.com> --- sound/soc/intel/boards/sof_sdw.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index d258728d64cf..26917f6f15cf 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -919,6 +919,9 @@ static int create_ssp_dailinks(struct snd_soc_card *card, char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", i); char *codec_name = devm_kasprintf(dev, GFP_KERNEL, "i2c-%s:0%d", ssp_info->acpi_id, j++); + if (!name || !cpu_dai_name || !codec_name) + return -ENOMEM; + int playback = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_PLAYBACK]; int capture = ssp_info->dais[0].direction[SNDRV_PCM_STREAM_CAPTURE]; @@ -985,6 +988,9 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, for (i = 0; i < hdmi_num; i++) { char *name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d", i + 1); char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "iDisp%d Pin", i + 1); + if (!name || !cpu_dai_name) + return -ENOMEM; + char *codec_name, *codec_dai_name; if (intel_ctx->hdmi.idisp_codec) { @@ -996,6 +1002,9 @@ static int create_hdmi_dailinks(struct snd_soc_card *card, codec_dai_name = "snd-soc-dummy-dai"; } + if (!codec_dai_name) + return -ENOMEM; + ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, 1, 0, // HDMI only supports playback cpu_dai_name, platform_component->name, @@ -1019,6 +1028,9 @@ static int create_bt_dailinks(struct snd_soc_card *card, SOF_BT_OFFLOAD_SSP_SHIFT; char *name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d-BT", port); char *cpu_dai_name = devm_kasprintf(dev, GFP_KERNEL, "SSP%d Pin", port); + if (!name || !cpu_dai_name) + return -ENOMEM; + int ret; ret = asoc_sdw_init_simple_dai_link(dev, *dai_links, be_id, name, -- 2.31.1

3 2

[PATCH] ASoC: imx-card: Set card.owner to avoid a warning calltrace if SND=m
by Hui Wang 02 Oct '24

02 Oct '24

In most Linux distribution kernels, the SND is set to m, in such a case, when booting the kernel on i.MX8MP EVK board, there is a warning calltrace like below: Call trace: snd_card_init+0x484/0x4cc [snd] snd_card_new+0x70/0xa8 [snd] snd_soc_bind_card+0x310/0xbd0 [snd_soc_core] snd_soc_register_card+0xf0/0x108 [snd_soc_core] devm_snd_soc_register_card+0x4c/0xa4 [snd_soc_core] That is because the card.owner is not set, a warning calltrace is raised in the snd_card_init() due to it. Fixes: aa736700f42f ("ASoC: imx-card: Add imx-card machine driver") Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/fsl/imx-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/imx-card.c b/sound/soc/fsl/imx-card.c index 98b37dd2b901..a7215bad6484 100644 --- a/sound/soc/fsl/imx-card.c +++ b/sound/soc/fsl/imx-card.c @@ -710,6 +710,7 @@ static int imx_card_probe(struct platform_device *pdev) data->plat_data = plat_data; data->card.dev = &pdev->dev; + data->card.owner = THIS_MODULE; dev_set_drvdata(&pdev->dev, &data->card); snd_soc_card_set_drvdata(&data->card, data); -- 2.34.1

2 1

[PATCH] ASoC: qcom: Fix NULL Dereference in asoc_qcom_lpass_cpu_platform_probe()
by Gax-c 02 Oct '24

02 Oct '24

A devm_kzalloc() in asoc_qcom_lpass_cpu_platform_probe() could possibly return NULL pointer. NULL Pointer Dereference may be triggerred without addtional check. Add a NULL check for the returned pointer. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Zichen Xie <zichenxie0106(a)gmail.com> Cc: stable(a)vger.kernel.org Reported-by: Zichen Xie <zichenxie0106(a)gmail.com> --- sound/soc/qcom/lpass-cpu.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 5a47f661e0c6..242bc16da36d 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1242,6 +1242,8 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) /* Allocation for i2sctl regmap fields */ drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl), GFP_KERNEL); + if (!drvdata->i2sctl) + return -ENOMEM; /* Initialize bitfields for dai I2SCTL register */ ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl, -- 2.25.1

3 2

[RFC PATCH v3 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 02 Oct '24

02 Oct '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20240731083843.59911-… 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. 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 patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

3 9

[RESEND PATCH v4] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 02 Oct '24

02 Oct '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. So far, the Software maintainer has not been confirmed. Once the maintainer was confimred, the guy will update his mail into audio converter section. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - Add Touch Screen support v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 35 ++++++++++++++++++++++++++--------- 1 file changed, 26 insertions(+), 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index c27f3190737f..689cc1ae70b5 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22897,25 +22897,43 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml -F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22925,9 +22943,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

2 1

[PATCH] ASoC: qcom: Fix NULL Dereference in asoc_qcom_lpass_cpu_platform_probe()
by Gax-c 01 Oct '24

01 Oct '24

A devm_kzalloc() in asoc_qcom_lpass_cpu_platform_probe() could possibly return NULL pointer. NULL Pointer Dereference may be triggerred without addtional check. Add a NULL check for the returned pointer. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Zichen Xie <zichenxie0106(a)gmail.com> Reported-by: Zichen Xie <zichenxie0106(a)gmail.com> Reported-by: Zijie Zhao <zzjas98(a)gmail.com> Reported-by: Chenyuan Yang <chenyuan0y(a)gmail.com> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 5a47f661e0c6..a8e56f47f237 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1243,6 +1243,9 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl), GFP_KERNEL); + if (!drvdata->i2sctl) + return -ENOMEM; + /* Initialize bitfields for dai I2SCTL register */ ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl, drvdata->lpaif_map); -- 2.25.1

2 1

[PATCH] Fix possible NULL Pointer Dereference in 'asoc_qcom_lpass_cpu_platform_probe'
by Gax-c 01 Oct '24

01 Oct '24

A 'devm_kzalloc' in 'asoc_qcom_lpass_cpu_platform_probe' could possibly return NULL pointer. NULL Pointer Dereference may be triggerred in 'asoc_qcom_lpass_cpu_platform_probe' without addtional check. Add a null check for the returned pointer. Fixes: b5022a36d28f ("ASoC: qcom: lpass: Use regmap_field for i2sctl and dmactl registers") Signed-off-by: Zichen Xie <zichenxie0106(a)gmail.com> Reported-by: Zichen Xie <zichenxie0106(a)gmail.com> Reported-by: Zijie Zhao <zzjas98(a)gmail.com> Reported-by: Chenyuan Yang <chenyuan0y(a)gmail.com> --- sound/soc/qcom/lpass-cpu.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 5a47f661e0c6..a8e56f47f237 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -1243,6 +1243,9 @@ int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev) drvdata->i2sctl = devm_kzalloc(&pdev->dev, sizeof(struct lpaif_i2sctl), GFP_KERNEL); + if (!drvdata->i2sctl) + return -ENOMEM; + /* Initialize bitfields for dai I2SCTL register */ ret = lpass_cpu_init_i2sctl_bitfields(dev, drvdata->i2sctl, drvdata->lpaif_map); -- 2.25.1

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[PATCH 0/3] ASoC: fsl_micfil: fix and improvement
by Shengjiu Wang 01 Oct '24

01 Oct '24

Fix the usage of regmap_write_bits(). Move mclk clock enablement to late stage. Enable the micfil error interrupt. Shengjiu Wang (3): ASoC: fsl_micfil: fix regmap_write_bits usage ASoC: fsl_micfil: Add mclk enable flag ASoC: fsl_micfil: Enable micfil error interrupt sound/soc/fsl/fsl_micfil.c | 49 ++++++++++++++++++++++++++++++-------- 1 file changed, 39 insertions(+), 10 deletions(-) -- 2.34.1

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[PATCH -next 0/5] sound/soc: fix some macro definitions and usages
by Hongbo Li 01 Oct '24

01 Oct '24

Many variable in macro are not used as we used macro_check script to detect and mamually check, let us address these issues. Hongbo Li (5): sound/soc: fix macro definition on TWL4030_OUTPUT_PGA sound/soc remove unused substream in macro soc_component_mark_pop sound/soc remove unused substream in macro soc_dai_mark_pop sound/soc remove unused substream in macro soc_link_mark_pop sound/soc: fix macro definition on STM_SAI_HAS_EXT_SYNC sound/soc/codecs/twl4030.c | 12 ++++++------ sound/soc/soc-component.c | 14 +++++++------- sound/soc/soc-dai.c | 10 +++++----- sound/soc/soc-link.c | 10 +++++----- sound/soc/stm/stm32_sai_sub.c | 2 +- 5 files changed, 24 insertions(+), 24 deletions(-) -- 2.34.1

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[PATCH v2 0/2] {ASoC,media}: constify snd_soc_component_driver when used as read-only
by Javier Carrasco 01 Oct '24

01 Oct '24

Declare `snd_soc_component_driver` as const to move their declarations to read-only sections for the drivers that do not modify the struct after its declaration. Apart from a single case under media/, the affected drivers are members of the ASoC subsystem. To: Tim Harvey <tharvey(a)gateworks.com> To: Mauro Carvalho Chehab <mchehab(a)kernel.org> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> To: Ray Jui <rjui(a)broadcom.com> To: Scott Branden <sbranden(a)broadcom.com> To: Broadcom internal kernel review list <bcm-kernel-feedback-list(a)broadcom.com> To: Shenghao Ding <shenghao-ding(a)ti.com> To: Kevin Lu <kevin-lu(a)ti.com> To: Baojun Xu <baojun.xu(a)ti.com> To: Olivier Moysan <olivier.moysan(a)foss.st.com> To: Arnaud Pouliquen <arnaud.pouliquen(a)foss.st.com> To: Maxime Coquelin <mcoquelin.stm32(a)gmail.com> To: Alexandre Torgue <alexandre.torgue(a)foss.st.com> To: Kunihiko Hayashi <hayashi.kunihiko(a)socionext.com> To: Masami Hiramatsu <mhiramat(a)kernel.org> Cc: linux-media(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-sound(a)vger.kernel.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: alsa-devel(a)alsa-project.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Signed-off-by: Javier Carrasco <javier.carrasco.cruz(a)gmail.com> Changes in v2: - drop cs43130 and sti-sas, as they modifiy the struct in the probe function. - Link to v1: https://lore.kernel.org/r/20240725-const_snd_soc_component_driver-v1-0-3d7e… --- Javier Carrasco (2): media: i2c: tda1997x: constify snd_soc_component_driver struct ASoC: constify snd_soc_component_driver struct drivers/media/i2c/tda1997x.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/bcm/cygnus-pcm.c | 2 +- sound/soc/codecs/cpcap.c | 2 +- sound/soc/codecs/pcm186x.c | 4 ++-- sound/soc/codecs/pcm5102a.c | 2 +- sound/soc/codecs/spdif_receiver.c | 2 +- sound/soc/codecs/spdif_transmitter.c | 2 +- sound/soc/codecs/tas6424.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/uniphier/evea.c | 2 +- 12 files changed, 13 insertions(+), 13 deletions(-) --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240725-const_snd_soc_component_driver-b9629a95b948 Best regards, -- Javier Carrasco <javier.carrasco.cruz(a)gmail.com>

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[PATCH V2 0/9] ASoC: amd: acp: refactor acp version differentiation logic
by Vijendar Mukunda 01 Oct '24

01 Oct '24

Currently different logics being used in the code for acp version differentiation. This patch series refactors the code to use acp pci revision id for handling acp version specific code. Changes since v1: - Add patch to update mach_params subsystem_rev field in machine select logic. Vijendar Mukunda (9): ASoC: amd: acp: simplify platform conditional checks code ASoC: amd: acp: use acp_rev for platform specific conditional checks ASoC: amd: acp: use acp pci revision id for platform differntiation ASoC: amd: acp: store acp pci rev id in platform driver private structure ASoC: amd: acp: pass acp pci revision id as platform data ASoC: amd: acp: update mach_params subsystem_rev field ASoC: amd: acp: remove unused variable from acp_card_drvdata structure ASoC: amd: acp: replace adata->platform conditional check ASoC: amd: acp: remove unused variable from acp platform driver sound/soc/amd/acp/acp-i2s.c | 38 +++++++++++++-------------- sound/soc/amd/acp/acp-legacy-common.c | 24 ++++++++--------- sound/soc/amd/acp/acp-legacy-mach.c | 8 +++--- sound/soc/amd/acp/acp-mach-common.c | 38 +++++++++++++++++++-------- sound/soc/amd/acp/acp-mach.h | 12 +++------ sound/soc/amd/acp/acp-pci.c | 7 +---- sound/soc/amd/acp/acp-pdm.c | 2 +- sound/soc/amd/acp/acp-platform.c | 15 ++++++----- sound/soc/amd/acp/acp-rembrandt.c | 4 +-- sound/soc/amd/acp/acp-renoir.c | 4 +-- sound/soc/amd/acp/acp-sof-mach.c | 6 ++--- sound/soc/amd/acp/acp63.c | 4 +-- sound/soc/amd/acp/acp70.c | 14 ++++------ sound/soc/amd/acp/acp_common.h | 19 ++++++++++++++ sound/soc/amd/acp/amd.h | 9 ++----- 15 files changed, 109 insertions(+), 95 deletions(-) create mode 100644 sound/soc/amd/acp/acp_common.h -- 2.34.1

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[PATCH v2 0/2] Updates for Atmel SSC DAI
by Andrei Simion 01 Oct '24

01 Oct '24

This patch set includes two updates for the Atmel SSC DAI driver: - Address the limitation with the S24_LE format. - Add stream names for DPCM and future use-cases. Codrin Ciubotariu (2): ASoC: atmel: atmel_ssc_dai: Add stream names ASoC: atmel: atmel_ssc_dai: Drop S24_LE support due to single channel limitation sound/soc/atmel/atmel_ssc_dai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) base-commit: 7083504315d64199a329de322fce989e1e10f4f7 -- 2.34.1

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[PATCH 0/2] Updates for mchp-spdif(tx/rx)
by Andrei Simion 01 Oct '24

01 Oct '24

This patch set includes two updates for the MCHP SPDIF RX and TX drivers. The patches remove the interface name from the stream_name, allowing the interface name and index to be set in the Device Tree (DT) using the sound-name-prefix string property. Codrin Ciubotariu (2): ASoC: atmel: mchp-spdiftx: Remove interface name from stream_name ASoC: atmel: mchp-spdifrx: Remove interface name from stream_name sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) base-commit: 5acd9952f95fb4b7da6d09a3be39195a80845eb6 -- 2.34.1

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[PATCH 0/8] ASoC: amd: acp: refactor acp version differentiation logic
by Vijendar Mukunda 01 Oct '24

01 Oct '24

Currently different logics being used in the code for acp version differentiation. This patch series refactors the code to use acp pci revision id for handling acp version specific code. Vijendar Mukunda (8): ASoC: amd: acp: simplify platform conditional checks code ASoC: amd: acp: use acp_rev for platform specific conditional checks ASoC: amd: acp: use acp pci revision id for platform differntiation ASoC: amd: acp: store acp pci rev id in platform driver private structure ASoC: amd: acp: pass acp pci revision id as platform data ASoC: amd: acp: remove unused variable from acp_card_drvdata structure ASoC: amd: acp: replace adata->platform conditional check ASoC: amd: acp: remove unused variable from acp platform driver sound/soc/amd/acp/acp-i2s.c | 38 +++++++++++++-------------- sound/soc/amd/acp/acp-legacy-common.c | 24 ++++++++--------- sound/soc/amd/acp/acp-legacy-mach.c | 8 +++--- sound/soc/amd/acp/acp-mach-common.c | 38 +++++++++++++++++++-------- sound/soc/amd/acp/acp-mach.h | 12 +++------ sound/soc/amd/acp/acp-pci.c | 7 +---- sound/soc/amd/acp/acp-pdm.c | 2 +- sound/soc/amd/acp/acp-platform.c | 14 +++++----- sound/soc/amd/acp/acp-rembrandt.c | 4 +-- sound/soc/amd/acp/acp-renoir.c | 4 +-- sound/soc/amd/acp/acp-sof-mach.c | 6 ++--- sound/soc/amd/acp/acp63.c | 4 +-- sound/soc/amd/acp/acp70.c | 14 ++++------ sound/soc/amd/acp/acp_common.h | 19 ++++++++++++++ sound/soc/amd/acp/amd.h | 9 ++----- 15 files changed, 108 insertions(+), 95 deletions(-) create mode 100644 sound/soc/amd/acp/acp_common.h -- 2.34.1

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[PATCH v28 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 01 Oct '24

01 Oct '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v28: - Updated comments and commit log in the stop endpoint sync patch. Clarified that the default stop endpoint completion routine won't fully run as expected since it has a completion associated w/ the command. - Added a null check for sb->xhci within xhci_sideband_create_interrupter(). This is to just ensure that caller has registered sideband before calling create interrupter. Changes in v27: - Added some comments and notes about the offload design. Enforcing the q6routing to only allow one USB mixer (PCM device) to be enabled at a time. - Modified SND_JACK_USB notifications for all USB audio offloadable devices plugged in - Rebased on latest XHCI secondary interrupter IMOD changes upstream. Modified the change in this series to allow for XHCI sideband to set the IMOD for sideband clients. - Updated documentation on how USB SND kcontrols are involved in the overall design. - Remove mutex locking from suspend/resume platform ops, as USB core ensures that the interface and device are in the RPM_ACTIVE state while disconnect is handled. Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (30): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: xhci-sideband: Set IMOD for xHCI sideband clients usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 456 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 31 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 47 +- drivers/usb/host/xhci.h | 16 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 70 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 139 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 32 +- sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 342 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 106 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1968 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5870 insertions(+), 55 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

3 41

[PATCH] ASoC: fsl_sai: Enable 'FIFO continue on error' FCONT bit
by Shengjiu Wang 01 Oct '24

01 Oct '24

FCONT=1 means On FIFO error, the SAI will continue from the same word that caused the FIFO error to set after the FIFO warning flag has been cleared. Set FCONT bit in control register to avoid the channel swap issue after SAI xrun. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 5 ++++- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index ab58a4461073..634168d2bb6e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -613,6 +613,9 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr4 |= FSL_SAI_CR4_FRSZ(slots); + /* Set to avoid channel swap */ + val_cr4 |= FSL_SAI_CR4_FCONT; + /* Set to output mode to avoid tri-stated data pins */ if (tx) val_cr4 |= FSL_SAI_CR4_CHMOD; @@ -699,7 +702,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, 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, + FSL_SAI_CR4_CHMOD_MASK | FSL_SAI_CR4_FCONT_MASK, val_cr4); regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx, ofs), FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index dadbd16ee394..9c4d19fe22c6 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -137,6 +137,7 @@ /* SAI Transmit and Receive Configuration 4 Register */ +#define FSL_SAI_CR4_FCONT_MASK BIT(28) #define FSL_SAI_CR4_FCONT BIT(28) #define FSL_SAI_CR4_FCOMB_SHIFT BIT(26) #define FSL_SAI_CR4_FCOMB_SOFT BIT(27) -- 2.34.1

2 1

[PATCH alsa-ucm-conf 1/2] sof-soundwire: Change mapped control names to make them unique
by Maciej Strozek 01 Oct '24

01 Oct '24

Both 'Speaker Digital' and 'cs42l43 PDM2' names were found as substrings in other alsa controls, confusing the MixerElems paths. Make them unique to avoid errors for hardware mutes. Also add 'Capture' string into microphone controls to make them work properly. Fixes: e87dde51d689 ("sof-soundwire: Add basic support for cs42l43") Fixes: e4277a204cd6 ("sof-soundwire: Add support for cs42l43/cs35l56 bridge configuration") Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- ucm2/codecs/cs42l43/init.conf | 4 ++-- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- ucm2/sof-soundwire/cs42l43-dmic.conf | 2 +- 3 files changed, 6 insertions(+), 6 deletions(-) diff --git a/ucm2/codecs/cs42l43/init.conf b/ucm2/codecs/cs42l43/init.conf index 10a6e1f..cf61833 100644 --- a/ucm2/codecs/cs42l43/init.conf +++ b/ucm2/codecs/cs42l43/init.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='cs42l43 PDM2 Switch'" { + "name='cs42l43 Microphone Capture Switch'" { "name='cs42l43 Decimator 3 Switch'".vindex.0 0 "name='cs42l43 Decimator 4 Switch'".vindex.1 0 } - "name='cs42l43 PDM2 Volume'" { + "name='cs42l43 Microphone Capture Volume'" { "name='cs42l43 Decimator 3 Volume'".vindex.0 0 "name='cs42l43 Decimator 4 Volume'".vindex.1 0 } diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 5435b11..7c36655 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -2,11 +2,11 @@ LibraryConfig.remap.Config { ctl.default.map { - "name='Speaker Digital Switch'" { + "name='Speaker Combined Switch'" { "name='AMPL Speaker Switch'".vindex.0 0 "name='AMPR Speaker Switch'".vindex.1 0 } - "name='Speaker Digital Volume'" { + "name='Speaker Combined Volume'" { "name='AMPL Speaker Volume'".vindex.0 0 "name='AMPR Speaker Volume'".vindex.1 0 } @@ -56,6 +56,6 @@ SectionDevice."Speaker" { PlaybackPriority 100 PlaybackPCM "hw:${CardId},0" PlaybackMixer "default:${CardId}" - PlaybackMixerElem "Speaker Digital" + PlaybackMixerElem "Speaker Combined" } } diff --git a/ucm2/sof-soundwire/cs42l43-dmic.conf b/ucm2/sof-soundwire/cs42l43-dmic.conf index d5f895d..b68a70d 100644 --- a/ucm2/sof-soundwire/cs42l43-dmic.conf +++ b/ucm2/sof-soundwire/cs42l43-dmic.conf @@ -21,6 +21,6 @@ SectionDevice."Mic" { CapturePriority 100 CapturePCM "hw:${CardId},4" CaptureMixer "default:${CardId}" - CaptureMixerElem "cs42l43 PDM2" + CaptureMixerElem "cs42l43 Microphone" } } -- 2.39.5

3 5

[PATCH] ASoC: rt-sdw-common: Common functions for Realtek soundwire driver
by Jack Yu 01 Oct '24

01 Oct '24

This is the first version of common functions for Realtek soundwire codec driver. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt-sdw-common.c | 241 +++++++++++++++++++++++++++++++ sound/soc/codecs/rt-sdw-common.h | 67 +++++++++ 4 files changed, 315 insertions(+) create mode 100644 sound/soc/codecs/rt-sdw-common.c create mode 100644 sound/soc/codecs/rt-sdw-common.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7092842480ef..b095eba9d634 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1545,6 +1545,11 @@ config SND_SOC_RL6231 default m if SND_SOC_RT1305=m default m if SND_SOC_RT1308=m +config SND_SOC_RT_SDW_COMMON + tristate + default y if SND_SOC_RT721_SDCA_SDW=y + default m if SND_SOC_RT721_SDCA_SDW=m + config SND_SOC_RL6347A tristate default y if SND_SOC_RT274=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 54cbc3feae32..4497b2e30cd3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -216,6 +216,7 @@ snd-soc-rk3308-y := rk3308_codec.o snd-soc-rk3328-y := rk3328_codec.o snd-soc-rk817-y := rk817_codec.o snd-soc-rl6231-y := rl6231.o +snd-soc-rt-sdw-common-y := rt-sdw-common.o snd-soc-rl6347a-y := rl6347a.o snd-soc-rt1011-y := rt1011.o snd-soc-rt1015-y := rt1015.o @@ -618,6 +619,7 @@ obj-$(CONFIG_SND_SOC_RK3308) += snd-soc-rk3308.o obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o obj-$(CONFIG_SND_SOC_RK817) += snd-soc-rk817.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o +obj-$(CONFIG_SND_SOC_RT_SDW_COMMON) += snd-soc-rt-sdw-common.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o obj-$(CONFIG_SND_SOC_RT1015) += snd-soc-rt1015.o diff --git a/sound/soc/codecs/rt-sdw-common.c b/sound/soc/codecs/rt-sdw-common.c new file mode 100644 index 000000000000..e758567e9c58 --- /dev/null +++ b/sound/soc/codecs/rt-sdw-common.c @@ -0,0 +1,241 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt-sdw-common.c +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// + +/* + * This file defines common functions used with Realtek soundwire codecs. + */ + +#include <linux/module.h> +#include <linux/regmap.h> +#include <linux/bitops.h> +#include <linux/soundwire/sdw_registers.h> +#include <sound/jack.h> + +#include "rt-sdw-common.h" + +/** + * rt_sdca_index_write - Write a value to Realtek defined register. + * + * @map: map for setting. + * @nid: Realtek-defined ID. + * @reg: register. + * @value: value. + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ +int rt_sdca_index_write(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int value) +{ + unsigned int addr = (nid << 20) | reg; + int ret; + + ret = regmap_write(map, addr, value); + if (ret < 0) + pr_err("Failed to set value: %06x <= %04x ret=%d\n", + addr, value, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_index_write); + +/** + * rt_sdca_index_read - Read value from Realtek defined register. + * + * @map: map for setting. + * @nid: Realtek-defined ID. + * @reg: register. + * @value: value. + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ +int rt_sdca_index_read(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int *value) +{ + unsigned int addr = (nid << 20) | reg; + int ret; + + ret = regmap_read(map, addr, value); + if (ret < 0) + pr_err("Failed to get value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_index_read); + +/** + * rt_sdca_index_update_bits - Update value on Realtek defined register. + * + * @map: map for setting. + * @nid: Realtek-defined ID. + * @reg: register. + * @mask: Bitmask to change + * @value: New value for bitmask + * + * A value of zero will be returned on success, a negative errno will + * be returned in error cases. + */ + +int rt_sdca_index_update_bits(struct regmap *map, + unsigned int nid, unsigned int reg, unsigned int mask, unsigned int val) +{ + unsigned int tmp; + int ret; + + ret = rt_sdca_index_read(map, nid, reg, &tmp); + if (ret < 0) + return ret; + + set_mask_bits(&tmp, mask, val); + return rt_sdca_index_write(map, nid, reg, tmp); +} +EXPORT_SYMBOL_GPL(rt_sdca_index_update_bits); + +/** + * rt_sdca_btn_type - Decision of button type. + * + * @buffer: UMP message buffer. + * + * A button type will be returned regarding to buffer, + * it returns zero if buffer cannot be recognized. + */ +int rt_sdca_btn_type(unsigned char *buffer) +{ + u8 btn_type = 0; + int ret; + + btn_type |= buffer[0] & 0xf; + btn_type |= (buffer[0] >> 4) & 0xf; + btn_type |= buffer[1] & 0xf; + btn_type |= (buffer[1] >> 4) & 0xf; + + if (btn_type & BIT(0)) + ret |= SND_JACK_BTN_2; + if (btn_type & BIT(1)) + ret |= SND_JACK_BTN_3; + if (btn_type & BIT(2)) + ret |= SND_JACK_BTN_0; + if (btn_type & BIT(3)) + ret |= SND_JACK_BTN_1; + + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_btn_type); + +/** + * rt_sdca_headset_detect - Headset jack type detection. + * + * @map: map for setting. + * @entity_id: SDCA entity ID. + * + * A headset jack type will be returned, a negative errno will + * be returned in error cases. + */ +int rt_sdca_headset_detect(struct regmap *map, unsigned int entity_id) +{ + unsigned int det_mode, jack_type; + int ret; + + /* get detected_mode */ + ret = regmap_read(map, SDW_SDCA_CTL(SDCA_NUM_JACK_CODEC, entity_id, + RT_SDCA_CTL_DETECTED_MODE, 0), &det_mode); + + if (ret < 0) + goto io_error; + + switch (det_mode) { + case 0x00: + jack_type = 0; + break; + case 0x03: + jack_type = SND_JACK_HEADPHONE; + break; + case 0x05: + jack_type = SND_JACK_HEADSET; + break; + } + + /* write selected_mode */ + if (det_mode) { + ret = regmap_write(map, SDW_SDCA_CTL(SDCA_NUM_JACK_CODEC, entity_id, + RT_SDCA_CTL_SELECTED_MODE, 0), det_mode); + if (ret < 0) + goto io_error; + } + + return jack_type; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +} +EXPORT_SYMBOL_GPL(rt_sdca_headset_detect); + +/** + * rt_sdca_button_detect - Read UMP message and decide button type. + * + * @map: map for setting. + * @entity_id: SDCA entity ID. + * @hid_buf_addr: HID buffer address. + * @hid_id: Report ID for HID. + * @mask1: Mask for button message. + * @mask2: Mask for button message. + * + * A button type will be returned regarding to buffer, + * it returns zero if buffer cannot be recognized. + */ +int rt_sdca_button_detect(struct regmap *map, unsigned int entity_id, + unsigned int hid_buf_addr, unsigned int hid_id, + unsigned int mask1, unsigned int mask2) +{ + unsigned int btn_type = 0, offset, idx, val, owner; + unsigned char buf[3]; + int ret; + + /* get current UMP message owner */ + ret = regmap_read(map, SDW_SDCA_CTL(SDCA_NUM_HID, entity_id, + RT_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), &owner); + if (ret < 0) + return 0; + + /* if owner is device then there is no button event from device */ + if (owner == 1) + return 0; + + /* read UMP message offset */ + ret = regmap_read(map, SDW_SDCA_CTL(SDCA_NUM_HID, entity_id, + RT_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), &offset); + if (ret < 0) + goto _end_btn_det_; + + for (idx = 0; idx < sizeof(buf); idx++) { + ret = regmap_read(map, hid_buf_addr + offset + idx, &val); + if (ret < 0) + goto _end_btn_det_; + buf[idx] = val & 0xff; + } + /* Report ID for HID */ + if (buf[0] == hid_id) + btn_type = rt_sdca_btn_type(&buf[1]); + +_end_btn_det_: + /* Host is owner, so set back to device */ + if (owner == 0) + /* set owner to device */ + regmap_write(map, + SDW_SDCA_CTL(SDCA_NUM_HID, entity_id, + RT_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), 0x01); + + return btn_type; +} +EXPORT_SYMBOL_GPL(rt_sdca_button_detect); + +MODULE_DESCRIPTION("Realtek soundwire common functions"); +MODULE_AUTHOR("jack yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt-sdw-common.h b/sound/soc/codecs/rt-sdw-common.h new file mode 100644 index 000000000000..09e5ffd4fb9e --- /dev/null +++ b/sound/soc/codecs/rt-sdw-common.h @@ -0,0 +1,67 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +// +// rt-sdw-common.h +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// + +/* + * This file defines common functions used with Realtek soundwire codecs. + */ + +#ifndef __RT_SDW_COMMON_H__ +#define __RT_SDW_COMMON_H__ + +#define SDCA_NUM_JACK_CODEC 0x01 +#define SDCA_NUM_MIC_ARRAY 0x02 +#define SDCA_NUM_HID 0x03 +#define SDCA_NUM_AMP 0x04 +#define RT_SDCA_CTL_SELECTED_MODE 0x01 +#define RT_SDCA_CTL_DETECTED_MODE 0x02 +#define RT_SDCA_CTL_HIDTX_CURRENT_OWNER 0x10 +#define RT_SDCA_CTL_HIDTX_MESSAGE_OFFSET 0x12 + +struct rt_sdca_dmic_kctrl_priv { + unsigned int reg_base; + unsigned int count; + unsigned int max; + unsigned int invert; +}; + +#define RT_SDCA_PR_VALUE(xreg_base, xcount, xmax, xinvert) \ + ((unsigned long)&(struct rt_sdca_dmic_kctrl_priv) \ + {.reg_base = xreg_base, .count = xcount, .max = xmax, \ + .invert = xinvert}) + +#define RT_SDCA_FU_CTRL(xname, reg_base, xmax, xinvert, xcount, \ + xinfo, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = xinfo, \ + .get = xget, \ + .put = xput, \ + .private_value = RT_SDCA_PR_VALUE(reg_base, xcount, xmax, xinvert)} + +#define RT_SDCA_EXT_TLV(xname, reg_base, xhandler_get,\ + xhandler_put, xcount, xmax, tlv_array, xinfo) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = xinfo, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = RT_SDCA_PR_VALUE(reg_base, xcount, xmax, 0) } + + +int rt_sdca_index_write(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int value); +int rt_sdca_index_read(struct regmap *map, unsigned int nid, + unsigned int reg, unsigned int *value); +int rt_sdca_index_update_bits(struct regmap *map, + unsigned int nid, unsigned int reg, unsigned int mask, unsigned int val); +int rt_sdca_btn_type(unsigned char *buffer); +int rt_sdca_headset_detect(struct regmap *map, unsigned int entity_id); +int rt_sdca_button_detect(struct regmap *map, unsigned int entity_id, + unsigned int hid_buf_addr, unsigned int hid_id, + unsigned int mask1, unsigned int mask2); + +#endif /* __RT_SDW_COMMON_H__ */ -- 2.34.1

1 0

[PATCH v3 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 01 Oct '24

01 Oct '24

Hi Mark This is v3 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v2 -> v3 - tidyup Intel files v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r091lwhg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 583 deletions(-) -- 2.43.0

1 13

[PATCH v2 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 01 Oct '24

01 Oct '24

Hi Mark This is v2 patch-set Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. v1 -> v2 - rebased on v6.12-rc1 - tidyup flag handling for sof Link: https://lore.kernel.org/r/87r0979uhx.wl-kuninori.morimoto.gx@renesas.com Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 7 +- 68 files changed, 318 insertions(+), 583 deletions(-) -- 2.43.0

2 15

[PATCH] sound: soc: codecs: tlv320aic3x: Fix codec gpio-reset
by Benjamin Hahn 01 Oct '24

01 Oct '24

The TLV320AIC3007 requires a hardware reset after power-up for proper operation. After all power supplies are at their specified values, the RESET pin must be driven low for at least 10 ns. Even though the datasheet sais min 10ns, I needed more than 10ns when testing this out. 15ns worked for me. Signed-off-by: Benjamin Hahn <B.Hahn(a)phytec.de> --- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 56e795a00e22..d002fc0b99b5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1367,8 +1367,11 @@ static int aic3x_regulator_event(struct notifier_block *nb, * Put codec to reset and require cache sync as at least one * of the supplies was disabled */ - if (aic3x->gpio_reset) + if (aic3x->gpio_reset) { gpiod_set_value(aic3x->gpio_reset, 1); + ndelay(15); + gpiod_set_value(aic3x->gpio_reset, 0); + } regcache_mark_dirty(aic3x->regmap); } @@ -1813,6 +1816,10 @@ int aic3x_probe(struct device *dev, struct regmap *regmap, kernel_ulong_t driver gpiod_set_consumer_name(aic3x->gpio_reset, "tlv320aic3x reset"); + /* CODEC datasheet says minimum of 10ns */ + ndelay(15); + gpiod_set_value(aic3x->gpio_reset, 0); + for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) aic3x->supplies[i].supply = aic3x_supply_names[i]; --- base-commit: ad46e8f95e931e113cb98253daf6d443ac244cde change-id: 20240930-wip-bhahn-tlv320aic3x_fix_reset-cca918a4889d Best regards, -- Benjamin Hahn <B.Hahn(a)phytec.de>

2 1

[PATCH] ASoC: fsl: Use maple tree register cache
by Mark Brown 01 Oct '24

01 Oct '24

Several of the NXP drivers use regmaps with a rbtree register cache. Since the maple tree cache is uisng a generally more modern data structure which makes implementation choices more suitable for modern systems let's convert these drivers to it. This should have no practical impact. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index a6cbaa6364c7..910b4ce7671c 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -169,7 +169,7 @@ static const struct regmap_config fsl_aud2htx_regmap_config = { .readable_reg = fsl_aud2htx_readable_reg, .volatile_reg = fsl_aud2htx_volatile_reg, .writeable_reg = fsl_aud2htx_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; static const struct of_device_id fsl_aud2htx_dt_ids[] = { diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 962f30912091..5149e732f763 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -1748,7 +1748,7 @@ static const struct regmap_config fsl_easrc_regmap_config = { .rd_table = &fsl_easrc_readable_table, .wr_table = &fsl_easrc_writeable_table, .volatile_table = &fsl_easrc_volatileable_table, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; #ifdef DEBUG diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 22b240a70ad4..c0dc9bebb77b 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -955,7 +955,7 @@ static const struct regmap_config fsl_micfil_regmap_config = { .readable_reg = fsl_micfil_readable_reg, .volatile_reg = fsl_micfil_volatile_reg, .writeable_reg = fsl_micfil_writeable_reg, - .cache_type = REGCACHE_RBTREE, + .cache_type = REGCACHE_MAPLE, }; /* END OF REGMAP */ --- base-commit: 98f7e32f20d28ec452afb208f9cffc08448a2652 change-id: 20240924-asoc-imx-maple-abb62f013533 Best regards, -- Mark Brown <broonie(a)kernel.org>

3 3

[PATCH] ASoC: meson: axg-iface: set continuous rates
by Jerome Brunet 01 Oct '24

01 Oct '24

The axg TDM HW does not depend on a selected set of rates. The hardware itself, just takes an input clock and work with it, regardless of its rate. In this way, the rates TDM can take are continuous. What might force the use of specific rate are the PLL available as clock and/or the codecs facing the TDM HW. Either way, this constraint does not belong in the TDM interface driver. Allow any rate as far as TDM is concerned by setting SNDRV_PCM_RATE_CONTINUOUS with an interval it has been tested with. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/axg-tdm-interface.c | 12 +++++++++--- sound/soc/meson/axg-tdm.h | 2 -- 2 files changed, 9 insertions(+), 5 deletions(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 62057c71f742..09103eef2a97 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -442,14 +442,18 @@ static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { .stream_name = "Playback", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .id = TDM_IFACE_PAD, @@ -461,7 +465,9 @@ static const struct snd_soc_dai_driver axg_tdm_iface_dai_drv[] = { .stream_name = "Loopback", .channels_min = 1, .channels_max = AXG_TDM_CHANNEL_MAX, - .rates = AXG_TDM_RATES, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 768000, .formats = AXG_TDM_FORMATS, }, .id = TDM_IFACE_LOOPBACK, diff --git a/sound/soc/meson/axg-tdm.h b/sound/soc/meson/axg-tdm.h index 1a17f546ce6e..acfcd48f8a00 100644 --- a/sound/soc/meson/axg-tdm.h +++ b/sound/soc/meson/axg-tdm.h @@ -15,8 +15,6 @@ #define AXG_TDM_NUM_LANES 4 #define AXG_TDM_CHANNEL_MAX 128 -#define AXG_TDM_RATES (SNDRV_PCM_RATE_5512 | \ - SNDRV_PCM_RATE_8000_768000) #define AXG_TDM_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_LE | \ --- base-commit: cd87a98b53518e44cf3c1a7c1c07c869ce33bf83 change-id: 20240920-asoc-axg-iface-continuous-b94cfa234787 Best regards, -- Jerome

2 1

[PATCH] ASoC: tas2781: Fix redundant parameter assignment
by Tang Bin 01 Oct '24

01 Oct '24

In these functions, the variable 'rc' is redundant, thus remove it. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/tas2781-i2c.c | 35 ++++++++-------------------------- 1 file changed, 8 insertions(+), 27 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 8a8d97dd7..68887799e 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -650,7 +650,6 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; @@ -659,9 +658,7 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, @@ -673,7 +670,6 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG; @@ -681,9 +677,7 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, reg = TAS2563_RUNTIME_RE_REG; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, @@ -696,7 +690,6 @@ static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); @@ -707,9 +700,7 @@ static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, else return -1; dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, @@ -721,13 +712,10 @@ static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, @@ -739,13 +727,10 @@ static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; - int rc = -1; guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; - rc = calib_data_get(tas_priv, reg, &dst[1]); - - return rc; + return calib_data_get(tas_priv, reg, &dst[1]); } static int tasdev_nop_get( @@ -1115,14 +1100,12 @@ static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); int dev_id = ucontrol->value.integer.value[0]; - int max = tas_priv->ndev - 1, rc; + int max = tas_priv->ndev - 1; dev_id = clamp(dev_id, 0, max); guard(mutex)(&tas_priv->codec_lock); - rc = tasdev_chn_switch(tas_priv, dev_id); - - return rc; + return tasdev_chn_switch(tas_priv, dev_id); } static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) @@ -1339,10 +1322,8 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) i++; } - rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, + return snd_soc_add_component_controls(priv->codec, cali_ctrls, nctrls < i ? nctrls : i); - - return rc; } static void tasdevice_fw_ready(const struct firmware *fmw, -- 2.33.0

2 1

[PATCH v2] ALSA: hda/conexant: fix some typos
by Oldherl Oh 30 Sep '24

30 Sep '24

Fix some typos in patch_conexant.c Signed-off-by: Oldherl Oh <me(a)oldherl.one> --- sound/pci/hda/patch_conexant.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index e851785ff058..ade42a8209c2 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -166,18 +166,18 @@ static void cxt_init_gpio_led(struct hda_codec *codec) static void cx_fixup_headset_recog(struct hda_codec *codec) { - unsigned int mic_persent; + unsigned int mic_present; /* fix some headset type recognize fail issue, such as EDIFIER headset */ - /* set micbiasd output current comparator threshold from 66% to 55%. */ + /* set micbias output current comparator threshold from 66% to 55%. */ snd_hda_codec_write(codec, 0x1c, 0, 0x320, 0x010); - /* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias registor + /* set OFF voltage for DFET from -1.2V to -0.8V, set headset micbias register * value adjustment trim from 2.2K ohms to 2.0K ohms. */ snd_hda_codec_write(codec, 0x1c, 0, 0x3b0, 0xe10); /* fix reboot headset type recognize fail issue */ - mic_persent = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); - if (mic_persent & AC_PINSENSE_PRESENCE) + mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); + if (mic_present & AC_PINSENSE_PRESENCE) /* enable headset mic VREF */ snd_hda_codec_write(codec, 0x19, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24); else @@ -249,9 +249,9 @@ static void cx_update_headset_mic_vref(struct hda_codec *codec, struct hda_jack_ { unsigned int mic_present; - /* In cx8070 and sn6140, the node 16 can only be config to headphone or disabled, - * the node 19 can only be config to microphone or disabled. - * Check hp&mic tag to process headset pulgin&plugout. + /* In cx8070 and sn6140, the node 16 can only be configured to headphone or disabled, + * the node 19 can only be configured to microphone or disabled. + * Check hp&mic tag to process headset plugin & plugout. */ mic_present = snd_hda_codec_read(codec, 0x19, 0, AC_VERB_GET_PIN_SENSE, 0x0); if (!(mic_present & AC_PINSENSE_PRESENCE)) /* mic plugout */ -- 2.46.2

2 1

ucm2: USB-Audio: add Steinberg UR22C (USB0499:172f)
by GitHub pull_request - opened 30 Sep '24

30 Sep '24

alsa-project/alsa-ucm-conf pull request #446 was opened from CoelacanthusHex: Similar to UR24C and UR44C, but with 2 inputs 2 outputs. Signed-off-by: Celeste Liu <CoelacanthusHex(a)gmail.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/446 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/446.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: hda/realtek: Fix the push button function for the ALC257
by Oder Chiou 30 Sep '24

30 Sep '24

The patch "Fix noise from speakers on Lenovo IdeaPad 3 15IAU7" caused that the headset push button cannot work properly in case of ALC257. This patch reverted it to correct the side effect. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f787ff4182d4..86db1a4ec24c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -587,6 +587,7 @@ static void alc_shutup_pins(struct hda_codec *codec) switch (codec->core.vendor_id) { case 0x10ec0236: case 0x10ec0256: + case 0x10ec0257: case 0x19e58326: case 0x10ec0283: case 0x10ec0285: -- 2.34.1

2 1

Fix some typos in hda/conexant
by Oldherl Oh 30 Sep '24

30 Sep '24

I'm trying to understand Conexant driver code in order to solve my microphone detection problem. As I am new to kernel development, I have made this simple typo-fixing-only patch as an exercise. Hope it is not disturbing.

2 2

[PATCH v2] ALSA: Fix typos in comments across various files
by Yu Jiaoliang 30 Sep '24

30 Sep '24

This patch fixes typos in comments within the ALSA subsystem. These changes improve code readability without affecting functionality. Signed-off-by: Yu Jiaoliang <yujiaoliang(a)vivo.com> --- v2: Update commit message. --- sound/core/compress_offload.c | 2 +- sound/core/oss/rate.c | 2 +- sound/core/pcm_native.c | 2 +- sound/core/sound.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index b8c0d6edbdd1..bdf1d78de833 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -288,7 +288,7 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream = &data->stream; guard(mutex)(&stream->device->lock); - /* write is allowed when stream is running or has been steup */ + /* write is allowed when stream is running or has been setup */ switch (stream->runtime->state) { case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 98269119347f..b56eeda5e30e 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -294,7 +294,7 @@ static int rate_action(struct snd_pcm_plugin *plugin, default: break; } - return 0; /* silenty ignore other actions */ + return 0; /* silently ignore other actions */ } int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug, diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 99e39b5359cc..b49bc89c0758 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3115,7 +3115,7 @@ struct snd_pcm_sync_ptr32 { } c; } __packed; -/* recalcuate the boundary within 32bit */ +/* recalculate the boundary within 32bit */ static snd_pcm_uframes_t recalculate_boundary(struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t boundary; diff --git a/sound/core/sound.c b/sound/core/sound.c index b9db9aa0bfcb..6531a67f13b3 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -133,7 +133,7 @@ static struct snd_minor *autoload_device(unsigned int minor) /* /dev/aloadSEQ */ snd_request_other(minor); } - mutex_lock(&sound_mutex); /* reacuire lock */ + mutex_lock(&sound_mutex); /* reacquire lock */ return snd_minors[minor]; } #else /* !CONFIG_MODULES */ -- 2.34.1

2 1

[PATCH v1] slimbus: Correct some typos in comments
by Shen Lichuan 29 Sep '24

29 Sep '24

Fixed some confusing typos that were currently identified with codespell, the details are as follows: -in the code comments: drivers/slimbus/messaging.c:226: transfered ==> transferred drivers/slimbus/messaging.c:230: vlaue ==> value drivers/slimbus/qcom-ctrl.c:127: Oder ==> Order drivers/slimbus/slimbus.h:263: Presense ==> Presence drivers/slimbus/slimbus.h:371: resposible ==> responsible Signed-off-by: Shen Lichuan <shenlichuan(a)vivo.com> --- drivers/slimbus/messaging.c | 4 ++-- drivers/slimbus/qcom-ctrl.c | 2 +- drivers/slimbus/slimbus.h | 4 ++-- 3 files changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/slimbus/messaging.c b/drivers/slimbus/messaging.c index 242570a5e565..b7dd2badc9d1 100644 --- a/drivers/slimbus/messaging.c +++ b/drivers/slimbus/messaging.c @@ -223,11 +223,11 @@ static u16 slim_slicesize(int code) /** * slim_xfer_msg() - Transfer a value info message on slim device * - * @sbdev: slim device to which this msg has to be transfered + * @sbdev: slim device to which this msg has to be transferred * @msg: value info message pointer * @mc: message code of the message * - * Called by drivers which want to transfer a vlaue or info elements. + * Called by drivers which want to transfer a value or info elements. * * Return: -ETIMEDOUT: If transmission of this message timed out */ diff --git a/drivers/slimbus/qcom-ctrl.c b/drivers/slimbus/qcom-ctrl.c index e25f9bdd9b23..039384d85e7b 100644 --- a/drivers/slimbus/qcom-ctrl.c +++ b/drivers/slimbus/qcom-ctrl.c @@ -124,7 +124,7 @@ static void qcom_slim_queue_tx(struct qcom_slim_ctrl *ctrl, void *buf, __iowrite32_copy(ctrl->base + tx_reg, buf, count); - /* Ensure Oder of subsequent writes */ + /* Ensure Order of subsequent writes */ mb(); } diff --git a/drivers/slimbus/slimbus.h b/drivers/slimbus/slimbus.h index 00a7f112574b..2067d47541f2 100644 --- a/drivers/slimbus/slimbus.h +++ b/drivers/slimbus/slimbus.h @@ -260,7 +260,7 @@ enum slim_ch_aux_bit_fmt { * struct slim_channel - SLIMbus channel, used for state machine * * @id: ID of channel - * @prrate: Presense rate of channel from Table 66 of SLIMbus 2.0 Specs + * @prrate: Presence rate of channel from Table 66 of SLIMbus 2.0 Specs * @seg_dist: segment distribution code from Table 20 of SLIMbus 2.0 Specs * @data_fmt: Data format of channel. * @aux_fmt: Aux format for this channel. @@ -368,7 +368,7 @@ struct slim_stream_runtime { * @get_laddr: It is possible that controller needs to set fixed logical * address table and get_laddr can be used in that case so that controller * can do this assignment. Use case is when the master is on the remote - * processor side, who is resposible for allocating laddr. + * processor side, who is responsible for allocating laddr. * @wakeup: This function pointer implements controller-specific procedure * to wake it up from clock-pause. Framework will call this to bring * the controller out of clock pause. -- 2.17.1

1 0

ucm2: Add Roland Bridge Cast V2
by GitHub pull_request - edited 28 Sep '24

28 Sep '24

alsa-project/alsa-ucm-conf pull request #445 was edited from Fumler: Roland Bridge Cast recently had a firmware update which brings in e.g. virtual surround and is dubbed as "V2" and has a separate usb device id than the existing Bridge Cast in the repo. This PR adds the changes needed for the new firmware to be mapped correctly, as a new USB audio device. Changelog: https://www.roland.com/global/support/by_product/bridge_cast/updates_driver… Thanks to [geoffreybennett](https://old.reddit.com/user/geoffreybennett) on reddit for helping me understand how to debug this stuff. Thread: https://old.reddit.com/r/linuxaudio/comments/1fq06z8/roland_bridge_cast_no_… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/445 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/445.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add Roland Bridge Cast V2
by GitHub pull_request - opened 28 Sep '24

28 Sep '24

alsa-project/alsa-ucm-conf pull request #445 was opened from Fumler: Roland Bridge Cast recently had a firmware update which brings in e.g. virtual surround and is dubbed as "V2" and has a separate usb device id than the existing Bridge Cast in the repo. This PR adds the changes needed for the new firmware to be mapped correctly, as a new USB audio device. Changelog: https://www.roland.com/global/support/by_product/bridge_cast/updates_driver… Thanks to [geoffreybennett](https://old.reddit.com/user/geoffreybennett) on reddit for helping me understand how to debug this stuff. Thread: https://old.reddit.com/r/linuxaudio/comments/1fq06z8/roland_bridge_cast_no_… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/445 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/445.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No sound from XonarDX
by GitHub issues - reopened 28 Sep '24

28 Sep '24

alsa-project/alsa-lib issue #408 was reopened from 3bitbrain: No matter what wiki or advice I try, no sound at all from a known good Xonar DX card. Distro : Gentoo, fresh install. Always un-muted everything in the console alsamixer. alsa-lib version 1.2.11 [alsa.txt](https://github.com/user-attachments/files/16697602/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-lib/issues/408 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param()
by (학생) 장인규 (컴퓨터공학과) 26 Sep '24

26 Sep '24

From 791716bf359b8540c519810848fd1f8006d7c3c5 Mon Sep 17 00:00:00 2001 From: Ingyu Jang <ingyujang25(a)unist.ac.kr> Date: Thu, 26 Sep 2024 19:40:04 +0900 Subject: [PATCH] sound: soc: uniphier: Handle regmap_write errors in aio_src_set_param() The aio_src_set_param() function did not previously check the return values of regmap_write() and regmap_update_bits(). If these functions fail, it could lead to silent failures when configuring the sample rate converter (SRC), causing improper behavior in audio processing without any indication of an error. This patch modifies aio_src_set_param to check the return values of regmap_write() and regmap_update_bits(). If either function returns an error, the error code is propagated back to the caller to ensure proper error handling. This change aligns with the existing error-handling behavior in functions like uniphier_aio_prepare(), where a failure in a sub-function should result in an immediate return of the error. Signed-off-by: Ingyu Jang <ingyujang25(a)unist.ac.kr> --- sound/soc/uniphier/aio-core.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 0eba60758134..2c4e8b873253 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -921,16 +921,19 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, { struct regmap *r = sub->aio->chip->regmap; u32 v; + int ret; if (sub->swm->dir != PORT_DIR_OUTPUT) return 0; - regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), + ret = regmap_write(r, OPORTMXSRC1CTR(sub->swm->oport.map), OPORTMXSRC1CTR_THMODE_SRC | OPORTMXSRC1CTR_SRCPATH_CALC | OPORTMXSRC1CTR_SYNC_ASYNC | OPORTMXSRC1CTR_FSIIPSEL_INNER | OPORTMXSRC1CTR_FSISEL_ACLK); + if (ret) + return ret; switch (params_rate(params)) { default: @@ -951,12 +954,18 @@ int aio_src_set_param(struct uniphier_aio_sub *sub, break; } - regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), + + ret = regmap_write(r, OPORTMXRATE_I(sub->swm->oport.map), v | OPORTMXRATE_I_ACLKSRC_APLL | OPORTMXRATE_I_LRCKSTP_STOP); - regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), + if (ret) + return ret; + + ret = regmap_update_bits(r, OPORTMXRATE_I(sub->swm->oport.map), OPORTMXRATE_I_LRCKSTP_MASK, OPORTMXRATE_I_LRCKSTP_START); + if (ret) + return ret; return 0; } -- 2.34.1

1 0

GoXLR - Dynamic Channel Count
by GitHub issues - edited 26 Sep '24

26 Sep '24

alsa-project/alsa-ucm-conf issue #444 was edited from FrostyCoolSlug: The next GoXLR firmware release is going to add two new output channels for the full sized device, labeled a 'Stream Mix 2'. Outside of general usage, there's no way to determine whether these channels exist or not outside of the UCM2 definitions as the device is booting up. Is there currently the facility to configure the devices virtual channels based on the number of channels reported by the hardware (so if there are 23 hardware channels, use the current configuration, if there are 25 hardware channels use an adjusted configuration), and if not, could you suggest the best way to handle this implementation? Regards, Craig McLure Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/444 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

GoXLR - Dynamic Output Channel Count
by GitHub issues - opened 26 Sep '24

26 Sep '24

alsa-project/alsa-ucm-conf issue #444 was opened from FrostyCoolSlug: The next GoXLR firmware release is going to add two new output channels for the full sized device, labeled a 'Stream Mix 2'. Outside of general usage, there's no way to determine whether these channels exist or not outside of the UCM2 definitions as the device is booting up. Is there currently the facility to configure the devices virtual channels based on the number of channels reported by the hardware (so if there are 23 hardware channels, use the current configuration, if there are 25 hardware channels use an adjusted configuration), and if not, could you suggest the best way to handle this implementation? Regards, Craig McLure Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/444 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 00/13] ASoC: remove dpcm_xxx flags
by Kuninori Morimoto 26 Sep '24

26 Sep '24

Hi Mark Now, we don't need to use dpcm_playback/capture flags for DPCM since v6.11, and thus we can use playback/capture_only flags instead too for DPCM, which is same as non-DPCM. Let's remove dpcm_playback/capture flags, and use playback/capture_only flags instead. [Before] [After] dpcm_playback = 1; /* no setting is needed */ dpcm_capture = 1; dpcm_playback = 1; playback_only = 1; dpcm_capture = 0; dpcm_playback = 0; capture_only = 1; dpcm_capture = 1; And then, because no-one uses dpcm_xxx flags, we don't need to have the code for both DPCM and non-DPCM. These can be handled by same code. This patch-set is based on latest linus/master branch. Kuninori Morimoto (13): ASoC: amd: remove dpcm_xxx flags ASoC: fsl: remove dpcm_xxx flags ASoC: sof: remove dpcm_xxx flags ASoC: intel: remove dpcm_xxx flags ASoC: samsung: remove dpcm_xxx flags ASoC: mediatek: remove dpcm_xxx flags ASoC: soc-core: remove dpcm_xxx flags ASoC: soc-compress: remove dpcm_xxx flags ASoC: soc-topology: remove dpcm_xxx flags ASoC: intel: boards: remove dpcm_xxx flags ASoC: soc-pcm: remove dpcm_xxx flags ASoC: doc: remove dpcm_xxx flags ASoC: soc-pcm: merge DPCM and non-DPCM validation check Documentation/sound/soc/dpcm.rst | 11 +-- include/sound/soc.h | 5 - sound/soc/amd/acp-da7219-max98357a.c | 20 ++-- sound/soc/amd/acp-es8336.c | 2 - sound/soc/amd/acp/acp-mach-common.c | 24 ++--- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +- sound/soc/amd/vangogh/acp5x-mach.c | 6 -- sound/soc/fsl/fsl-asoc-card.c | 24 ++--- sound/soc/fsl/imx-audmix.c | 18 ++-- sound/soc/fsl/imx-card.c | 4 +- sound/soc/intel/avs/boards/da7219.c | 2 - sound/soc/intel/avs/boards/dmic.c | 4 +- sound/soc/intel/avs/boards/es8336.c | 2 - sound/soc/intel/avs/boards/hdaudio.c | 4 - sound/soc/intel/avs/boards/i2s_test.c | 2 - sound/soc/intel/avs/boards/max98357a.c | 2 +- sound/soc/intel/avs/boards/max98373.c | 2 - sound/soc/intel/avs/boards/max98927.c | 2 - sound/soc/intel/avs/boards/nau8825.c | 2 - sound/soc/intel/avs/boards/rt274.c | 2 - sound/soc/intel/avs/boards/rt286.c | 2 - sound/soc/intel/avs/boards/rt298.c | 2 - sound/soc/intel/avs/boards/rt5514.c | 2 +- sound/soc/intel/avs/boards/rt5663.c | 2 - sound/soc/intel/avs/boards/rt5682.c | 2 - sound/soc/intel/avs/boards/ssm4567.c | 2 - sound/soc/intel/boards/bdw-rt5650.c | 4 - sound/soc/intel/boards/bdw-rt5677.c | 4 - sound/soc/intel/boards/bdw_rt286.c | 10 +- sound/soc/intel/boards/bytcht_cx2072x.c | 6 +- sound/soc/intel/boards/bytcht_da7213.c | 6 +- sound/soc/intel/boards/bytcht_es8316.c | 6 +- sound/soc/intel/boards/bytcht_nocodec.c | 6 +- sound/soc/intel/boards/bytcr_rt5640.c | 6 +- sound/soc/intel/boards/bytcr_rt5651.c | 6 +- sound/soc/intel/boards/bytcr_wm5102.c | 6 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 6 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 6 +- sound/soc/intel/boards/ehl_rt5660.c | 14 ++- sound/soc/intel/boards/hsw_rt5640.c | 10 +- sound/soc/intel/boards/sof_board_helpers.c | 15 +-- sound/soc/intel/boards/sof_es8336.c | 8 +- sound/soc/intel/boards/sof_pcm512x.c | 9 +- sound/soc/intel/boards/sof_wm8804.c | 2 - sound/soc/mediatek/mt2701/mt2701-cs42448.c | 20 +--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 6 +- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 24 ++--- sound/soc/mediatek/mt7986/mt7986-wm8960.c | 6 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 6 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 10 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 34 +++---- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 34 +++---- sound/soc/mediatek/mt8186/mt8186-mt6366.c | 86 +++++++---------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 58 ++++++------ .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 78 +++++++--------- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 60 ++++++------ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 14 +-- sound/soc/samsung/odroid.c | 10 +- sound/soc/soc-compress.c | 15 ++- sound/soc/soc-core.c | 20 +--- sound/soc/soc-pcm.c | 93 +++++++------------ sound/soc/soc-topology-test.c | 2 - sound/soc/soc-topology.c | 4 +- sound/soc/sof/nocodec.c | 4 - 68 files changed, 315 insertions(+), 583 deletions(-) -- 2.43.0 Thank you for your help !! Best regards --- Kuninori Morimoto

1 14

Re: [PATCH v2] MAINTAINERS: Update maintainer list for MICROCHIP ASOC, SSC and MCP16502 drivers
by Nicolas Ferre 25 Sep '24

25 Sep '24

(adding Mark and alsa-devel to the thread) On 25/09/2024 at 15:07, Andrei Simion wrote: > To help Claudiu and offload the work, add myself to the maintainer list for > those drivers. > > Acked-by: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> Acked-by: Nicolas Ferre <nicolas.ferre(a)microchip.com> Thanks a lot to both of you to continue this work. Best regards, Nicolas > Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> > --- > v1 -> v2: > - add acked-by tag > - adjust commit title > - update maintainer list for MICROCHIP SSC DRIVER > --- > MAINTAINERS | 3 +++ > 1 file changed, 3 insertions(+) > > diff --git a/MAINTAINERS b/MAINTAINERS > index 09d0f09c36cc..12e6777bb6b4 100644 > --- a/MAINTAINERS > +++ b/MAINTAINERS > @@ -15088,6 +15088,7 @@ F: drivers/spi/spi-at91-usart.c > > MICROCHIP AUDIO ASOC DRIVERS > M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> > +M: Andrei Simion <andrei.simion(a)microchip.com> > L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) > S: Supported > F: Documentation/devicetree/bindings/sound/atmel* > @@ -15196,6 +15197,7 @@ F: include/video/atmel_lcdc.h > > MICROCHIP MCP16502 PMIC DRIVER > M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> > +M: Andrei Simion <andrei.simion(a)microchip.com> > L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) > S: Supported > F: Documentation/devicetree/bindings/regulator/microchip,mcp16502.yaml > @@ -15327,6 +15329,7 @@ F: drivers/spi/spi-atmel.* > > MICROCHIP SSC DRIVER > M: Claudiu Beznea <claudiu.beznea(a)tuxon.dev> > +M: Andrei Simion <andrei.simion(a)microchip.com> > L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) > S: Supported > F: Documentation/devicetree/bindings/misc/atmel-ssc.txt > > base-commit: 2b7275670032a98cba266bd1b8905f755b3e650f

1 0

[PATCH REVIEW 1/2] ASoC: codecs: lpass-rx-macro: fix RXn(rx,n) macro for DSM_CTL and SEC7 regs
by Alexey Klimov 25 Sep '24

25 Sep '24

Turns out some registers of pre-2.5 version of rxmacro codecs are not located at the expected offsets but 0xc further away in memory. So far the detected registers are CDC_RX_RX2_RX_PATH_SEC7 and CDC_RX_RX2_RX_PATH_DSM_CTL. CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) macro incorrectly generates the address 0x540 for RX2 but it should be 0x54C and it also overwrites CDC_RX_RX2_RX_PATH_SEC7 which is located at 0x540. The same goes for CDC_RX_RXn_RX_PATH_SEC7(rx, n). Fix this by introducing additional rxn_reg_stride2 offset. For 2.5 version and above this offset will be equal to 0. With such change the corresponding RXn() macros will generate the same values for 2.5 codec version for all RX paths and the same old values for pre-2.5 version for RX0 and RX1. However for the latter case with RX2 path it will also add 2 * rxn_reg_stride2 on top. Signed-off-by: Alexey Klimov <alexey.klimov(a)linaro.org> --- sound/soc/codecs/lpass-rx-macro.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 71e0d3bffd3f..9288ddb705fe 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -202,12 +202,14 @@ #define CDC_RX_RXn_RX_PATH_SEC3(rx, n) (0x042c + rx->rxn_reg_stride * n) #define CDC_RX_RX0_RX_PATH_SEC4 (0x0430) #define CDC_RX_RX0_RX_PATH_SEC7 (0x0434) -#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) (0x0434 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_SEC7(rx, n) \ + (0x0434 + rx->rxn_reg_stride * n + n * (n - 1) * rx->rxn_reg_stride2) #define CDC_RX_DSM_OUT_DELAY_SEL_MASK GENMASK(2, 0) #define CDC_RX_DSM_OUT_DELAY_TWO_SAMPLE 0x2 #define CDC_RX_RX0_RX_PATH_MIX_SEC0 (0x0438) #define CDC_RX_RX0_RX_PATH_MIX_SEC1 (0x043C) -#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) (0x0440 + rx->rxn_reg_stride * n) +#define CDC_RX_RXn_RX_PATH_DSM_CTL(rx, n) \ + (0x0440 + rx->rxn_reg_stride * n + n * (n - 1) * rx->rxn_reg_stride2) #define CDC_RX_RXn_DSM_CLK_EN_MASK BIT(0) #define CDC_RX_RX0_RX_PATH_DSM_CTL (0x0440) #define CDC_RX_RX0_RX_PATH_DSM_DATA1 (0x0444) @@ -645,6 +647,7 @@ struct rx_macro { int rx_mclk_cnt; enum lpass_codec_version codec_version; int rxn_reg_stride; + int rxn_reg_stride2; bool is_ear_mode_on; bool hph_pwr_mode; bool hph_hd2_mode; @@ -3821,6 +3824,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_0: case LPASS_CODEC_VERSION_2_1: rx->rxn_reg_stride = 0x80; + rx->rxn_reg_stride2 = 0x6; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_pre_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) @@ -3834,6 +3838,7 @@ static int rx_macro_probe(struct platform_device *pdev) case LPASS_CODEC_VERSION_2_7: case LPASS_CODEC_VERSION_2_8: rx->rxn_reg_stride = 0xc0; + rx->rxn_reg_stride2 = 0x0; def_count = ARRAY_SIZE(rx_defaults) + ARRAY_SIZE(rx_2_5_defaults); reg_defaults = kmalloc_array(def_count, sizeof(struct reg_default), GFP_KERNEL); if (!reg_defaults) -- 2.45.2

3 5

[PATCH AUTOSEL 6.6 063/139] ASoC: codecs: wsa883x: Handle reading version failure
by Sasha Levin 25 Sep '24

25 Sep '24

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 2fbf16992e5aa14acf0441320033a01a32309ded ] If reading version and variant from registers fails (which is unlikely but possible, because it is a read over bus), the driver will proceed and perform device configuration based on uninitialized stack variables. Handle it a bit better - bail out without doing any init and failing the update status Soundwire callback. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Link: https://patch.msgid.link/20240710-asoc-wsa88xx-version-v1-2-f1c54966ccde@li… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wsa883x.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index 2169d93989841..1831d4487ba9d 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -998,15 +998,19 @@ static const struct reg_sequence reg_init[] = { {WSA883X_GMAMP_SUP1, 0xE2}, }; -static void wsa883x_init(struct wsa883x_priv *wsa883x) +static int wsa883x_init(struct wsa883x_priv *wsa883x) { struct regmap *regmap = wsa883x->regmap; - int variant, version; + int variant, version, ret; - regmap_read(regmap, WSA883X_OTP_REG_0, &variant); + ret = regmap_read(regmap, WSA883X_OTP_REG_0, &variant); + if (ret) + return ret; wsa883x->variant = variant & WSA883X_ID_MASK; - regmap_read(regmap, WSA883X_CHIP_ID0, &version); + ret = regmap_read(regmap, WSA883X_CHIP_ID0, &version); + if (ret) + return ret; wsa883x->version = version; switch (wsa883x->variant) { @@ -1041,6 +1045,8 @@ static void wsa883x_init(struct wsa883x_priv *wsa883x) WSA883X_DRE_OFFSET_MASK, wsa883x->comp_offset); } + + return 0; } static int wsa883x_update_status(struct sdw_slave *slave, @@ -1049,7 +1055,7 @@ static int wsa883x_update_status(struct sdw_slave *slave, struct wsa883x_priv *wsa883x = dev_get_drvdata(&slave->dev); if (status == SDW_SLAVE_ATTACHED && slave->dev_num > 0) - wsa883x_init(wsa883x); + return wsa883x_init(wsa883x); return 0; } -- 2.43.0

1 0

Some volume controls are swapped
by GitHub issues - opened 25 Sep '24

25 Sep '24

alsa-project/alsa-lib issue #410 was opened from suddenfall: When using amixer and alsamixer, I noticed that the volume controls for certain channels are reversed. I have verified the problem with two sound cards: Terratec Aureon XFire8.0HD hifidom DSC8-MAIN Both are USB sound cards (UAC2). Both sound cards have the channel map: FL FR FC LFE RL RR SL SR (cat stream0) The channels FC/LFE are swapped with RL/RR. I am using Ubuntu 24.04. The version of libasound2t64 is 1.2.11-1build2. I am happy to help with troubleshooting. Issue URL : https://github.com/alsa-project/alsa-lib/issues/410 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH AUTOSEL 6.10 095/197] ASoC: Intel: boards: always check the result of acpi_dev_get_first_match_dev()
by Sasha Levin 25 Sep '24

25 Sep '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 14e91ddd5c02d8c3e5a682ebfa0546352b459911 ] The code seems mostly copy-pasted, with some machine drivers forgetting to test if the 'adev' result is NULL. Add this check when missing, and use -ENOENT consistently as an error code. Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Closes: https://lore.kernel.org/alsa-devel/918944d2-3d00-465e-a9d1-5d57fc966113@sta… Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240827123215.258859-4-yung-chuan.liao@linux.inte… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcht_cx2072x.c | 4 ++++ sound/soc/intel/boards/bytcht_da7213.c | 4 ++++ sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++++ sound/soc/intel/boards/cht_bsw_rt5672.c | 4 ++++ sound/soc/intel/boards/sof_es8336.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 4 ++++ 9 files changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index df3c2a7b64d23..8c2b4ab764bba 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -255,7 +255,11 @@ static int snd_byt_cht_cx2072x_probe(struct platform_device *pdev) snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); byt_cht_cx2072x_dais[dai_index].codecs->name = codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); /* override platform name, if required */ diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index 08c598b7e1eee..9178bbe8d9950 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -258,7 +258,11 @@ static int bytcht_da7213_probe(struct platform_device *pdev) snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); dailink[dai_index].codecs->name = codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); /* override platform name, if required */ diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 77b91ea4dc32c..3539c9ff0fd2c 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -562,7 +562,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) byt_cht_es8316_dais[dai_index].codecs->name = codec_name; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index db4a33680d948..4479825c08b5e 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1693,7 +1693,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) byt_rt5640_dais[dai_index].codecs->name = byt_rt5640_codec_name; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 8514b79f389bb..1f54da98aacf4 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -926,7 +926,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) byt_rt5651_dais[dai_index].codecs->name = byt_rt5651_codec_name; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 1da9ceee4d593..ac23a8b7cafca 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -582,7 +582,11 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snprintf(cht_rt5645_codec_name, sizeof(cht_rt5645_codec_name), "i2c-%s", acpi_dev_name(adev)); cht_dailink[dai_index].codecs->name = cht_rt5645_codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + /* acpi_get_first_physical_node() returns a borrowed ref, no need to deref */ codec_dev = acpi_get_first_physical_node(adev); acpi_dev_put(adev); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index d68e5bc755dee..c6c469d51243e 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -479,7 +479,11 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snprintf(drv->codec_name, sizeof(drv->codec_name), "i2c-%s", acpi_dev_name(adev)); cht_dailink[dai_index].codecs->name = drv->codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); /* Use SSP0 on Bay Trail CR devices */ diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index c1fcc156a5752..809532238c44f 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -681,7 +681,7 @@ static int sof_es8336_probe(struct platform_device *pdev) dai_links[0].codecs->dai_name = "ES8326 HiFi"; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/sof_wm8804.c b/sound/soc/intel/boards/sof_wm8804.c index 4cb0d463bf404..9c5b3f8f09f36 100644 --- a/sound/soc/intel/boards/sof_wm8804.c +++ b/sound/soc/intel/boards/sof_wm8804.c @@ -270,7 +270,11 @@ static int sof_wm8804_probe(struct platform_device *pdev) snprintf(codec_name, sizeof(codec_name), "%s%s", "i2c-", acpi_dev_name(adev)); dailink[dai_index].codecs->name = codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); snd_soc_card_set_drvdata(card, ctx); -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 082/197] ASoC: codecs: wsa883x: Handle reading version failure
by Sasha Levin 25 Sep '24

25 Sep '24

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 2fbf16992e5aa14acf0441320033a01a32309ded ] If reading version and variant from registers fails (which is unlikely but possible, because it is a read over bus), the driver will proceed and perform device configuration based on uninitialized stack variables. Handle it a bit better - bail out without doing any init and failing the update status Soundwire callback. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Link: https://patch.msgid.link/20240710-asoc-wsa88xx-version-v1-2-f1c54966ccde@li… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wsa883x.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index 2169d93989841..1831d4487ba9d 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -998,15 +998,19 @@ static const struct reg_sequence reg_init[] = { {WSA883X_GMAMP_SUP1, 0xE2}, }; -static void wsa883x_init(struct wsa883x_priv *wsa883x) +static int wsa883x_init(struct wsa883x_priv *wsa883x) { struct regmap *regmap = wsa883x->regmap; - int variant, version; + int variant, version, ret; - regmap_read(regmap, WSA883X_OTP_REG_0, &variant); + ret = regmap_read(regmap, WSA883X_OTP_REG_0, &variant); + if (ret) + return ret; wsa883x->variant = variant & WSA883X_ID_MASK; - regmap_read(regmap, WSA883X_CHIP_ID0, &version); + ret = regmap_read(regmap, WSA883X_CHIP_ID0, &version); + if (ret) + return ret; wsa883x->version = version; switch (wsa883x->variant) { @@ -1041,6 +1045,8 @@ static void wsa883x_init(struct wsa883x_priv *wsa883x) WSA883X_DRE_OFFSET_MASK, wsa883x->comp_offset); } + + return 0; } static int wsa883x_update_status(struct sdw_slave *slave, @@ -1049,7 +1055,7 @@ static int wsa883x_update_status(struct sdw_slave *slave, struct wsa883x_priv *wsa883x = dev_get_drvdata(&slave->dev); if (status == SDW_SLAVE_ATTACHED && slave->dev_num > 0) - wsa883x_init(wsa883x); + return wsa883x_init(wsa883x); return 0; } -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 111/244] ASoC: Intel: boards: always check the result of acpi_dev_get_first_match_dev()
by Sasha Levin 25 Sep '24

25 Sep '24

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 14e91ddd5c02d8c3e5a682ebfa0546352b459911 ] The code seems mostly copy-pasted, with some machine drivers forgetting to test if the 'adev' result is NULL. Add this check when missing, and use -ENOENT consistently as an error code. Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Closes: https://lore.kernel.org/alsa-devel/918944d2-3d00-465e-a9d1-5d57fc966113@sta… Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://patch.msgid.link/20240827123215.258859-4-yung-chuan.liao@linux.inte… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcht_cx2072x.c | 4 ++++ sound/soc/intel/boards/bytcht_da7213.c | 4 ++++ sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 ++++ sound/soc/intel/boards/cht_bsw_rt5672.c | 4 ++++ sound/soc/intel/boards/sof_es8336.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 4 ++++ 9 files changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index df3c2a7b64d23..8c2b4ab764bba 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -255,7 +255,11 @@ static int snd_byt_cht_cx2072x_probe(struct platform_device *pdev) snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); byt_cht_cx2072x_dais[dai_index].codecs->name = codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); /* override platform name, if required */ diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index 08c598b7e1eee..9178bbe8d9950 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -258,7 +258,11 @@ static int bytcht_da7213_probe(struct platform_device *pdev) snprintf(codec_name, sizeof(codec_name), "i2c-%s", acpi_dev_name(adev)); dailink[dai_index].codecs->name = codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); /* override platform name, if required */ diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 77b91ea4dc32c..3539c9ff0fd2c 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -562,7 +562,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) byt_cht_es8316_dais[dai_index].codecs->name = codec_name; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index db4a33680d948..4479825c08b5e 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1693,7 +1693,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) byt_rt5640_dais[dai_index].codecs->name = byt_rt5640_codec_name; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 8514b79f389bb..1f54da98aacf4 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -926,7 +926,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) byt_rt5651_dais[dai_index].codecs->name = byt_rt5651_codec_name; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index 1da9ceee4d593..ac23a8b7cafca 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -582,7 +582,11 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snprintf(cht_rt5645_codec_name, sizeof(cht_rt5645_codec_name), "i2c-%s", acpi_dev_name(adev)); cht_dailink[dai_index].codecs->name = cht_rt5645_codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + /* acpi_get_first_physical_node() returns a borrowed ref, no need to deref */ codec_dev = acpi_get_first_physical_node(adev); acpi_dev_put(adev); diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index d68e5bc755dee..c6c469d51243e 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -479,7 +479,11 @@ static int snd_cht_mc_probe(struct platform_device *pdev) snprintf(drv->codec_name, sizeof(drv->codec_name), "i2c-%s", acpi_dev_name(adev)); cht_dailink[dai_index].codecs->name = drv->codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); /* Use SSP0 on Bay Trail CR devices */ diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 2a88efaa6d26b..b45d0501f1090 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -681,7 +681,7 @@ static int sof_es8336_probe(struct platform_device *pdev) dai_links[0].codecs->dai_name = "ES8326 HiFi"; } else { dev_err(dev, "Error cannot find '%s' dev\n", mach->id); - return -ENXIO; + return -ENOENT; } codec_dev = acpi_get_first_physical_node(adev); diff --git a/sound/soc/intel/boards/sof_wm8804.c b/sound/soc/intel/boards/sof_wm8804.c index b2d02cc92a6a8..0a5ce34d7f7bb 100644 --- a/sound/soc/intel/boards/sof_wm8804.c +++ b/sound/soc/intel/boards/sof_wm8804.c @@ -270,7 +270,11 @@ static int sof_wm8804_probe(struct platform_device *pdev) snprintf(codec_name, sizeof(codec_name), "%s%s", "i2c-", acpi_dev_name(adev)); dailink[dai_index].codecs->name = codec_name; + } else { + dev_err(&pdev->dev, "Error cannot find '%s' dev\n", mach->id); + return -ENOENT; } + acpi_dev_put(adev); snd_soc_card_set_drvdata(card, ctx); -- 2.43.0

1 0

[PATCH AUTOSEL 6.11 098/244] ASoC: codecs: wsa883x: Handle reading version failure
by Sasha Levin 25 Sep '24

25 Sep '24

From: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> [ Upstream commit 2fbf16992e5aa14acf0441320033a01a32309ded ] If reading version and variant from registers fails (which is unlikely but possible, because it is a read over bus), the driver will proceed and perform device configuration based on uninitialized stack variables. Handle it a bit better - bail out without doing any init and failing the update status Soundwire callback. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Link: https://patch.msgid.link/20240710-asoc-wsa88xx-version-v1-2-f1c54966ccde@li… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wsa883x.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wsa883x.c b/sound/soc/codecs/wsa883x.c index 3e4fdaa3f44fb..53f6de4340548 100644 --- a/sound/soc/codecs/wsa883x.c +++ b/sound/soc/codecs/wsa883x.c @@ -997,15 +997,19 @@ static const struct reg_sequence reg_init[] = { {WSA883X_GMAMP_SUP1, 0xE2}, }; -static void wsa883x_init(struct wsa883x_priv *wsa883x) +static int wsa883x_init(struct wsa883x_priv *wsa883x) { struct regmap *regmap = wsa883x->regmap; - int variant, version; + int variant, version, ret; - regmap_read(regmap, WSA883X_OTP_REG_0, &variant); + ret = regmap_read(regmap, WSA883X_OTP_REG_0, &variant); + if (ret) + return ret; wsa883x->variant = variant & WSA883X_ID_MASK; - regmap_read(regmap, WSA883X_CHIP_ID0, &version); + ret = regmap_read(regmap, WSA883X_CHIP_ID0, &version); + if (ret) + return ret; wsa883x->version = version; switch (wsa883x->variant) { @@ -1040,6 +1044,8 @@ static void wsa883x_init(struct wsa883x_priv *wsa883x) WSA883X_DRE_OFFSET_MASK, wsa883x->comp_offset); } + + return 0; } static int wsa883x_update_status(struct sdw_slave *slave, @@ -1048,7 +1054,7 @@ static int wsa883x_update_status(struct sdw_slave *slave, struct wsa883x_priv *wsa883x = dev_get_drvdata(&slave->dev); if (status == SDW_SLAVE_ATTACHED && slave->dev_num > 0) - wsa883x_init(wsa883x); + return wsa883x_init(wsa883x); return 0; } -- 2.43.0

1 0

rt722: add mic led support
by GitHub pull_request - opened 25 Sep '24

25 Sep '24

alsa-project/alsa-ucm-conf pull request #443 was opened from shumingfan: rt722: add mic led support Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/443 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/443.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: rt721-sdca: Add RT721 SDCA driver
by Jack Yu 25 Sep '24

25 Sep '24

This is the initial codec driver for rt721-sdca. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt721-sdca-sdw.c | 550 +++++++++ sound/soc/codecs/rt721-sdca-sdw.h | 150 +++ sound/soc/codecs/rt721-sdca.c | 1790 +++++++++++++++++++++++++++++ sound/soc/codecs/rt721-sdca.h | 277 +++++ 6 files changed, 2776 insertions(+) create mode 100644 sound/soc/codecs/rt721-sdca-sdw.c create mode 100644 sound/soc/codecs/rt721-sdca-sdw.h create mode 100644 sound/soc/codecs/rt721-sdca.c create mode 100644 sound/soc/codecs/rt721-sdca.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7092842480ef..74c14da4ae09 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -222,6 +222,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT712_SDCA_DMIC_SDW imply SND_SOC_RT715_SDW imply SND_SOC_RT715_SDCA_SDW + imply SND_SOC_RT721_SDCA_SDW imply SND_SOC_RT722_SDCA_SDW imply SND_SOC_RT1308_SDW imply SND_SOC_RT1316_SDW @@ -1743,6 +1744,12 @@ config SND_SOC_RT712_SDCA_DMIC_SDW select REGMAP_SOUNDWIRE select REGMAP_SOUNDWIRE_MBQ +config SND_SOC_RT721_SDCA_SDW + tristate "Realtek RT721 SDCA Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + select REGMAP_SOUNDWIRE_MBQ + config SND_SOC_RT722_SDCA_SDW tristate "Realtek RT722 SDCA Codec - SDW" depends on SOUNDWIRE diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 54cbc3feae32..3477c6352acc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -259,6 +259,7 @@ snd-soc-rt712-sdca-y := rt712-sdca.o rt712-sdca-sdw.o snd-soc-rt712-sdca-dmic-y := rt712-sdca-dmic.o snd-soc-rt715-y := rt715.o rt715-sdw.o snd-soc-rt715-sdca-y := rt715-sdca.o rt715-sdca-sdw.o +snd-soc-rt721-sdca-y := rt721-sdca.o rt721-sdca-sdw.o snd-soc-rt722-sdca-y := rt722-sdca.o rt722-sdca-sdw.o snd-soc-rt9120-y := rt9120.o snd-soc-rtq9128-y := rtq9128.o @@ -662,6 +663,7 @@ obj-$(CONFIG_SND_SOC_RT712_SDCA_SDW) += snd-soc-rt712-sdca.o obj-$(CONFIG_SND_SOC_RT712_SDCA_DMIC_SDW) += snd-soc-rt712-sdca-dmic.o obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o obj-$(CONFIG_SND_SOC_RT715_SDCA_SDW) += snd-soc-rt715-sdca.o +obj-$(CONFIG_SND_SOC_RT721_SDCA_SDW) += snd-soc-rt721-sdca.o obj-$(CONFIG_SND_SOC_RT722_SDCA_SDW) += snd-soc-rt722-sdca.o obj-$(CONFIG_SND_SOC_RT9120) += snd-soc-rt9120.o obj-$(CONFIG_SND_SOC_RTQ9128) += snd-soc-rtq9128.o diff --git a/sound/soc/codecs/rt721-sdca-sdw.c b/sound/soc/codecs/rt721-sdca-sdw.c new file mode 100644 index 000000000000..3a8dbbc01152 --- /dev/null +++ b/sound/soc/codecs/rt721-sdca-sdw.c @@ -0,0 +1,550 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt721-sdca-sdw.c -- rt721 SDCA ALSA SoC audio driver +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/pm_runtime.h> +#include <linux/soundwire/sdw_registers.h> + +#include "rt721-sdca.h" +#include "rt721-sdca-sdw.h" + +static bool rt721_sdca_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f01 ... 0x2f0a: + case 0x2f35: + case 0x2f50: + case 0x2f51: + case 0x2f58 ... 0x2f5d: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XUV, + RT721_SDCA_CTL_XUV, 0): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_SELECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_CURRENT_OWNER, 0) ... SDW_SDCA_CTL(FUNC_NUM_HID, + RT721_SDCA_ENT_HID01, RT721_SDCA_CTL_HIDTX_MESSAGE_LENGTH, 0): + case RT721_BUF_ADDR_HID1 ... RT721_BUF_ADDR_HID2: + return true; + default: + return false; + } +} + +static bool rt721_sdca_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f01: + case 0x2f51: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XUV, + RT721_SDCA_CTL_XUV, 0): + case SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_CURRENT_OWNER, 0) ... SDW_SDCA_CTL(FUNC_NUM_HID, + RT721_SDCA_ENT_HID01, RT721_SDCA_CTL_HIDTX_MESSAGE_LENGTH, 0): + case RT721_BUF_ADDR_HID1 ... RT721_BUF_ADDR_HID2: + return true; + default: + return false; + } +} + +static bool rt721_sdca_mbq_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0900007: + case 0x0a00005: + case 0x0c00005: + case 0x0d00014: + case 0x0310100: + case 0x2000001: + case 0x2000002: + case 0x2000003: + case 0x2000013: + case 0x200003c: + case 0x2000046: + case 0x5810000: + case 0x5810036: + case 0x5810037: + case 0x5810038: + case 0x5810039: + case 0x5b10018: + case 0x5b10019: + case 0x5f00045: + case 0x5f00048: + case 0x6100000: + case 0x6100005: + case 0x6100006: + case 0x610000d: + case 0x6100010: + case 0x6100011: + case 0x6100013: + case 0x6100015: + case 0x6100017: + case 0x6100025: + case 0x6100029: + case 0x610002c ... 0x610002f: + case 0x6100053 ... 0x6100055: + case 0x6100057: + case 0x610005a: + case 0x610005b: + case 0x610006a: + case 0x610006d: + case 0x6100092: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_L): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_R): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_L): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_R): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_R): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_01): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_02): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_03): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_04): + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_R): + return true; + default: + return false; + } +} + +static bool rt721_sdca_mbq_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0310100: + case 0x0a00005: + case 0x0c00005: + case 0x0d00014: + case 0x2000000: + case 0x200000d: + case 0x2000019: + case 0x2000020: + case 0x2000030: + case 0x2000046: + case 0x2000067: + case 0x2000084: + case 0x2000086: + case 0x5810000: + case 0x5810036: + case 0x5810037: + case 0x5810038: + case 0x5810039: + case 0x5b10018: + case 0x5b10019: + return true; + default: + return false; + } +} + +static const struct regmap_config rt721_sdca_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt721_sdca_readable_register, + .volatile_reg = rt721_sdca_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = rt721_sdca_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt721_sdca_reg_defaults), + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct regmap_config rt721_sdca_mbq_regmap = { + .name = "sdw-mbq", + .reg_bits = 32, + .val_bits = 16, + .readable_reg = rt721_sdca_mbq_readable_register, + .volatile_reg = rt721_sdca_mbq_volatile_register, + .max_register = 0x41000312, + .reg_defaults = rt721_sdca_mbq_defaults, + .num_reg_defaults = ARRAY_SIZE(rt721_sdca_mbq_defaults), + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt721_sdca_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(&slave->dev); + + if (status == SDW_SLAVE_UNATTACHED) + rt721->hw_init = false; + + if (status == SDW_SLAVE_ATTACHED) { + if (rt721->hs_jack) { + /* + * Due to the SCP_SDCA_INTMASK will be cleared by any reset, and then + * if the device attached again, we will need to set the setting back. + * It could avoid losing the jack detection interrupt. + * This also could sync with the cache value as the rt721_sdca_jack_init set. + */ + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, + SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + } + } + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt721->hw_init || status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt721_sdca_io_init(&slave->dev, slave); +} + +static int rt721_sdca_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + sdw_slave_read_prop(slave); + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* + * port = 1 for headphone playback + * port = 2 for headset-mic capture + * port = 3 for speaker playback + * port = 6 for digital-mic capture + */ + prop->source_ports = BIT(6) | BIT(2); /* BITMAP: 01000100 */ + prop->sink_ports = BIT(3) | BIT(1); /* BITMAP: 00001010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 200; + + /* wake-up event */ + prop->wake_capable = 1; + + /* Three data lanes are supported by rt721-sdca codec */ + prop->lane_control_support = true; + + return 0; +} + +static int rt721_sdca_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(&slave->dev); + int ret, stat; + int count = 0, retry = 3; + unsigned int sdca_cascade, scp_sdca_stat1, scp_sdca_stat2 = 0; + + if (cancel_delayed_work_sync(&rt721->jack_detect_work)) { + dev_warn(&slave->dev, "%s the pending delayed_work was cancelled", __func__); + /* avoid the HID owner doesn't change to device */ + if (rt721->scp_sdca_stat2) + scp_sdca_stat2 = rt721->scp_sdca_stat2; + } + + /* + * The critical section below intentionally protects a rather large piece of code. + * We don't want to allow the system suspend to disable an interrupt while we are + * processing it, which could be problematic given the quirky SoundWire interrupt + * scheme. We do want however to prevent new workqueues from being scheduled if + * the disable_irq flag was set during system suspend. + */ + mutex_lock(&rt721->disable_irq_lock); + + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + + rt721->scp_sdca_stat1 = ret; + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + + rt721->scp_sdca_stat2 = ret; + if (scp_sdca_stat2) + rt721->scp_sdca_stat2 |= scp_sdca_stat2; + do { + /* clear flag */ + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + if (ret & SDW_SCP_SDCA_INTMASK_SDCA_0) { + ret = sdw_update_no_pm(rt721->slave, SDW_SCP_SDCA_INT1, + SDW_SCP_SDCA_INT_SDCA_0, SDW_SCP_SDCA_INT_SDCA_0); + if (ret < 0) + goto io_error; + } else if (ret & SDW_SCP_SDCA_INTMASK_SDCA_6) { + ret = sdw_update_no_pm(rt721->slave, SDW_SCP_SDCA_INT1, + SDW_SCP_SDCA_INT_SDCA_6, SDW_SCP_SDCA_INT_SDCA_6); + if (ret < 0) + goto io_error; + } + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + if (ret & SDW_SCP_SDCA_INTMASK_SDCA_8) { + ret = sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INT2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + if (ret < 0) + goto io_error; + } + + /* check if flag clear or not */ + ret = sdw_read_no_pm(rt721->slave, SDW_DP0_INT); + if (ret < 0) + goto io_error; + sdca_cascade = ret & SDW_DP0_SDCA_CASCADE; + + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + scp_sdca_stat1 = ret & SDW_SCP_SDCA_INTMASK_SDCA_0; + + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + scp_sdca_stat2 = ret & SDW_SCP_SDCA_INTMASK_SDCA_8; + + stat = scp_sdca_stat1 || scp_sdca_stat2 || sdca_cascade; + + count++; + } while (stat != 0 && count < retry); + + if (stat) + dev_warn(&slave->dev, + "%s scp_sdca_stat1=0x%x, scp_sdca_stat2=0x%x\n", __func__, + rt721->scp_sdca_stat1, rt721->scp_sdca_stat2); + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + + if (status->sdca_cascade && !rt721->disable_irq) + mod_delayed_work(system_power_efficient_wq, + &rt721->jack_detect_work, msecs_to_jiffies(280)); + + mutex_unlock(&rt721->disable_irq_lock); + + return 0; + +io_error: + mutex_unlock(&rt721->disable_irq_lock); + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +} + +static const struct sdw_slave_ops rt721_sdca_slave_ops = { + .read_prop = rt721_sdca_read_prop, + .interrupt_callback = rt721_sdca_interrupt_callback, + .update_status = rt721_sdca_update_status, +}; + +static int rt721_sdca_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap, *mbq_regmap; + + /* Regmap Initialization */ + mbq_regmap = devm_regmap_init_sdw_mbq(slave, &rt721_sdca_mbq_regmap); + if (IS_ERR(mbq_regmap)) + return PTR_ERR(mbq_regmap); + + regmap = devm_regmap_init_sdw(slave, &rt721_sdca_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return rt721_sdca_init(&slave->dev, regmap, mbq_regmap, slave); +} + +static int rt721_sdca_sdw_remove(struct sdw_slave *slave) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(&slave->dev); + + if (rt721->hw_init) { + cancel_delayed_work_sync(&rt721->jack_detect_work); + cancel_delayed_work_sync(&rt721->jack_btn_check_work); + } + + if (rt721->first_hw_init) + pm_runtime_disable(&slave->dev); + + mutex_destroy(&rt721->calibrate_mutex); + mutex_destroy(&rt721->disable_irq_lock); + + return 0; +} + +static const struct sdw_device_id rt721_sdca_id[] = { + SDW_SLAVE_ENTRY_EXT(0x025d, 0x721, 0x3, 0x1, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt721_sdca_id); + +static int __maybe_unused rt721_sdca_dev_suspend(struct device *dev) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev); + + if (!rt721->hw_init) + return 0; + + cancel_delayed_work_sync(&rt721->jack_detect_work); + cancel_delayed_work_sync(&rt721->jack_btn_check_work); + + regcache_cache_only(rt721->regmap, true); + regcache_cache_only(rt721->mbq_regmap, true); + + return 0; +} + +static int __maybe_unused rt721_sdca_dev_system_suspend(struct device *dev) +{ + struct rt721_sdca_priv *rt721_sdca = dev_get_drvdata(dev); + struct sdw_slave *slave = dev_to_sdw_dev(dev); + int ret1, ret2; + + if (!rt721_sdca->hw_init) + return 0; + + /* + * prevent new interrupts from being handled after the + * deferred work completes and before the parent disables + * interrupts on the link + */ + mutex_lock(&rt721_sdca->disable_irq_lock); + rt721_sdca->disable_irq = true; + ret1 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK1, + SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6, 0); + ret2 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK2, + SDW_SCP_SDCA_INTMASK_SDCA_8, 0); + mutex_unlock(&rt721_sdca->disable_irq_lock); + + if (ret1 < 0 || ret2 < 0) { + /* log but don't prevent suspend from happening */ + dev_dbg(&slave->dev, "%s: could not disable SDCA interrupts\n:", __func__); + } + + return rt721_sdca_dev_suspend(dev); +} + +#define RT721_PROBE_TIMEOUT 5000 + +static int __maybe_unused rt721_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt721->first_hw_init) + return 0; + + if (!slave->unattach_request) { + mutex_lock(&rt721->disable_irq_lock); + if (rt721->disable_irq == true) { + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt721->disable_irq = false; + } + mutex_unlock(&rt721->disable_irq_lock); + goto regmap_sync; + } + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT721_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt721->regmap, false); + regcache_sync(rt721->regmap); + regcache_cache_only(rt721->mbq_regmap, false); + regcache_sync(rt721->mbq_regmap); + return 0; +} + +static const struct dev_pm_ops rt721_sdca_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt721_sdca_dev_system_suspend, rt721_sdca_dev_resume) + SET_RUNTIME_PM_OPS(rt721_sdca_dev_suspend, rt721_sdca_dev_resume, NULL) +}; + +static struct sdw_driver rt721_sdca_sdw_driver = { + .driver = { + .name = "rt721-sdca", + .owner = THIS_MODULE, + .pm = &rt721_sdca_pm, + }, + .probe = rt721_sdca_sdw_probe, + .remove = rt721_sdca_sdw_remove, + .ops = &rt721_sdca_slave_ops, + .id_table = rt721_sdca_id, +}; +module_sdw_driver(rt721_sdca_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT721 SDCA SDW driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt721-sdca-sdw.h b/sound/soc/codecs/rt721-sdca-sdw.h new file mode 100644 index 000000000000..214b31b82583 --- /dev/null +++ b/sound/soc/codecs/rt721-sdca-sdw.h @@ -0,0 +1,150 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt721-sdca-sdw.h -- RT721 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2024 Realtek Semiconductor Corp. + */ + +#ifndef __RT721_SDW_H__ +#define __RT721_SDW_H__ + +#include <linux/regmap.h> +#include <linux/soundwire/sdw_registers.h> + +static const struct reg_default rt721_sdca_reg_defaults[] = { + { 0x202d, 0x00 }, + { 0x2f01, 0x00 }, + { 0x2f02, 0x09 }, + { 0x2f03, 0x08 }, + { 0x2f04, 0x00 }, + { 0x2f05, 0x0e }, + { 0x2f06, 0x01 }, + { 0x2f09, 0x00 }, + { 0x2f0a, 0x00 }, + { 0x2f35, 0x00 }, + { 0x2f50, 0xf0 }, + { 0x2f58, 0x07 }, + { 0x2f59, 0x07 }, + { 0x2f5a, 0x00 }, + { 0x2f5b, 0x07 }, + { 0x2f5c, 0x27 }, + { 0x2f5d, 0x07 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS01, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS11, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE12, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE40, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_03), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_04), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_CS1F, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_IT26, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_PDE2A, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_CS31, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_OT23, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x01 }, +}; + +static const struct reg_default rt721_sdca_mbq_defaults[] = { + { 0x0900007, 0xc004 }, + { 0x2000001, 0x0000 }, + { 0x2000002, 0x0000 }, + { 0x2000003, 0x0000 }, + { 0x2000013, 0x8001 }, + { 0x200003c, 0x0000 }, + { 0x2000046, 0x3400 }, + { 0x5f00044, 0x6040 }, + { 0x5f00045, 0x3333 }, + { 0x5f00048, 0x0000 }, + { 0x6100005, 0x0005 }, + { 0x6100006, 0x0000 }, + { 0x610000d, 0x0051 }, + { 0x6100010, 0x0180 }, + { 0x6100011, 0x0000 }, + { 0x6100013, 0x0000 }, + { 0x6100015, 0x0000 }, + { 0x6100017, 0x8049 }, + { 0x6100025, 0x1000 }, + { 0x6100029, 0x0809 }, + { 0x610002c, 0x2828 }, + { 0x610002d, 0x2929 }, + { 0x610002e, 0x3529 }, + { 0x610002f, 0x2901 }, + { 0x6100053, 0x2630 }, + { 0x6100054, 0x2a2a }, + { 0x6100055, 0x152f }, + { 0x6100057, 0x2200 }, + { 0x610005a, 0x2a4b }, + { 0x610005b, 0x2a00 }, + { 0x610006a, 0x0102 }, + { 0x610006d, 0x0102 }, + { 0x6100092, 0x4f61 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_L), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_R), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_L), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_R), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, RT721_SDCA_CTL_FU_CH_GAIN, + CH_L), 0xfe00 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, RT721_SDCA_CTL_FU_CH_GAIN, + CH_R), 0xfe00 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_01), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_02), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_03), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_04), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_01), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_02), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_03), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_04), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_L), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_R), + 0x0000 }, +}; + +#endif /* __RT721_SDW_H__ */ diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c new file mode 100644 index 000000000000..cca749ebbd31 --- /dev/null +++ b/sound/soc/codecs/rt721-sdca.c @@ -0,0 +1,1790 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt721-sdca.c -- rt721 SDCA ALSA SoC audio driver +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// +// + +#include <linux/bitops.h> +#include <sound/core.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <sound/initval.h> +#include <sound/jack.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/pcm.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/slab.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "rt721-sdca.h" + +static void rt721_sdca_dmic_preset(struct rt721_sdca_priv *rt721) +{ + /* Power down group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8000); + /* VREF_HV_EN_AUTO_FAST */ + rt721_sdca_index_write(rt721, RT721_ANA_POW_PART, + RT721_VREF1_HV_CTRL1, 0xe000); + /* Power up group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8007); + /* Set AD07/08 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL9, 0x2a2a); + /* Set AD10 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL10, 0x2a00); + /* Set DMIC1/DMIC2 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL6, 0x2a2a); + /* Set DMIC1/DMIC2 IT entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL5, 0x2626); + /* Set AD10 FU entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL8, 0x1e00); + /* Set DMIC1/DMIC2 FU input gain floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL7, 0x1515); + /* Set DMIC2 FU input gain channel floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_CH_FLOAT_CTL3, 0x0304); + /* Set AD10 FU channel floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_CH_FLOAT_CTL4, 0x0304); + /* vf71f_r12_07_06 and vf71f_r13_07_06 = 2’b00 */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_CTL1, 0x0000); + /* Enable vf707_r12_05/vf707_r13_05 */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_IT26, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x01); + /* Set usd_flag_sel, usd_in_sel */ + regmap_write(rt721->mbq_regmap, 0x5910009, 0x2e01); + /* Set RC calibration */ + rt721_sdca_index_write(rt721, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b00); + rt721_sdca_index_write(rt721, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b40); + /* Fine tune PDE2A latency */ + regmap_write(rt721->regmap, 0x2f5c, 0x25); +} + +static void rt721_sdca_amp_preset(struct rt721_sdca_priv *rt721) +{; + /* Power down group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8000); + /* VREF_HV_EN_AUTO_FAST */ + rt721_sdca_index_write(rt721, RT721_ANA_POW_PART, + RT721_VREF1_HV_CTRL1, 0xe000); + /* Power up group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8007); + /* Reset dc_cal_top */ + regmap_write(rt721->mbq_regmap, 0x5810000, 0x6420); + /* Turn back to normal dc_cal_top */ + regmap_write(rt721->mbq_regmap, 0x5810000, 0x6421); + /* W1C Trigger Calibration */ + regmap_write(rt721->mbq_regmap, 0x5810000, 0xe421); + /* DAC04 FU entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_CH_FLOAT_CTL6, 0x5561); + /* Set EAPD high */ + rt721_sdca_index_write(rt721, RT721_VENDOR_REG, + RT721_GPIO_PAD_CTRL5, 0x8003); + /* Enable vf707_r14 */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_OT23, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x04); + /* FU 23 SPK mute control - L */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x00); + /* FU 23 SPK mute control - R */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x00); + /* FU 55 DAC04 mute control - L */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x00); + /* FU 55 DAC04 mute control - R */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x00); +} + +static void rt721_sdca_jack_preset(struct rt721_sdca_priv *rt721) +{ + /* Power down group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8000); + /* VREF_HV_EN_AUTO_FAST */ + rt721_sdca_index_write(rt721, RT721_ANA_POW_PART, + RT721_VREF1_HV_CTRL1, 0xe000); + /* Power up group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8007); + /* GE0 mode related control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_GE_REL_CTRL1, 0x8011); + /* Button A, B, C, D bypass mode */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL3, 0xcf00); + /* HID1 slot enable */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL4, 0x000f); + /* Report ID for HID1 */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL1, 0x1100); + /* OSC/OOC for slot 2, 3 */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL5, 0x0c12); + /* Set JD de-bounce clock control */ + rt721_sdca_index_write(rt721, RT721_JD_CTRL, + RT721_JD_1PIN_GAT_CTRL2, 0xc002); + /* RC calibration -1 */ + rt721_sdca_index_write(rt721, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b00); + /* RC calibration -2 */ + rt721_sdca_index_write(rt721, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b40); + /* pow_clk_12p288mhz_dre03 change to register mode */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON14, 0x3333); + /* Tune calibration timing control */ + regmap_write(rt721->mbq_regmap, 0x5810035, 0x0036); + /* calibration HP amp output select control from Efuse */ + regmap_write(rt721->mbq_regmap, 0x5810030, 0xee00); + /* FSM related control */ + rt721_sdca_index_write(rt721, RT721_CAP_PORT_CTRL, + RT721_HP_AMP_2CH_CAL1, 0x0140); + /* HP calibration related control */ + regmap_write(rt721->mbq_regmap, 0x5810000, 0x0021); + /* W1C HP calibration*/ + regmap_write(rt721->mbq_regmap, 0x5810000, 0x8021); + /* reg_sel_cin_hp_0010/0011 */ + rt721_sdca_index_write(rt721, RT721_CAP_PORT_CTRL, + RT721_HP_AMP_2CH_CAL18, 0x5522); + regmap_write(rt721->mbq_regmap, 0x5b10007, 0x2000); + /* sel_sensing_lr_hp */ + regmap_write(rt721->mbq_regmap, 0x5B10017, 0x1b0f); + /* Release HP-JD */ + rt721_sdca_index_write(rt721, RT721_CBJ_CTRL, + RT721_CBJ_A0_GAT_CTRL1, 0x2a02); + /* en_osw gating auto done bit */ + rt721_sdca_index_write(rt721, RT721_CAP_PORT_CTRL, + RT721_HP_AMP_2CH_CAL4, 0xa105); + /* pow_clk_en_sw_amp_detect_sel to register mode */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON14, 0x3b33); + /* cp_sw_hp to auto mode */ + regmap_write(rt721->mbq_regmap, 0x310400, 0x3023); + /* pow_clk_en_sw_amp_detect power up */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON14, 0x3f33); + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON13, 0x6048); + /* switch size detect threshold */ + regmap_write(rt721->mbq_regmap, 0x310401, 0x3000); + regmap_write(rt721->mbq_regmap, 0x310402, 0x1b00); + /* en_hp_amp_detect auto mode */ + regmap_write(rt721->mbq_regmap, 0x310300, 0x000f); + /* amp detect threshold */ + regmap_write(rt721->mbq_regmap, 0x310301, 0x3000); + regmap_write(rt721->mbq_regmap, 0x310302, 0x1b00); + /* gating_sdw_link_rst_n_1_cbj_reg */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON17, 0x0008); + /* CKXEN_SDAC chopper function */ + rt721_sdca_index_write(rt721, RT721_DAC_CTRL, + RT721_DAC_2CH_CTRL3, 0x55ff); + /* CKXSEL_SDAC chopper frequency */ + rt721_sdca_index_write(rt721, RT721_DAC_CTRL, + RT721_DAC_2CH_CTRL4, 0xcc00); + /* Bias current for SDAC */ + rt721_sdca_index_write(rt721, RT721_ANA_POW_PART, + RT721_MBIAS_LV_CTRL2, 0x6677); + /* VREF2 level selection */ + rt721_sdca_index_write(rt721, RT721_ANA_POW_PART, + RT721_VREF2_LV_CTRL1, 0x7600); + /* ADC09/MIC2 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL2, 0x1234); + /* LINE2 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL3, 0x3512); + /* DAC03/HP power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL1, 0x4040); + /* ADC27 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL4, 0x1201); + /* Fine tune PDE40 latency */ + regmap_write(rt721->regmap, 0x2f58, 0x07); +} + +int rt721_sdca_index_write(struct rt721_sdca_priv *rt721, + unsigned int nid, unsigned int reg, unsigned int value) +{ + struct regmap *regmap = rt721->mbq_regmap; + unsigned int addr = (nid << 20) | reg; + int ret; + + ret = regmap_write(regmap, addr, value); + if (ret < 0) + dev_err(&rt721->slave->dev, + "Failed to set private value: %06x <= %04x ret=%d\n", + addr, value, ret); + + return ret; +} + +int rt721_sdca_index_read(struct rt721_sdca_priv *rt721, + unsigned int nid, unsigned int reg, unsigned int *value) +{ + int ret; + struct regmap *regmap = rt721->mbq_regmap; + unsigned int addr = (nid << 20) | reg; + + ret = regmap_read(regmap, addr, value); + if (ret < 0) + dev_err(&rt721->slave->dev, + "Failed to get private value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} + +static int rt721_sdca_index_update_bits(struct rt721_sdca_priv *rt721, + unsigned int nid, unsigned int reg, unsigned int mask, unsigned int val) +{ + unsigned int tmp; + int ret; + + ret = rt721_sdca_index_read(rt721, nid, reg, &tmp); + if (ret < 0) + return ret; + + set_mask_bits(&tmp, mask, val); + return rt721_sdca_index_write(rt721, nid, reg, tmp); +} + +static int rt721_sdca_btn_type(unsigned char *buffer) +{ + if ((*buffer & 0xf0) == 0x10 || (*buffer & 0x0f) == 0x01 || + (*(buffer + 1) == 0x01) || (*(buffer + 1) == 0x10)) + return SND_JACK_BTN_2; + else if ((*buffer & 0xf0) == 0x20 || (*buffer & 0x0f) == 0x02 || + (*(buffer + 1) == 0x02) || (*(buffer + 1) == 0x20)) + return SND_JACK_BTN_3; + else if ((*buffer & 0xf0) == 0x40 || (*buffer & 0x0f) == 0x04 || + (*(buffer + 1) == 0x04) || (*(buffer + 1) == 0x40)) + return SND_JACK_BTN_0; + else if ((*buffer & 0xf0) == 0x80 || (*buffer & 0x0f) == 0x08 || + (*(buffer + 1) == 0x08) || (*(buffer + 1) == 0x80)) + return SND_JACK_BTN_1; + + return 0; +} + +static unsigned int rt721_sdca_button_detect(struct rt721_sdca_priv *rt721) +{ + unsigned int btn_type = 0, offset, idx, val, owner; + int ret; + unsigned char buf[3]; + + /* get current UMP message owner */ + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), &owner); + if (ret < 0) + return 0; + + /* if owner is device then there is no button event from device */ + if (owner == 1) + return 0; + + /* read UMP message offset */ + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), &offset); + if (ret < 0) + goto _end_btn_det_; + + for (idx = 0; idx < sizeof(buf); idx++) { + ret = regmap_read(rt721->regmap, + RT721_BUF_ADDR_HID1 + offset + idx, &val); + if (ret < 0) + goto _end_btn_det_; + buf[idx] = val & 0xff; + } + + if (buf[0] == 0x11) + btn_type = rt721_sdca_btn_type(&buf[1]); + +_end_btn_det_: + /* Host is owner, so set back to device */ + if (owner == 0) + /* set owner to device */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), 0x01); + + return btn_type; +} + +static int rt721_sdca_headset_detect(struct rt721_sdca_priv *rt721) +{ + unsigned int det_mode; + int ret; + + /* get detected_mode */ + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0), &det_mode); + + if (ret < 0) + goto io_error; + + switch (det_mode) { + case 0x00: + rt721->jack_type = 0; + break; + case 0x03: + rt721->jack_type = SND_JACK_HEADPHONE; + break; + case 0x05: + rt721->jack_type = SND_JACK_HEADSET; + break; + } + + /* write selected_mode */ + if (det_mode) { + ret = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_SELECTED_MODE, 0), det_mode); + if (ret < 0) + goto io_error; + } + + dev_dbg(&rt721->slave->dev, + "%s, detected_mode=0x%x\n", __func__, det_mode); + + return 0; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +} + +static void rt721_sdca_jack_detect_handler(struct work_struct *work) +{ + struct rt721_sdca_priv *rt721 = + container_of(work, struct rt721_sdca_priv, jack_detect_work.work); + int btn_type = 0, ret; + + if (!rt721->hs_jack) + return; + + if (!rt721->component->card || !rt721->component->card->instantiated) + return; + + /* SDW_SCP_SDCA_INT_SDCA_6 is used for jack detection */ + if (rt721->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { + ret = rt721_sdca_headset_detect(rt721); + if (ret < 0) + return; + } + + /* SDW_SCP_SDCA_INT_SDCA_8 is used for button detection */ + if (rt721->scp_sdca_stat2 & SDW_SCP_SDCA_INT_SDCA_8) + btn_type = rt721_sdca_button_detect(rt721); + + if (rt721->jack_type == 0) + btn_type = 0; + + dev_dbg(&rt721->slave->dev, + "in %s, jack_type=%d\n", __func__, rt721->jack_type); + dev_dbg(&rt721->slave->dev, + "in %s, btn_type=0x%x\n", __func__, btn_type); + pr_info("rtk debug, jack_type=%d\n", rt721->jack_type); + pr_info("rtk debug, btn_type=0x%x\n", btn_type); + dev_dbg(&rt721->slave->dev, + "in %s, scp_sdca_stat1=0x%x, scp_sdca_stat2=0x%x\n", __func__, + rt721->scp_sdca_stat1, rt721->scp_sdca_stat2); + + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt721->jack_btn_check_work, msecs_to_jiffies(200)); + } +} + +static void rt721_sdca_btn_check_handler(struct work_struct *work) +{ + struct rt721_sdca_priv *rt721 = + container_of(work, struct rt721_sdca_priv, jack_btn_check_work.work); + int btn_type = 0, ret, idx; + unsigned int det_mode, offset, val; + unsigned char buf[3]; + + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0), &det_mode); + if (ret < 0) + goto io_error; + + /* pin attached */ + if (det_mode) { + /* read UMP message offset */ + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), &offset); + if (ret < 0) + goto io_error; + + for (idx = 0; idx < sizeof(buf); idx++) { + ret = regmap_read(rt721->regmap, + RT721_BUF_ADDR_HID1 + offset + idx, &val); + if (ret < 0) + goto io_error; + buf[idx] = val & 0xff; + } + + if (buf[0] == 0x11) + btn_type = rt721_sdca_btn_type(&buf[1]); + } else + rt721->jack_type = 0; + + dev_dbg(&rt721->slave->dev, "%s, btn_type=0x%x\n", __func__, btn_type); + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt721->jack_btn_check_work, msecs_to_jiffies(200)); + } + + return; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); +} + +static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721) +{ + mutex_lock(&rt721->calibrate_mutex); + if (rt721->hs_jack) { + /* set SCP_SDCA_IntMask1[0]=1 */ + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, + SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6); + /* set SCP_SDCA_IntMask2[0]=1 */ + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + dev_dbg(&rt721->slave->dev, "in %s enable\n", __func__); + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_UAJ_CTL, 0x036E); + /* set XU(et03h) & XU(et0Dh) to Not bypassed */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XU03, + RT721_SDCA_CTL_SELECTED_MODE, 0), 0); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XU0D, + RT721_SDCA_CTL_SELECTED_MODE, 0), 0); + /* trigger GE interrupt */ + rt721_sdca_index_update_bits(rt721, RT721_HDA_SDCA_FLOAT, + RT721_GE_REL_CTRL1, 0x4000, 0x4000); + } + mutex_unlock(&rt721->calibrate_mutex); +} + +static int rt721_sdca_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + int ret; + + rt721->hs_jack = hs_jack; + + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); + return 0; + } + + rt721_sdca_jack_init(rt721); + + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + + return 0; +} + +/* For SDCA control DAC/ADC Gain */ +static int rt721_sdca_set_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned int read_l, read_r, gain_l_val, gain_r_val; + unsigned int adc_vol_flag = 0, changed = 0; + unsigned int lvalue, rvalue; + const unsigned int interval_offset = 0xc0; + const unsigned int tendA = 0x200; + const unsigned int tendB = 0xa00; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume") || + strstr(ucontrol->id.name, "FU0F Capture Volume")) + adc_vol_flag = 1; + + regmap_read(rt721->mbq_regmap, mc->reg, &lvalue); + regmap_read(rt721->mbq_regmap, mc->rreg, &rvalue); + + /* L Channel */ + gain_l_val = ucontrol->value.integer.value[0]; + if (gain_l_val > mc->max) + gain_l_val = mc->max; + + if (mc->shift == 8) { + /* boost gain */ + gain_l_val = gain_l_val * tendB; + } else if (mc->shift == 1) { + /* FU33 boost gain */ + if (gain_l_val == 0) + gain_l_val = 0x8000; + else + gain_l_val = (gain_l_val - 1) * tendA; + } else { + /* ADC/DAC gain */ + if (adc_vol_flag) + gain_l_val = 0x1e00 - ((mc->max - gain_l_val) * interval_offset); + else + gain_l_val = 0 - ((mc->max - gain_l_val) * interval_offset); + gain_l_val &= 0xffff; + } + + /* R Channel */ + gain_r_val = ucontrol->value.integer.value[1]; + if (gain_r_val > mc->max) + gain_r_val = mc->max; + + if (mc->shift == 8) { + /* boost gain */ + gain_r_val = gain_r_val * tendB; + } else if (mc->shift == 1) { + /* FU33 boost gain */ + if (gain_r_val == 0) + gain_r_val = 0x8000; + else + gain_r_val = (gain_r_val - 1) * tendA; + } else { + /* ADC/DAC gain */ + if (adc_vol_flag) + gain_r_val = 0x1e00 - ((mc->max - gain_r_val) * interval_offset); + else + gain_r_val = 0 - ((mc->max - gain_r_val) * interval_offset); + gain_r_val &= 0xffff; + } + + if (lvalue != gain_l_val || rvalue != gain_r_val) + changed = 1; + else + return 0; + + /* Lch*/ + regmap_write(rt721->mbq_regmap, mc->reg, gain_l_val); + + /* Rch */ + regmap_write(rt721->mbq_regmap, mc->rreg, gain_r_val); + + regmap_read(rt721->mbq_regmap, mc->reg, &read_l); + regmap_read(rt721->mbq_regmap, mc->rreg, &read_r); + if (read_r == gain_r_val && read_l == gain_l_val) + return changed; + + return -EIO; +} + +static int rt721_sdca_set_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int read_l, read_r, ctl_l = 0, ctl_r = 0; + unsigned int adc_vol_flag = 0; + const unsigned int interval_offset = 0xc0; + const unsigned int tendB = 0xa00; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume") || + strstr(ucontrol->id.name, "FU0F Capture Volume")) + adc_vol_flag = 1; + + regmap_read(rt721->mbq_regmap, mc->reg, &read_l); + regmap_read(rt721->mbq_regmap, mc->rreg, &read_r); + + if (mc->shift == 8) /* boost gain */ + ctl_l = read_l / tendB; + else { + if (adc_vol_flag) + ctl_l = mc->max - (((0x1e00 - read_l) & 0xffff) / interval_offset); + else + ctl_l = mc->max - (((0 - read_l) & 0xffff) / interval_offset); + } + + if (read_l != read_r) { + if (mc->shift == 8) /* boost gain */ + ctl_r = read_r / tendB; + else { /* ADC/DAC gain */ + if (adc_vol_flag) + ctl_r = mc->max - (((0x1e00 - read_r) & 0xffff) / interval_offset); + else + ctl_r = mc->max - (((0 - read_r) & 0xffff) / interval_offset); + } + } else { + ctl_r = ctl_l; + } + + ucontrol->value.integer.value[0] = ctl_l; + ucontrol->value.integer.value[1] = ctl_r; + + return 0; +} + +static int rt721_sdca_set_fu1e_capture_ctl(struct rt721_sdca_priv *rt721) +{ + int err, i; + unsigned int ch_mute; + + for (i = 0; i < ARRAY_SIZE(rt721->fu1e_mixer_mute); i++) { + ch_mute = rt721->fu1e_dapm_mute || rt721->fu1e_mixer_mute[i]; + err = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_01) + i, ch_mute); + if (err < 0) + return err; + } + + return 0; +} + +static int rt721_sdca_fu1e_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + unsigned int i; + + for (i = 0; i < p->count; i++) + ucontrol->value.integer.value[i] = !rt721->fu1e_mixer_mute[i]; + + return 0; +} + +static int rt721_sdca_fu1e_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + int err, changed = 0, i; + + for (i = 0; i < p->count; i++) { + if (rt721->fu1e_mixer_mute[i] != !ucontrol->value.integer.value[i]) + changed = 1; + rt721->fu1e_mixer_mute[i] = !ucontrol->value.integer.value[i]; + } + + err = rt721_sdca_set_fu1e_capture_ctl(rt721); + if (err < 0) + return err; + + return changed; +} + +static int rt721_sdca_set_fu0f_capture_ctl(struct rt721_sdca_priv *rt721) +{ + int err; + unsigned int ch_l, ch_r; + + ch_l = (rt721->fu0f_dapm_mute || rt721->fu0f_mixer_l_mute) ? 0x01 : 0x00; + ch_r = (rt721->fu0f_dapm_mute || rt721->fu0f_mixer_r_mute) ? 0x01 : 0x00; + + err = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_L), ch_l); + if (err < 0) + return err; + + err = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_R), ch_r); + if (err < 0) + return err; + + return 0; +} + +static int rt721_sdca_fu0f_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !rt721->fu0f_mixer_l_mute; + ucontrol->value.integer.value[1] = !rt721->fu0f_mixer_r_mute; + return 0; +} + +static int rt721_sdca_fu0f_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + int err, changed = 0; + + if (rt721->fu0f_mixer_l_mute != !ucontrol->value.integer.value[0] || + rt721->fu0f_mixer_r_mute != !ucontrol->value.integer.value[1]) + changed = 1; + + rt721->fu0f_mixer_l_mute = !ucontrol->value.integer.value[0]; + rt721->fu0f_mixer_r_mute = !ucontrol->value.integer.value[1]; + err = rt721_sdca_set_fu0f_capture_ctl(rt721); + if (err < 0) + return err; + + return changed; +} + +static int rt721_sdca_fu_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + + if (p->max == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = p->count; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = p->max; + return 0; +} + +static int rt721_sdca_dmic_set_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + unsigned int boost_step = 0x0a00; + unsigned int vol_max = 0x1e00; + unsigned int regvalue, ctl, i; + unsigned int adc_vol_flag = 0; + const unsigned int interval_offset = 0xc0; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume")) + adc_vol_flag = 1; + + /* check all channels */ + for (i = 0; i < p->count; i++) { + regmap_read(rt721->mbq_regmap, p->reg_base + i, &regvalue); + + if (!adc_vol_flag) /* boost gain */ + ctl = regvalue / boost_step; + else { /* ADC gain */ + if (adc_vol_flag) + ctl = p->max - (((vol_max - regvalue) & 0xffff) / interval_offset); + else + ctl = p->max - (((0 - regvalue) & 0xffff) / interval_offset); + } + + ucontrol->value.integer.value[i] = ctl; + } + + return 0; +} + +static int rt721_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned int boost_step = 0x0a00; + unsigned int vol_max = 0x1e00; + unsigned int gain_val[4]; + unsigned int i, adc_vol_flag = 0, changed = 0; + unsigned int regvalue[4]; + const unsigned int interval_offset = 0xc0; + int err; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume")) + adc_vol_flag = 1; + + /* check all channels */ + for (i = 0; i < p->count; i++) { + regmap_read(rt721->mbq_regmap, p->reg_base + i, &regvalue[i]); + + gain_val[i] = ucontrol->value.integer.value[i]; + if (gain_val[i] > p->max) + gain_val[i] = p->max; + + if (!adc_vol_flag) /* boost gain */ + gain_val[i] = gain_val[i] * boost_step; + else { /* ADC gain */ + gain_val[i] = vol_max - ((p->max - gain_val[i]) * interval_offset); + gain_val[i] &= 0xffff; + } + + if (regvalue[i] != gain_val[i]) + changed = 1; + } + + if (!changed) + return 0; + + for (i = 0; i < p->count; i++) { + err = regmap_write(rt721->mbq_regmap, p->reg_base + i, gain_val[i]); + if (err < 0) + dev_err(&rt721->slave->dev, "%#08x can't be set\n", p->reg_base + i); + } + + return changed; +} + +#define RT721_SDCA_PR_VALUE(xreg_base, xcount, xmax, xinvert) \ + ((unsigned long)&(struct rt721_sdca_dmic_kctrl_priv) \ + {.reg_base = xreg_base, .count = xcount, .max = xmax, \ + .invert = xinvert}) + +#define RT721_SDCA_FU_CTRL(xname, reg_base, xmax, xinvert, xcount) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = rt721_sdca_fu_info, \ + .get = rt721_sdca_fu1e_capture_get, \ + .put = rt721_sdca_fu1e_capture_put, \ + .private_value = RT721_SDCA_PR_VALUE(reg_base, xcount, xmax, xinvert)} + +#define RT721_SDCA_EXT_TLV(xname, reg_base, xhandler_get,\ + xhandler_put, xcount, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = rt721_sdca_fu_info, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = RT721_SDCA_PR_VALUE(reg_base, xcount, xmax, 0) } + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol_tlv, 0, 1000, 0); +static const DECLARE_TLV_DB_SCALE(mic2_boost_vol_tlv, -200, 200, 0); + +static const struct snd_kcontrol_new rt721_sdca_controls[] = { + /* Headphone playback settings */ + SOC_DOUBLE_R_EXT_TLV("FU05 Playback Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_VOLUME, CH_R), 0, 0x57, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, out_vol_tlv), + /* Headset mic capture settings */ + SOC_DOUBLE_EXT("FU0F Capture Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt721_sdca_fu0f_capture_get, rt721_sdca_fu0f_capture_put), + SOC_DOUBLE_R_EXT_TLV("FU0F Capture Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_VOLUME, CH_R), 0, 0x3f, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("FU33 Boost Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_R), 1, 0x15, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, mic2_boost_vol_tlv), + /* AMP playback settings */ + SOC_DOUBLE_R_EXT_TLV("FU06 Playback Volume", + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_VOLUME, CH_R), 0, 0x57, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, out_vol_tlv), + /* DMIC capture settings */ + RT721_SDCA_FU_CTRL("FU1E Capture Switch", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_01), 1, 1, 4), + RT721_SDCA_EXT_TLV("FU1E Capture Volume", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_VOLUME, CH_01), + rt721_sdca_dmic_set_gain_get, rt721_sdca_dmic_set_gain_put, + 4, 0x3f, mic_vol_tlv), + RT721_SDCA_EXT_TLV("FU15 Boost Volume", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, + RT721_SDCA_CTL_FU_CH_GAIN, CH_01), + rt721_sdca_dmic_set_gain_get, rt721_sdca_dmic_set_gain_put, + 4, 3, boost_vol_tlv), +}; + +static int rt721_sdca_adc_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned int val = 0, mask_sft, mask; + + if (strstr(ucontrol->id.name, "ADC 09 Mux")) { + mask_sft = 12; + mask = 0x7; + } else if (strstr(ucontrol->id.name, "ADC 08 R Mux")) { + mask_sft = 10; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 08 L Mux")) { + mask_sft = 8; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 R Mux")) { + mask_sft = 6; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 L Mux")) { + mask_sft = 4; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 R Mux")) { + mask_sft = 2; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 L Mux")) { + mask_sft = 0; + mask = 0x3; + } else + return -EINVAL; + + rt721_sdca_index_read(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, &val); + + ucontrol->value.enumerated.item[0] = (val >> mask_sft) & mask; + + return 0; +} + +static int rt721_sdca_adc_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val, val2 = 0, change, mask_sft, mask; + unsigned int check; + + if (item[0] >= e->items) + return -EINVAL; + + if (strstr(ucontrol->id.name, "ADC 09 Mux")) { + mask_sft = 12; + mask = 0x7; + } else if (strstr(ucontrol->id.name, "ADC 08 R Mux")) { + mask_sft = 10; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 08 L Mux")) { + mask_sft = 8; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 R Mux")) { + mask_sft = 6; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 L Mux")) { + mask_sft = 4; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 R Mux")) { + mask_sft = 2; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 L Mux")) { + mask_sft = 0; + mask = 0x3; + } else + return -EINVAL; + + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + rt721_sdca_index_read(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, &val2); + + if (strstr(ucontrol->id.name, "ADC 09 Mux")) + val2 = (val2 >> mask_sft) & 0x7; + else + val2 = (val2 >> mask_sft) & 0x3; + + if (val == val2) + change = 0; + else + change = 1; + + if (change) { + rt721_sdca_index_read(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, &check); + rt721_sdca_index_update_bits(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, mask << mask_sft, + val << mask_sft); + } + + snd_soc_dapm_mux_update_power(dapm, kcontrol, + item[0], e, NULL); + + return change; +} + +static const char * const adc09_mux_text[] = { + "MIC2", + "LINE1", + "LINE2", +}; +static const char * const adc07_10_mux_text[] = { + "DMIC1 RE", + "DMIC1 FE", + "DMIC2 RE", + "DMIC2 FE", +}; + +static SOC_ENUM_SINGLE_DECL( + rt721_adc09_enum, SND_SOC_NOPM, 0, adc09_mux_text); +static SOC_ENUM_SINGLE_DECL( + rt721_dmic_enum, SND_SOC_NOPM, 0, adc07_10_mux_text); + +static const struct snd_kcontrol_new rt721_sdca_adc09_mux = + SOC_DAPM_ENUM_EXT("ADC 09 Mux", rt721_adc09_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc08_r_mux = + SOC_DAPM_ENUM_EXT("ADC 08 R Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc08_l_mux = + SOC_DAPM_ENUM_EXT("ADC 08 L Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc10_r_mux = + SOC_DAPM_ENUM_EXT("ADC 10 R Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc10_l_mux = + SOC_DAPM_ENUM_EXT("ADC 10 L Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc07_r_mux = + SOC_DAPM_ENUM_EXT("ADC 07 R Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc07_l_mux = + SOC_DAPM_ENUM_EXT("ADC 07 L Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); + + +static int rt721_sdca_fu42_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(100); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_L), unmute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_R), unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_L), mute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_R), mute); + break; + } + return 0; +} + +static int rt721_sdca_fu21_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_L), unmute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_R), unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_L), mute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_R), mute); + break; + } + return 0; +} + +static int rt721_sdca_fu23_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_L), unmute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_R), unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_L), mute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_R), mute); + break; + } + return 0; +} + +static int rt721_sdca_fu113_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt721->fu1e_dapm_mute = false; + rt721_sdca_set_fu1e_capture_ctl(rt721); + break; + case SND_SOC_DAPM_PRE_PMD: + rt721->fu1e_dapm_mute = true; + rt721_sdca_set_fu1e_capture_ctl(rt721); + break; + } + return 0; +} + +static int rt721_sdca_fu36_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt721->fu0f_dapm_mute = false; + rt721_sdca_set_fu0f_capture_ctl(rt721); + break; + case SND_SOC_DAPM_PRE_PMD: + rt721->fu0f_dapm_mute = true; + rt721_sdca_set_fu0f_capture_ctl(rt721); + break; + } + return 0; +} + +static int rt721_sdca_pde47_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE40, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE40, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static int rt721_sdca_pde41_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE41, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE41, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static int rt721_sdca_pde11_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_PDE2A, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_PDE2A, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static int rt721_sdca_pde34_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE12, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE12, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget rt721_sdca_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("HP"), + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + SND_SOC_DAPM_INPUT("DMIC1_2"), + SND_SOC_DAPM_INPUT("DMIC3_4"), + + SND_SOC_DAPM_SUPPLY("PDE 41", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde41_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 47", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde47_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 11", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde11_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 34", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde34_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_DAC_E("FU 21", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu21_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("FU 23", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu23_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("FU 42", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu42_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("FU 36", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu36_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("FU 113", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu113_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MUX("ADC 09 Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc09_mux), + SND_SOC_DAPM_MUX("ADC 08 R Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc08_r_mux), + SND_SOC_DAPM_MUX("ADC 08 L Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc08_l_mux), + SND_SOC_DAPM_MUX("ADC 10 R Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc10_r_mux), + SND_SOC_DAPM_MUX("ADC 10 L Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc10_l_mux), + SND_SOC_DAPM_MUX("ADC 07 R Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc07_r_mux), + SND_SOC_DAPM_MUX("ADC 07 L Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc07_l_mux), + + SND_SOC_DAPM_AIF_IN("DP1RX", "DP1 Headphone Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP2TX", "DP2 Headset Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DP3RX", "DP3 Speaker Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP6TX", "DP6 DMic Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route rt721_sdca_audio_map[] = { + {"FU 42", NULL, "DP1RX"}, + {"FU 21", NULL, "DP3RX"}, + {"FU 23", NULL, "DP3RX"}, + + {"ADC 09 Mux", "MIC2", "MIC2"}, + {"ADC 09 Mux", "LINE1", "LINE1"}, + {"ADC 09 Mux", "LINE2", "LINE2"}, + {"ADC 07 R Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 07 R Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 07 R Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 07 R Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 07 L Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 07 L Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 07 L Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 07 L Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 08 R Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 08 R Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 08 R Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 08 R Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 08 L Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 08 L Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 08 L Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 08 L Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 10 R Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 10 R Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 10 R Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 10 R Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 10 L Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 10 L Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 10 L Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 10 L Mux", "DMIC2 FE", "DMIC3_4"}, + {"FU 36", NULL, "PDE 34"}, + {"FU 36", NULL, "ADC 09 Mux"}, + {"FU 113", NULL, "PDE 11"}, + {"FU 113", NULL, "ADC 07 R Mux"}, + {"FU 113", NULL, "ADC 07 L Mux"}, + {"FU 113", NULL, "ADC 10 R Mux"}, + {"FU 113", NULL, "ADC 10 L Mux"}, + {"DP2TX", NULL, "FU 36"}, + {"DP6TX", NULL, "FU 113"}, + + {"HP", NULL, "PDE 47"}, + {"HP", NULL, "FU 42"}, + {"SPK", NULL, "PDE 41"}, + {"SPK", NULL, "FU 21"}, + {"SPK", NULL, "FU 23"}, +}; + +static int rt721_sdca_parse_dt(struct rt721_sdca_priv *rt721, struct device *dev) +{ + device_property_read_u32(dev, "realtek,jd-src", &rt721->jd_src); + + return 0; +} + +static int rt721_sdca_probe(struct snd_soc_component *component) +{ + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + int ret; + + rt721_sdca_parse_dt(rt721, &rt721->slave->dev); + rt721->component = component; + + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + + return 0; +} + +static const struct snd_soc_component_driver soc_sdca_dev_rt721 = { + .probe = rt721_sdca_probe, + .controls = rt721_sdca_controls, + .num_controls = ARRAY_SIZE(rt721_sdca_controls), + .dapm_widgets = rt721_sdca_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt721_sdca_dapm_widgets), + .dapm_routes = rt721_sdca_audio_map, + .num_dapm_routes = ARRAY_SIZE(rt721_sdca_audio_map), + .set_jack = rt721_sdca_set_jack_detect, + .endianness = 1, +}; + +static int rt721_sdca_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void rt721_sdca_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int rt721_sdca_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_runtime *sdw_stream; + int retval, port, num_channels; + unsigned int sampling_rate; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!rt721->slave) + return -EINVAL; + + /* + * RT721_AIF1 with port = 1 for headphone playback + * RT721_AIF1 with port = 2 for headset-mic capture + * RT721_AIF2 with port = 3 for speaker playback + * RT721_AIF3 with port = 6 for digital-mic capture + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + if (dai->id == RT721_AIF1) + port = 1; + else if (dai->id == RT721_AIF2) + port = 3; + else + return -EINVAL; + } else { + direction = SDW_DATA_DIR_TX; + if (dai->id == RT721_AIF1) + port = 2; + else if (dai->id == RT721_AIF3) + port = 6; + else + return -EINVAL; + } + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = GENMASK(num_channels - 1, 0); + port_config.num = port; + + retval = sdw_stream_add_slave(rt721->slave, &stream_config, + &port_config, 1, sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + if (params_channels(params) > 16) { + dev_err(component->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate = RT721_SDCA_RATE_8000HZ; + break; + case 16000: + sampling_rate = RT721_SDCA_RATE_16000HZ; + break; + case 24000: + sampling_rate = RT721_SDCA_RATE_24000HZ; + break; + case 32000: + sampling_rate = RT721_SDCA_RATE_32000HZ; + break; + case 44100: + sampling_rate = RT721_SDCA_RATE_44100HZ; + break; + case 48000: + sampling_rate = RT721_SDCA_RATE_48000HZ; + break; + case 96000: + sampling_rate = RT721_SDCA_RATE_96000HZ; + break; + case 192000: + sampling_rate = RT721_SDCA_RATE_192000HZ; + break; + case 384000: + sampling_rate = RT721_SDCA_RATE_384000HZ; + break; + case 768000: + sampling_rate = RT721_SDCA_RATE_768000HZ; + break; + default: + dev_err(component->dev, "Rate %d is not supported\n", + params_rate(params)); + return -EINVAL; + } + + /* set sampling frequency */ + if (dai->id == RT721_AIF1) { + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS01, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS11, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + } + + if (dai->id == RT721_AIF2) + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_CS31, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + + if (dai->id == RT721_AIF3) + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_CS1F, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + + return 0; +} + +static int rt721_sdca_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt721->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt721->slave, sdw_stream); + return 0; +} + +#define RT721_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) +#define RT721_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops rt721_sdca_ops = { + .hw_params = rt721_sdca_pcm_hw_params, + .hw_free = rt721_sdca_pcm_hw_free, + .set_stream = rt721_sdca_set_sdw_stream, + .shutdown = rt721_sdca_shutdown, +}; + +static struct snd_soc_dai_driver rt721_sdca_dai[] = { + { + .name = "rt721-sdca-aif1", + .id = RT721_AIF1, + .playback = { + .stream_name = "DP1 Headphone Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .capture = { + .stream_name = "DP2 Headset Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .ops = &rt721_sdca_ops, + }, + { + .name = "rt721-sdca-aif2", + .id = RT721_AIF2, + .playback = { + .stream_name = "DP3 Speaker Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .ops = &rt721_sdca_ops, + }, + { + .name = "rt721-sdca-aif3", + .id = RT721_AIF3, + .capture = { + .stream_name = "DP6 DMic Capture", + .channels_min = 1, + .channels_max = 4, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .ops = &rt721_sdca_ops, + } +}; + +int rt721_sdca_init(struct device *dev, struct regmap *regmap, + struct regmap *mbq_regmap, struct sdw_slave *slave) +{ + struct rt721_sdca_priv *rt721; + + rt721 = devm_kzalloc(dev, sizeof(*rt721), GFP_KERNEL); + if (!rt721) + return -ENOMEM; + + dev_set_drvdata(dev, rt721); + rt721->slave = slave; + rt721->regmap = regmap; + rt721->mbq_regmap = mbq_regmap; + + regcache_cache_only(rt721->regmap, true); + regcache_cache_only(rt721->mbq_regmap, true); + + mutex_init(&rt721->calibrate_mutex); + mutex_init(&rt721->disable_irq_lock); + + INIT_DELAYED_WORK(&rt721->jack_detect_work, rt721_sdca_jack_detect_handler); + INIT_DELAYED_WORK(&rt721->jack_btn_check_work, rt721_sdca_btn_check_handler); + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt721->hw_init = false; + rt721->first_hw_init = false; + rt721->fu1e_dapm_mute = true; + rt721->fu0f_dapm_mute = true; + rt721->fu0f_mixer_l_mute = rt721->fu0f_mixer_r_mute = true; + rt721->fu1e_mixer_mute[0] = rt721->fu1e_mixer_mute[1] = + rt721->fu1e_mixer_mute[2] = rt721->fu1e_mixer_mute[3] = true; + + return devm_snd_soc_register_component(dev, + &soc_sdca_dev_rt721, rt721_sdca_dai, ARRAY_SIZE(rt721_sdca_dai)); +} + +int rt721_sdca_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev); + + rt721->disable_irq = false; + + if (rt721->hw_init) + return 0; + + regcache_cache_only(rt721->regmap, false); + regcache_cache_only(rt721->mbq_regmap, false); + if (rt721->first_hw_init) { + regcache_cache_bypass(rt721->regmap, true); + regcache_cache_bypass(rt721->mbq_regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + rt721_sdca_dmic_preset(rt721); + rt721_sdca_amp_preset(rt721); + rt721_sdca_jack_preset(rt721); + if (rt721->first_hw_init) { + regcache_cache_bypass(rt721->regmap, false); + regcache_mark_dirty(rt721->regmap); + regcache_cache_bypass(rt721->mbq_regmap, false); + regcache_mark_dirty(rt721->mbq_regmap); + } else + rt721->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt721->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +MODULE_DESCRIPTION("ASoC RT721 SDCA SDW driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt721-sdca.h b/sound/soc/codecs/rt721-sdca.h new file mode 100644 index 000000000000..372a788d384d --- /dev/null +++ b/sound/soc/codecs/rt721-sdca.h @@ -0,0 +1,277 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt721-sdca.h -- RT721 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2024 Realtek Semiconductor Corp. + */ + +#ifndef __RT721_H__ +#define __RT721_H__ + +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/soc.h> +#include <linux/workqueue.h> + +struct rt721_sdca_priv { + struct regmap *regmap; + struct regmap *mbq_regmap; + struct snd_soc_component *component; + struct sdw_slave *slave; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; + struct mutex calibrate_mutex; + struct mutex disable_irq_lock; + bool disable_irq; + /* For Headset jack & Headphone */ + unsigned int scp_sdca_stat1; + unsigned int scp_sdca_stat2; + struct snd_soc_jack *hs_jack; + struct delayed_work jack_detect_work; + struct delayed_work jack_btn_check_work; + int jack_type; + int jd_src; + bool fu0f_dapm_mute; + bool fu0f_mixer_l_mute; + bool fu0f_mixer_r_mute; + /* For DMIC */ + bool fu1e_dapm_mute; + bool fu1e_mixer_mute[4]; +}; + +struct rt721_sdca_dmic_kctrl_priv { + unsigned int reg_base; + unsigned int count; + unsigned int max; + unsigned int invert; +}; + +/* NID */ +#define RT721_ANA_POW_PART 0x01 +#define RT721_DAC_CTRL 0x04 +#define RT721_JD_CTRL 0x09 +#define RT721_CBJ_CTRL 0x0a +#define RT721_CAP_PORT_CTRL 0x0c +#define RT721_CLASD_AMP_CTRL 0x0d +#define RT721_VENDOR_REG 0x20 +#define RT721_RC_CALIB_CTRL 0x40 +#define RT721_VENDOR_EQ_L 0x53 +#define RT721_VENDOR_EQ_R 0x54 +#define RT721_VENDOR_HP_CALI 0x56 +#define RT721_VENDOR_CHARGE_PUMP 0x57 +#define RT721_VENDOR_CLASD_CALI 0x58 +#define RT721_VENDOR_IMS_DRE 0x5b +#define RT721_VENDOR_SPK_EFUSE 0x5c +#define RT721_VENDOR_LEVEL_CTRL 0x5d +#define RT721_VENDOR_ANA_CTL 0x5f +#define RT721_HDA_SDCA_FLOAT 0x61 + +/* Index (NID:01h) */ +#define RT721_MBIAS_LV_CTRL2 0x07 +#define RT721_VREF1_HV_CTRL1 0x0a +#define RT721_VREF2_LV_CTRL1 0x0b + +/* Index (NID:04h) */ +#define RT721_DAC_2CH_CTRL3 0x02 +#define RT721_DAC_2CH_CTRL4 0x03 + +/* Index (NID:09h) */ +#define RT721_JD_1PIN_GAT_CTRL2 0x07 + +/* Index (NID:0ah) */ +#define RT721_CBJ_A0_GAT_CTRL1 0x04 +#define RT721_CBJ_A0_GAT_CTRL2 0x05 + +/* Index (NID:0Ch) */ +#define RT721_HP_AMP_2CH_CAL1 0x05 +#define RT721_HP_AMP_2CH_CAL4 0x08 +#define RT721_HP_AMP_2CH_CAL18 0x1b + +/* Index (NID:0dh) */ +#define RT721_CLASD_AMP_2CH_CAL 0x14 + +/* Index (NID:20h) */ +#define RT721_JD_PRODUCT_NUM 0x00 +#define RT721_ANALOG_BIAS_CTL3 0x04 +#define RT721_JD_CTRL1 0x09 +#define RT721_LDO2_3_CTL1 0x0e +#define RT721_GPIO_PAD_CTRL5 0x13 +#define RT721_LDO1_CTL 0x1a +#define RT721_HP_JD_CTRL 0x24 +#define RT721_VD_HIDDEN_CTRL 0x26 +#define RT721_CLSD_CTRL6 0x3c +#define RT721_COMBO_JACK_AUTO_CTL1 0x45 +#define RT721_COMBO_JACK_AUTO_CTL2 0x46 +#define RT721_COMBO_JACK_AUTO_CTL3 0x47 +#define RT721_DIGITAL_MISC_CTRL4 0x4a +#define RT721_VREFO_GAT 0x63 +#define RT721_FSM_CTL 0x67 +#define RT721_SDCA_INTR_REC 0x82 +#define RT721_SW_CONFIG1 0x8a +#define RT721_SW_CONFIG2 0x8b + +/* Index (NID:40h) */ +#define RT721_RC_CALIB_CTRL0 0x00 + +/* Index (NID:58h) */ +#define RT721_DAC_DC_CALI_CTL1 0x01 +#define RT721_DAC_DC_CALI_CTL2 0x02 +#define RT721_DAC_DC_CALI_CTL3 0x03 + +/* Index (NID:5fh) */ +#define RT721_MISC_POWER_CTL0 0x00 +#define RT721_MISC_POWER_CTL31 0x31 +#define RT721_UAJ_TOP_TCON13 0x44 +#define RT721_UAJ_TOP_TCON14 0x45 +#define RT721_UAJ_TOP_TCON17 0x48 + +/* Index (NID:61h) */ +#define RT721_HDA_LEGACY_MUX_CTL0 0x00 +#define RT721_HDA_LEGACY_UAJ_CTL 0x02 +#define RT721_HDA_LEGACY_CTL1 0x05 +#define RT721_HDA_LEGACY_RESET_CTL 0x06 +#define RT721_GE_REL_CTRL1 0x0d +#define RT721_HDA_LEGACY_GPIO_WAKE_EN_CTL 0x0e +#define RT721_GE_SDCA_RST_CTRL 0x10 +#define RT721_INT_RST_EN_CTRL 0x11 +#define RT721_XU_EVENT_EN 0x13 +#define RT721_INLINE_CTL2 0x17 +#define RT721_UMP_HID_CTRL1 0x18 +#define RT721_UMP_HID_CTRL2 0x19 +#define RT721_UMP_HID_CTRL3 0x1a +#define RT721_UMP_HID_CTRL4 0x1b +#define RT721_UMP_HID_CTRL5 0x1c +#define RT721_FUNC_FLOAT_CTL0 0x22 +#define RT721_FUNC_FLOAT_CTL1 0x23 +#define RT721_FUNC_FLOAT_CTL2 0x24 +#define RT721_FUNC_FLOAT_CTL3 0x25 +#define RT721_ENT_FLOAT_CTL0 0x29 +#define RT721_ENT_FLOAT_CTL1 0x2c +#define RT721_ENT_FLOAT_CTL2 0x2d +#define RT721_ENT_FLOAT_CTL3 0x2e +#define RT721_ENT_FLOAT_CTL4 0x2f +#define RT721_CH_FLOAT_CTL1 0x45 +#define RT721_CH_FLOAT_CTL2 0x46 +#define RT721_ENT_FLOAT_CTL5 0x53 +#define RT721_ENT_FLOAT_CTL6 0x54 +#define RT721_ENT_FLOAT_CTL7 0x55 +#define RT721_ENT_FLOAT_CTL8 0x57 +#define RT721_ENT_FLOAT_CTL9 0x5a +#define RT721_ENT_FLOAT_CTL10 0x5b +#define RT721_CH_FLOAT_CTL3 0x6a +#define RT721_CH_FLOAT_CTL4 0x6d +#define RT721_CH_FLOAT_CTL5 0x70 +#define RT721_CH_FLOAT_CTL6 0x92 + +/* Parameter & Verb control 01 (0x26)(NID:20h) */ +#define RT721_HIDDEN_REG_SW_RESET (0x1 << 14) + +/* Buffer address for HID */ +#define RT721_BUF_ADDR_HID1 0x44030000 +#define RT721_BUF_ADDR_HID2 0x44030020 + +/* RT721 SDCA Control - function number */ +#define FUNC_NUM_JACK_CODEC 0x01 +#define FUNC_NUM_MIC_ARRAY 0x02 +#define FUNC_NUM_HID 0x03 +#define FUNC_NUM_AMP 0x04 + +/* RT721 SDCA entity */ +#define RT721_SDCA_ENT_HID01 0x01 +#define RT721_SDCA_ENT_XUV 0x03 +#define RT721_SDCA_ENT_GE49 0x49 +#define RT721_SDCA_ENT_USER_FU05 0x05 +#define RT721_SDCA_ENT_USER_FU06 0x06 +#define RT721_SDCA_ENT_USER_FU0F 0x0f +#define RT721_SDCA_ENT_USER_FU10 0x19 +#define RT721_SDCA_ENT_USER_FU1E 0x1e +#define RT721_SDCA_ENT_FU15 0x15 +#define RT721_SDCA_ENT_PDE23 0x23 +#define RT721_SDCA_ENT_PDE40 0x40 +#define RT721_SDCA_ENT_PDE41 0x41 +#define RT721_SDCA_ENT_PDE11 0x11 +#define RT721_SDCA_ENT_PDE12 0x12 +#define RT721_SDCA_ENT_PDE2A 0x2a +#define RT721_SDCA_ENT_CS01 0x01 +#define RT721_SDCA_ENT_CS11 0x11 +#define RT721_SDCA_ENT_CS1F 0x1f +#define RT721_SDCA_ENT_CS1C 0x1c +#define RT721_SDCA_ENT_CS31 0x31 +#define RT721_SDCA_ENT_OT23 0x42 +#define RT721_SDCA_ENT_IT26 0x26 +#define RT721_SDCA_ENT_IT09 0x09 +#define RT721_SDCA_ENT_PLATFORM_FU15 0x15 +#define RT721_SDCA_ENT_PLATFORM_FU44 0x44 +#define RT721_SDCA_ENT_XU03 0x03 +#define RT721_SDCA_ENT_XU0D 0x0d +#define RT721_SDCA_ENT_FU55 0x55 + +/* RT721 SDCA control */ +#define RT721_SDCA_CTL_SAMPLE_FREQ_INDEX 0x10 +#define RT721_SDCA_CTL_FU_MUTE 0x01 +#define RT721_SDCA_CTL_FU_VOLUME 0x02 +#define RT721_SDCA_CTL_HIDTX_CURRENT_OWNER 0x10 +#define RT721_SDCA_CTL_HIDTX_SET_OWNER_TO_DEVICE 0x11 +#define RT721_SDCA_CTL_HIDTX_MESSAGE_OFFSET 0x12 +#define RT721_SDCA_CTL_HIDTX_MESSAGE_LENGTH 0x13 +#define RT721_SDCA_CTL_SELECTED_MODE 0x01 +#define RT721_SDCA_CTL_DETECTED_MODE 0x02 +#define RT721_SDCA_CTL_REQ_POWER_STATE 0x01 +#define RT721_SDCA_CTL_VENDOR_DEF 0x30 +#define RT721_SDCA_CTL_XUV 0x34 +#define RT721_SDCA_CTL_FU_CH_GAIN 0x0b + +/* RT721 SDCA channel */ +#define CH_L 0x01 +#define CH_R 0x02 +#define CH_01 0x01 +#define CH_02 0x02 +#define CH_03 0x03 +#define CH_04 0x04 +#define CH_08 0x08 +#define CH_09 0x09 +#define CH_0A 0x0a + +/* sample frequency index */ +#define RT721_SDCA_RATE_8000HZ 0x01 +#define RT721_SDCA_RATE_11025HZ 0x02 +#define RT721_SDCA_RATE_12000HZ 0x03 +#define RT721_SDCA_RATE_16000HZ 0x04 +#define RT721_SDCA_RATE_22050HZ 0x05 +#define RT721_SDCA_RATE_24000HZ 0x06 +#define RT721_SDCA_RATE_32000HZ 0x07 +#define RT721_SDCA_RATE_44100HZ 0x08 +#define RT721_SDCA_RATE_48000HZ 0x09 +#define RT721_SDCA_RATE_88200HZ 0x0a +#define RT721_SDCA_RATE_96000HZ 0x0b +#define RT721_SDCA_RATE_176400HZ 0x0c +#define RT721_SDCA_RATE_192000HZ 0x0d +#define RT721_SDCA_RATE_384000HZ 0x0e +#define RT721_SDCA_RATE_768000HZ 0x0f + +enum { + RT721_AIF1, /* For headset mic and headphone */ + RT721_AIF2, /* For speaker */ + RT721_AIF3, /* For dmic */ + RT721_AIFS, +}; + +enum rt721_sdca_jd_src { + RT721_JD_NULL, + RT721_JD1, + RT721_JD2, +}; + +int rt721_sdca_io_init(struct device *dev, struct sdw_slave *slave); +int rt721_sdca_init(struct device *dev, struct regmap *regmap, + struct regmap *mbq_regmap, struct sdw_slave *slave); +int rt721_sdca_index_write(struct rt721_sdca_priv *rt721, + unsigned int nid, unsigned int reg, unsigned int value); +int rt721_sdca_index_read(struct rt721_sdca_priv *rt721, + unsigned int nid, unsigned int reg, unsigned int *value); + +int rt721_sdca_jack_detect(struct rt721_sdca_priv *rt721, bool *hp, bool *mic); +#endif /* __RT721_H__ */ -- 2.34.1

4 7

[PATCH] ALSA: line6: add hw monitor volume control to POD HD500X
by Hans P Moller 25 Sep '24

25 Sep '24

Add hw monitor volume control for POD HD500X. This is done adding LINE6_CAP_HWMON_CTL to the capabilities Signed-off-by: Hans P. Moller <hmoller(a)uc.cl> --- sound/usb/line6/podhd.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/podhd.c b/sound/usb/line6/podhd.c index ffd8c157a281..70de08635f54 100644 --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -507,7 +507,7 @@ static const struct line6_properties podhd_properties_table[] = { [LINE6_PODHD500X] = { .id = "PODHD500X", .name = "POD HD500X", - .capabilities = LINE6_CAP_CONTROL + .capabilities = LINE6_CAP_CONTROL | LINE6_CAP_HWMON_CTL | LINE6_CAP_PCM | LINE6_CAP_HWMON, .altsetting = 1, .ep_ctrl_r = 0x81, -- 2.43.0

1 0

[PATCH] Revert "ALSA: memalloc: Workaround for Xen PV"
by Ariadne Conill 25 Sep '24

25 Sep '24

This patch attempted to work around a DMA issue involving Xen, but causes subtle kernel memory corruption. When I brought up this patch in the XenDevel matrix channel, I was told that it had been requested by the Qubes OS developers because they were trying to fix an issue where the sound stack would fail after a few hours of uptime. They wound up disabling SG buffering entirely instead as a workaround. Accordingly, I propose that we should revert this workaround patch, since it causes kernel memory corruption and that the ALSA and Xen communities should collaborate on fixing the underlying problem in such a way that SG buffering works correctly under Xen. This reverts commit 53466ebdec614f915c691809b0861acecb941e30. Signed-off-by: Ariadne Conill <ariadne(a)ariadne.space> Cc: stable(a)vger.kernel.org Cc: xen-devel(a)lists.xenproject.org Cc: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.de> --- sound/core/memalloc.c | 87 +++++++++---------------------------------- 1 file changed, 18 insertions(+), 69 deletions(-) diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index f901504b5afc..81025f50a542 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -541,15 +541,16 @@ static void *snd_dma_noncontig_alloc(struct snd_dma_buffer *dmab, size_t size) struct sg_table *sgt; void *p; -#ifdef CONFIG_SND_DMA_SGBUF - if (cpu_feature_enabled(X86_FEATURE_XENPV)) - return snd_dma_sg_fallback_alloc(dmab, size); -#endif sgt = dma_alloc_noncontiguous(dmab->dev.dev, size, dmab->dev.dir, DEFAULT_GFP, 0); #ifdef CONFIG_SND_DMA_SGBUF - if (!sgt && !get_dma_ops(dmab->dev.dev)) + if (!sgt && !get_dma_ops(dmab->dev.dev)) { + if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) + dmab->dev.type = SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; + else + dmab->dev.type = SNDRV_DMA_TYPE_DEV_SG_FALLBACK; return snd_dma_sg_fallback_alloc(dmab, size); + } #endif if (!sgt) return NULL; @@ -716,38 +717,19 @@ static const struct snd_malloc_ops snd_dma_sg_wc_ops = { /* Fallback SG-buffer allocations for x86 */ struct snd_dma_sg_fallback { - bool use_dma_alloc_coherent; size_t count; struct page **pages; - /* DMA address array; the first page contains #pages in ~PAGE_MASK */ - dma_addr_t *addrs; }; static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab, struct snd_dma_sg_fallback *sgbuf) { - size_t i, size; - - if (sgbuf->pages && sgbuf->addrs) { - i = 0; - while (i < sgbuf->count) { - if (!sgbuf->pages[i] || !sgbuf->addrs[i]) - break; - size = sgbuf->addrs[i] & ~PAGE_MASK; - if (WARN_ON(!size)) - break; - if (sgbuf->use_dma_alloc_coherent) - dma_free_coherent(dmab->dev.dev, size << PAGE_SHIFT, - page_address(sgbuf->pages[i]), - sgbuf->addrs[i] & PAGE_MASK); - else - do_free_pages(page_address(sgbuf->pages[i]), - size << PAGE_SHIFT, false); - i += size; - } - } + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; + size_t i; + + for (i = 0; i < sgbuf->count && sgbuf->pages[i]; i++) + do_free_pages(page_address(sgbuf->pages[i]), PAGE_SIZE, wc); kvfree(sgbuf->pages); - kvfree(sgbuf->addrs); kfree(sgbuf); } @@ -756,36 +738,24 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) struct snd_dma_sg_fallback *sgbuf; struct page **pagep, *curp; size_t chunk, npages; - dma_addr_t *addrp; dma_addr_t addr; void *p; - - /* correct the type */ - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_SG) - dmab->dev.type = SNDRV_DMA_TYPE_DEV_SG_FALLBACK; - else if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG) - dmab->dev.type = SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; + bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK; sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL); if (!sgbuf) return NULL; - sgbuf->use_dma_alloc_coherent = cpu_feature_enabled(X86_FEATURE_XENPV); size = PAGE_ALIGN(size); sgbuf->count = size >> PAGE_SHIFT; sgbuf->pages = kvcalloc(sgbuf->count, sizeof(*sgbuf->pages), GFP_KERNEL); - sgbuf->addrs = kvcalloc(sgbuf->count, sizeof(*sgbuf->addrs), GFP_KERNEL); - if (!sgbuf->pages || !sgbuf->addrs) + if (!sgbuf->pages) goto error; pagep = sgbuf->pages; - addrp = sgbuf->addrs; - chunk = (PAGE_SIZE - 1) << PAGE_SHIFT; /* to fit in low bits in addrs */ + chunk = size; while (size > 0) { chunk = min(size, chunk); - if (sgbuf->use_dma_alloc_coherent) - p = dma_alloc_coherent(dmab->dev.dev, chunk, &addr, DEFAULT_GFP); - else - p = do_alloc_pages(dmab->dev.dev, chunk, &addr, false); + p = do_alloc_pages(dmab->dev.dev, chunk, &addr, wc); if (!p) { if (chunk <= PAGE_SIZE) goto error; @@ -797,25 +767,17 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) size -= chunk; /* fill pages */ npages = chunk >> PAGE_SHIFT; - *addrp = npages; /* store in lower bits */ curp = virt_to_page(p); - while (npages--) { + while (npages--) *pagep++ = curp++; - *addrp++ |= addr; - addr += PAGE_SIZE; - } } p = vmap(sgbuf->pages, sgbuf->count, VM_MAP, PAGE_KERNEL); if (!p) goto error; - - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) - set_pages_array_wc(sgbuf->pages, sgbuf->count); - dmab->private_data = sgbuf; /* store the first page address for convenience */ - dmab->addr = sgbuf->addrs[0] & PAGE_MASK; + dmab->addr = snd_sgbuf_get_addr(dmab, 0); return p; error: @@ -825,23 +787,10 @@ static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size) static void snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab) { - struct snd_dma_sg_fallback *sgbuf = dmab->private_data; - - if (dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK) - set_pages_array_wb(sgbuf->pages, sgbuf->count); vunmap(dmab->area); __snd_dma_sg_fallback_free(dmab, dmab->private_data); } -static dma_addr_t snd_dma_sg_fallback_get_addr(struct snd_dma_buffer *dmab, - size_t offset) -{ - struct snd_dma_sg_fallback *sgbuf = dmab->private_data; - size_t index = offset >> PAGE_SHIFT; - - return (sgbuf->addrs[index] & PAGE_MASK) | (offset & ~PAGE_MASK); -} - static int snd_dma_sg_fallback_mmap(struct snd_dma_buffer *dmab, struct vm_area_struct *area) { @@ -856,8 +805,8 @@ static const struct snd_malloc_ops snd_dma_sg_fallback_ops = { .alloc = snd_dma_sg_fallback_alloc, .free = snd_dma_sg_fallback_free, .mmap = snd_dma_sg_fallback_mmap, - .get_addr = snd_dma_sg_fallback_get_addr, /* reuse vmalloc helpers */ + .get_addr = snd_dma_vmalloc_get_addr, .get_page = snd_dma_vmalloc_get_page, .get_chunk_size = snd_dma_vmalloc_get_chunk_size, }; -- 2.39.2

5 12

[PATCH v27 00/32] Introduce QC USB SND audio offloading support
by Wesley Cheng 25 Sep '24

25 Sep '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v27: - Added some comments and notes about the offload design. Enforcing the q6routing to only allow one USB mixer (PCM device) to be enabled at a time. - Modified SND_JACK_USB notifications for all USB audio offloadable devices plugged in - Rebased on latest XHCI secondary interrupter IMOD changes upstream. Modified the change in this series to allow for XHCI sideband to set the IMOD for sideband clients. - Updated documentation on how USB SND kcontrols are involved in the overall design. - Remove mutex locking from suspend/resume platform ops, as USB core ensures that the interface and device are in the RPM_ACTIVE state while disconnect is handled. Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (30): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: xhci-sideband: Set IMOD for xHCI sideband clients usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 456 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 31 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 41 +- drivers/usb/host/xhci.h | 16 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 70 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 139 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 32 +- sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 342 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 106 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1968 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5864 insertions(+), 55 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

2 38

[PATCH] ASoC: atmel: mchp-pdmc: Skip ALSA restoration if substream runtime is uninitialized
by Andrei Simion 24 Sep '24

24 Sep '24

Update the driver to prevent alsa-restore.service from failing when reading data from /var/lib/alsa/asound.state at boot. Ensure that the restoration of ALSA mixer configurations is skipped if substream->runtime is NULL. Fixes: 50291652af52 ("ASoC: atmel: mchp-pdmc: add PDMC driver") Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> --- sound/soc/atmel/mchp-pdmc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index 939cd44ebc8a..06dc3c48e7e8 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -302,6 +302,9 @@ static int mchp_pdmc_chmap_ctl_put(struct snd_kcontrol *kcontrol, if (!substream) return -ENODEV; + if (!substream->runtime) + return 0; /* just for avoiding error from alsactl restore */ + map = mchp_pdmc_chmap_get(substream, info); if (!map) return -EINVAL; base-commit: 4d0326b60bb753627437fff0f76bf1525bcda422 -- 2.34.1

2 4

[PATCH] ASoC: rt721-sdca: Add RT721 SDCA driver
by Jack Yu 24 Sep '24

24 Sep '24

This is the initial codec driver for rt721-sdca. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt721-sdca-sdw.c | 550 +++++++++ sound/soc/codecs/rt721-sdca-sdw.h | 151 +++ sound/soc/codecs/rt721-sdca.c | 1790 +++++++++++++++++++++++++++++ sound/soc/codecs/rt721-sdca.h | 277 +++++ 6 files changed, 2777 insertions(+) create mode 100644 sound/soc/codecs/rt721-sdca-sdw.c create mode 100644 sound/soc/codecs/rt721-sdca-sdw.h create mode 100644 sound/soc/codecs/rt721-sdca.c create mode 100644 sound/soc/codecs/rt721-sdca.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7092842480ef..74c14da4ae09 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -222,6 +222,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT712_SDCA_DMIC_SDW imply SND_SOC_RT715_SDW imply SND_SOC_RT715_SDCA_SDW + imply SND_SOC_RT721_SDCA_SDW imply SND_SOC_RT722_SDCA_SDW imply SND_SOC_RT1308_SDW imply SND_SOC_RT1316_SDW @@ -1743,6 +1744,12 @@ config SND_SOC_RT712_SDCA_DMIC_SDW select REGMAP_SOUNDWIRE select REGMAP_SOUNDWIRE_MBQ +config SND_SOC_RT721_SDCA_SDW + tristate "Realtek RT721 SDCA Codec - SDW" + depends on SOUNDWIRE + select REGMAP_SOUNDWIRE + select REGMAP_SOUNDWIRE_MBQ + config SND_SOC_RT722_SDCA_SDW tristate "Realtek RT722 SDCA Codec - SDW" depends on SOUNDWIRE diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 54cbc3feae32..3477c6352acc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -259,6 +259,7 @@ snd-soc-rt712-sdca-y := rt712-sdca.o rt712-sdca-sdw.o snd-soc-rt712-sdca-dmic-y := rt712-sdca-dmic.o snd-soc-rt715-y := rt715.o rt715-sdw.o snd-soc-rt715-sdca-y := rt715-sdca.o rt715-sdca-sdw.o +snd-soc-rt721-sdca-y := rt721-sdca.o rt721-sdca-sdw.o snd-soc-rt722-sdca-y := rt722-sdca.o rt722-sdca-sdw.o snd-soc-rt9120-y := rt9120.o snd-soc-rtq9128-y := rtq9128.o @@ -662,6 +663,7 @@ obj-$(CONFIG_SND_SOC_RT712_SDCA_SDW) += snd-soc-rt712-sdca.o obj-$(CONFIG_SND_SOC_RT712_SDCA_DMIC_SDW) += snd-soc-rt712-sdca-dmic.o obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o obj-$(CONFIG_SND_SOC_RT715_SDCA_SDW) += snd-soc-rt715-sdca.o +obj-$(CONFIG_SND_SOC_RT721_SDCA_SDW) += snd-soc-rt721-sdca.o obj-$(CONFIG_SND_SOC_RT722_SDCA_SDW) += snd-soc-rt722-sdca.o obj-$(CONFIG_SND_SOC_RT9120) += snd-soc-rt9120.o obj-$(CONFIG_SND_SOC_RTQ9128) += snd-soc-rtq9128.o diff --git a/sound/soc/codecs/rt721-sdca-sdw.c b/sound/soc/codecs/rt721-sdca-sdw.c new file mode 100644 index 000000000000..3a8dbbc01152 --- /dev/null +++ b/sound/soc/codecs/rt721-sdca-sdw.c @@ -0,0 +1,550 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt721-sdca-sdw.c -- rt721 SDCA ALSA SoC audio driver +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// +// + +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/module.h> +#include <linux/mod_devicetable.h> +#include <linux/pm_runtime.h> +#include <linux/soundwire/sdw_registers.h> + +#include "rt721-sdca.h" +#include "rt721-sdca-sdw.h" + +static bool rt721_sdca_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f01 ... 0x2f0a: + case 0x2f35: + case 0x2f50: + case 0x2f51: + case 0x2f58 ... 0x2f5d: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XUV, + RT721_SDCA_CTL_XUV, 0): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_SELECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_CURRENT_OWNER, 0) ... SDW_SDCA_CTL(FUNC_NUM_HID, + RT721_SDCA_ENT_HID01, RT721_SDCA_CTL_HIDTX_MESSAGE_LENGTH, 0): + case RT721_BUF_ADDR_HID1 ... RT721_BUF_ADDR_HID2: + return true; + default: + return false; + } +} + +static bool rt721_sdca_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x2f01: + case 0x2f51: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XUV, + RT721_SDCA_CTL_XUV, 0): + case SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_CURRENT_OWNER, 0) ... SDW_SDCA_CTL(FUNC_NUM_HID, + RT721_SDCA_ENT_HID01, RT721_SDCA_CTL_HIDTX_MESSAGE_LENGTH, 0): + case RT721_BUF_ADDR_HID1 ... RT721_BUF_ADDR_HID2: + return true; + default: + return false; + } +} + +static bool rt721_sdca_mbq_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0900007: + case 0x0a00005: + case 0x0c00005: + case 0x0d00014: + case 0x0310100: + case 0x2000001: + case 0x2000002: + case 0x2000003: + case 0x2000013: + case 0x200003c: + case 0x2000046: + case 0x5810000: + case 0x5810036: + case 0x5810037: + case 0x5810038: + case 0x5810039: + case 0x5b10018: + case 0x5b10019: + case 0x5f00045: + case 0x5f00048: + case 0x6100000: + case 0x6100005: + case 0x6100006: + case 0x610000d: + case 0x6100010: + case 0x6100011: + case 0x6100013: + case 0x6100015: + case 0x6100017: + case 0x6100025: + case 0x6100029: + case 0x610002c ... 0x610002f: + case 0x6100053 ... 0x6100055: + case 0x6100057: + case 0x610005a: + case 0x610005b: + case 0x610006a: + case 0x610006d: + case 0x6100092: + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_L): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_R): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_L): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_R): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_R): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_01): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_02): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_03): + case SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_04): + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_L): + case SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_R): + return true; + default: + return false; + } +} + +static bool rt721_sdca_mbq_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x0310100: + case 0x0a00005: + case 0x0c00005: + case 0x0d00014: + case 0x2000000: + case 0x200000d: + case 0x2000019: + case 0x2000020: + case 0x2000030: + case 0x2000046: + case 0x2000067: + case 0x2000084: + case 0x2000086: + case 0x5810000: + case 0x5810036: + case 0x5810037: + case 0x5810038: + case 0x5810039: + case 0x5b10018: + case 0x5b10019: + return true; + default: + return false; + } +} + +static const struct regmap_config rt721_sdca_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = rt721_sdca_readable_register, + .volatile_reg = rt721_sdca_volatile_register, + .max_register = 0x44ffffff, + .reg_defaults = rt721_sdca_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(rt721_sdca_reg_defaults), + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct regmap_config rt721_sdca_mbq_regmap = { + .name = "sdw-mbq", + .reg_bits = 32, + .val_bits = 16, + .readable_reg = rt721_sdca_mbq_readable_register, + .volatile_reg = rt721_sdca_mbq_volatile_register, + .max_register = 0x41000312, + .reg_defaults = rt721_sdca_mbq_defaults, + .num_reg_defaults = ARRAY_SIZE(rt721_sdca_mbq_defaults), + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt721_sdca_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(&slave->dev); + + if (status == SDW_SLAVE_UNATTACHED) + rt721->hw_init = false; + + if (status == SDW_SLAVE_ATTACHED) { + if (rt721->hs_jack) { + /* + * Due to the SCP_SDCA_INTMASK will be cleared by any reset, and then + * if the device attached again, we will need to set the setting back. + * It could avoid losing the jack detection interrupt. + * This also could sync with the cache value as the rt721_sdca_jack_init set. + */ + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, + SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + } + } + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt721->hw_init || status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt721_sdca_io_init(&slave->dev, slave); +} + +static int rt721_sdca_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval; + int i, j; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + sdw_slave_read_prop(slave); + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* + * port = 1 for headphone playback + * port = 2 for headset-mic capture + * port = 3 for speaker playback + * port = 6 for digital-mic capture + */ + prop->source_ports = BIT(6) | BIT(2); /* BITMAP: 01000100 */ + prop->sink_ports = BIT(3) | BIT(1); /* BITMAP: 00001010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + j = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[j].num = bit; + dpn[j].type = SDW_DPN_FULL; + dpn[j].simple_ch_prep_sm = true; + dpn[j].ch_prep_timeout = 10; + j++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 200; + + /* wake-up event */ + prop->wake_capable = 1; + + /* Three data lanes are supported by rt721-sdca codec */ + prop->lane_control_support = true; + + return 0; +} + +static int rt721_sdca_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(&slave->dev); + int ret, stat; + int count = 0, retry = 3; + unsigned int sdca_cascade, scp_sdca_stat1, scp_sdca_stat2 = 0; + + if (cancel_delayed_work_sync(&rt721->jack_detect_work)) { + dev_warn(&slave->dev, "%s the pending delayed_work was cancelled", __func__); + /* avoid the HID owner doesn't change to device */ + if (rt721->scp_sdca_stat2) + scp_sdca_stat2 = rt721->scp_sdca_stat2; + } + + /* + * The critical section below intentionally protects a rather large piece of code. + * We don't want to allow the system suspend to disable an interrupt while we are + * processing it, which could be problematic given the quirky SoundWire interrupt + * scheme. We do want however to prevent new workqueues from being scheduled if + * the disable_irq flag was set during system suspend. + */ + mutex_lock(&rt721->disable_irq_lock); + + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + + rt721->scp_sdca_stat1 = ret; + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + + rt721->scp_sdca_stat2 = ret; + if (scp_sdca_stat2) + rt721->scp_sdca_stat2 |= scp_sdca_stat2; + do { + /* clear flag */ + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + if (ret & SDW_SCP_SDCA_INTMASK_SDCA_0) { + ret = sdw_update_no_pm(rt721->slave, SDW_SCP_SDCA_INT1, + SDW_SCP_SDCA_INT_SDCA_0, SDW_SCP_SDCA_INT_SDCA_0); + if (ret < 0) + goto io_error; + } else if (ret & SDW_SCP_SDCA_INTMASK_SDCA_6) { + ret = sdw_update_no_pm(rt721->slave, SDW_SCP_SDCA_INT1, + SDW_SCP_SDCA_INT_SDCA_6, SDW_SCP_SDCA_INT_SDCA_6); + if (ret < 0) + goto io_error; + } + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + if (ret & SDW_SCP_SDCA_INTMASK_SDCA_8) { + ret = sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INT2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + if (ret < 0) + goto io_error; + } + + /* check if flag clear or not */ + ret = sdw_read_no_pm(rt721->slave, SDW_DP0_INT); + if (ret < 0) + goto io_error; + sdca_cascade = ret & SDW_DP0_SDCA_CASCADE; + + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + if (ret < 0) + goto io_error; + scp_sdca_stat1 = ret & SDW_SCP_SDCA_INTMASK_SDCA_0; + + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + if (ret < 0) + goto io_error; + scp_sdca_stat2 = ret & SDW_SCP_SDCA_INTMASK_SDCA_8; + + stat = scp_sdca_stat1 || scp_sdca_stat2 || sdca_cascade; + + count++; + } while (stat != 0 && count < retry); + + if (stat) + dev_warn(&slave->dev, + "%s scp_sdca_stat1=0x%x, scp_sdca_stat2=0x%x\n", __func__, + rt721->scp_sdca_stat1, rt721->scp_sdca_stat2); + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT1); + ret = sdw_read_no_pm(rt721->slave, SDW_SCP_SDCA_INT2); + + if (status->sdca_cascade && !rt721->disable_irq) + mod_delayed_work(system_power_efficient_wq, + &rt721->jack_detect_work, msecs_to_jiffies(280)); + + mutex_unlock(&rt721->disable_irq_lock); + + return 0; + +io_error: + mutex_unlock(&rt721->disable_irq_lock); + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +} + +static const struct sdw_slave_ops rt721_sdca_slave_ops = { + .read_prop = rt721_sdca_read_prop, + .interrupt_callback = rt721_sdca_interrupt_callback, + .update_status = rt721_sdca_update_status, +}; + +static int rt721_sdca_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap, *mbq_regmap; + + /* Regmap Initialization */ + mbq_regmap = devm_regmap_init_sdw_mbq(slave, &rt721_sdca_mbq_regmap); + if (IS_ERR(mbq_regmap)) + return PTR_ERR(mbq_regmap); + + regmap = devm_regmap_init_sdw(slave, &rt721_sdca_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return rt721_sdca_init(&slave->dev, regmap, mbq_regmap, slave); +} + +static int rt721_sdca_sdw_remove(struct sdw_slave *slave) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(&slave->dev); + + if (rt721->hw_init) { + cancel_delayed_work_sync(&rt721->jack_detect_work); + cancel_delayed_work_sync(&rt721->jack_btn_check_work); + } + + if (rt721->first_hw_init) + pm_runtime_disable(&slave->dev); + + mutex_destroy(&rt721->calibrate_mutex); + mutex_destroy(&rt721->disable_irq_lock); + + return 0; +} + +static const struct sdw_device_id rt721_sdca_id[] = { + SDW_SLAVE_ENTRY_EXT(0x025d, 0x721, 0x3, 0x1, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt721_sdca_id); + +static int __maybe_unused rt721_sdca_dev_suspend(struct device *dev) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev); + + if (!rt721->hw_init) + return 0; + + cancel_delayed_work_sync(&rt721->jack_detect_work); + cancel_delayed_work_sync(&rt721->jack_btn_check_work); + + regcache_cache_only(rt721->regmap, true); + regcache_cache_only(rt721->mbq_regmap, true); + + return 0; +} + +static int __maybe_unused rt721_sdca_dev_system_suspend(struct device *dev) +{ + struct rt721_sdca_priv *rt721_sdca = dev_get_drvdata(dev); + struct sdw_slave *slave = dev_to_sdw_dev(dev); + int ret1, ret2; + + if (!rt721_sdca->hw_init) + return 0; + + /* + * prevent new interrupts from being handled after the + * deferred work completes and before the parent disables + * interrupts on the link + */ + mutex_lock(&rt721_sdca->disable_irq_lock); + rt721_sdca->disable_irq = true; + ret1 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK1, + SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6, 0); + ret2 = sdw_update_no_pm(slave, SDW_SCP_SDCA_INTMASK2, + SDW_SCP_SDCA_INTMASK_SDCA_8, 0); + mutex_unlock(&rt721_sdca->disable_irq_lock); + + if (ret1 < 0 || ret2 < 0) { + /* log but don't prevent suspend from happening */ + dev_dbg(&slave->dev, "%s: could not disable SDCA interrupts\n:", __func__); + } + + return rt721_sdca_dev_suspend(dev); +} + +#define RT721_PROBE_TIMEOUT 5000 + +static int __maybe_unused rt721_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt721->first_hw_init) + return 0; + + if (!slave->unattach_request) { + mutex_lock(&rt721->disable_irq_lock); + if (rt721->disable_irq == true) { + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK1, SDW_SCP_SDCA_INTMASK_SDCA_6); + sdw_write_no_pm(slave, SDW_SCP_SDCA_INTMASK2, SDW_SCP_SDCA_INTMASK_SDCA_8); + rt721->disable_irq = false; + } + mutex_unlock(&rt721->disable_irq_lock); + goto regmap_sync; + } + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT721_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt721->regmap, false); + regcache_sync(rt721->regmap); + regcache_cache_only(rt721->mbq_regmap, false); + regcache_sync(rt721->mbq_regmap); + return 0; +} + +static const struct dev_pm_ops rt721_sdca_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt721_sdca_dev_system_suspend, rt721_sdca_dev_resume) + SET_RUNTIME_PM_OPS(rt721_sdca_dev_suspend, rt721_sdca_dev_resume, NULL) +}; + +static struct sdw_driver rt721_sdca_sdw_driver = { + .driver = { + .name = "rt721-sdca", + .owner = THIS_MODULE, + .pm = &rt721_sdca_pm, + }, + .probe = rt721_sdca_sdw_probe, + .remove = rt721_sdca_sdw_remove, + .ops = &rt721_sdca_slave_ops, + .id_table = rt721_sdca_id, +}; +module_sdw_driver(rt721_sdca_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT721 SDCA SDW driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt721-sdca-sdw.h b/sound/soc/codecs/rt721-sdca-sdw.h new file mode 100644 index 000000000000..d96f30c9d290 --- /dev/null +++ b/sound/soc/codecs/rt721-sdca-sdw.h @@ -0,0 +1,151 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt721-sdca-sdw.h -- RT721 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2024 Realtek Semiconductor Corp. + */ + +#ifndef __RT721_SDW_H__ +#define __RT721_SDW_H__ + +#include <linux/regmap.h> +#include <linux/soundwire/sdw_registers.h> + +static const struct reg_default rt721_sdca_reg_defaults[] = { + { 0x202d, 0x00 }, + { 0x2f01, 0x00 }, + { 0x2f02, 0x09 }, + { 0x2f03, 0x08 }, + { 0x2f04, 0x00 }, + { 0x2f05, 0x0e }, + { 0x2f06, 0x01 }, + { 0x2f09, 0x00 }, + { 0x2f0a, 0x00 }, + { 0x2f35, 0x00 }, + { 0x2f50, 0xf0 }, + { 0x2f58, 0x07 }, + { 0x2f59, 0x07 }, + { 0x2f5a, 0x00 }, + { 0x2f5b, 0x07 }, + { 0x2f5c, 0x27 }, + { 0x2f5d, 0x07 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS01, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS11, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE12, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE40, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_03), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_04), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_CS1F, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_IT26, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_PDE2A, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_CS31, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), 0x09 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_L), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_R), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_OT23, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x00 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x01 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x01 }, +}; + +static const struct reg_default rt721_sdca_mbq_defaults[] = { + { 0x0900007, 0xc004 }, + { 0x2000001, 0x0000 }, + { 0x2000002, 0x0000 }, + { 0x2000003, 0x0000 }, + { 0x2000013, 0x8001 }, + { 0x200003c, 0x0000 }, + { 0x2000046, 0x3400 }, + // { 0x5810000, 0x702d }, + { 0x5f00044, 0x6040 }, + { 0x5f00045, 0x3333 }, + { 0x5f00048, 0x0000 }, + { 0x6100005, 0x0005 }, + { 0x6100006, 0x0000 }, + { 0x610000d, 0x0051 }, + { 0x6100010, 0x0180 }, + { 0x6100011, 0x0000 }, + { 0x6100013, 0x0000 }, + { 0x6100015, 0x0000 }, + { 0x6100017, 0x8049 }, + { 0x6100025, 0x1000 }, + { 0x6100029, 0x0809 }, + { 0x610002c, 0x2828 }, + { 0x610002d, 0x2929 }, + { 0x610002e, 0x3529 }, + { 0x610002f, 0x2901 }, + { 0x6100053, 0x2630 }, + { 0x6100054, 0x2a2a }, + { 0x6100055, 0x152f }, + { 0x6100057, 0x2200 }, + { 0x610005a, 0x2a4b }, + { 0x610005b, 0x2a00 }, + { 0x610006a, 0x0102 }, + { 0x610006d, 0x0102 }, + { 0x6100092, 0x4f61 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_L), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, RT721_SDCA_CTL_FU_VOLUME, + CH_R), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_L), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, RT721_SDCA_CTL_FU_VOLUME, + CH_R), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, RT721_SDCA_CTL_FU_CH_GAIN, + CH_L), 0xfe00 }, + { SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, RT721_SDCA_CTL_FU_CH_GAIN, + CH_R), 0xfe00 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_01), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_02), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_03), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, RT721_SDCA_CTL_FU_CH_GAIN, CH_04), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_01), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_02), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_03), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, RT721_SDCA_CTL_FU_VOLUME, + CH_04), 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_L), + 0x0000 }, + { SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, RT721_SDCA_CTL_FU_VOLUME, CH_R), + 0x0000 }, +}; + +#endif /* __RT721_SDW_H__ */ diff --git a/sound/soc/codecs/rt721-sdca.c b/sound/soc/codecs/rt721-sdca.c new file mode 100644 index 000000000000..cca749ebbd31 --- /dev/null +++ b/sound/soc/codecs/rt721-sdca.c @@ -0,0 +1,1790 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt721-sdca.c -- rt721 SDCA ALSA SoC audio driver +// +// Copyright(c) 2024 Realtek Semiconductor Corp. +// +// + +#include <linux/bitops.h> +#include <sound/core.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <sound/initval.h> +#include <sound/jack.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <sound/pcm.h> +#include <linux/pm_runtime.h> +#include <sound/pcm_params.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/slab.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> + +#include "rt721-sdca.h" + +static void rt721_sdca_dmic_preset(struct rt721_sdca_priv *rt721) +{ + /* Power down group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8000); + /* VREF_HV_EN_AUTO_FAST */ + rt721_sdca_index_write(rt721, RT721_ANA_POW_PART, + RT721_VREF1_HV_CTRL1, 0xe000); + /* Power up group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8007); + /* Set AD07/08 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL9, 0x2a2a); + /* Set AD10 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL10, 0x2a00); + /* Set DMIC1/DMIC2 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL6, 0x2a2a); + /* Set DMIC1/DMIC2 IT entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL5, 0x2626); + /* Set AD10 FU entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL8, 0x1e00); + /* Set DMIC1/DMIC2 FU input gain floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL7, 0x1515); + /* Set DMIC2 FU input gain channel floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_CH_FLOAT_CTL3, 0x0304); + /* Set AD10 FU channel floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_CH_FLOAT_CTL4, 0x0304); + /* vf71f_r12_07_06 and vf71f_r13_07_06 = 2’b00 */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_CTL1, 0x0000); + /* Enable vf707_r12_05/vf707_r13_05 */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_IT26, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x01); + /* Set usd_flag_sel, usd_in_sel */ + regmap_write(rt721->mbq_regmap, 0x5910009, 0x2e01); + /* Set RC calibration */ + rt721_sdca_index_write(rt721, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b00); + rt721_sdca_index_write(rt721, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b40); + /* Fine tune PDE2A latency */ + regmap_write(rt721->regmap, 0x2f5c, 0x25); +} + +static void rt721_sdca_amp_preset(struct rt721_sdca_priv *rt721) +{; + /* Power down group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8000); + /* VREF_HV_EN_AUTO_FAST */ + rt721_sdca_index_write(rt721, RT721_ANA_POW_PART, + RT721_VREF1_HV_CTRL1, 0xe000); + /* Power up group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8007); + /* Reset dc_cal_top */ + regmap_write(rt721->mbq_regmap, 0x5810000, 0x6420); + /* Turn back to normal dc_cal_top */ + regmap_write(rt721->mbq_regmap, 0x5810000, 0x6421); + /* W1C Trigger Calibration */ + regmap_write(rt721->mbq_regmap, 0x5810000, 0xe421); + /* DAC04 FU entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_CH_FLOAT_CTL6, 0x5561); + /* Set EAPD high */ + rt721_sdca_index_write(rt721, RT721_VENDOR_REG, + RT721_GPIO_PAD_CTRL5, 0x8003); + /* Enable vf707_r14 */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_OT23, + RT721_SDCA_CTL_VENDOR_DEF, 0), 0x04); + /* FU 23 SPK mute control - L */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x00); + /* FU 23 SPK mute control - R */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x00); + /* FU 55 DAC04 mute control - L */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_01), 0x00); + /* FU 55 DAC04 mute control - R */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_FU55, + RT721_SDCA_CTL_FU_MUTE, CH_02), 0x00); +} + +static void rt721_sdca_jack_preset(struct rt721_sdca_priv *rt721) +{ + /* Power down group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8000); + /* VREF_HV_EN_AUTO_FAST */ + rt721_sdca_index_write(rt721, RT721_ANA_POW_PART, + RT721_VREF1_HV_CTRL1, 0xe000); + /* Power up group1/2/3_mic_pdb */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_MISC_POWER_CTL31, 0x8007); + /* GE0 mode related control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_GE_REL_CTRL1, 0x8011); + /* Button A, B, C, D bypass mode */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL3, 0xcf00); + /* HID1 slot enable */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL4, 0x000f); + /* Report ID for HID1 */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL1, 0x1100); + /* OSC/OOC for slot 2, 3 */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_UMP_HID_CTRL5, 0x0c12); + /* Set JD de-bounce clock control */ + rt721_sdca_index_write(rt721, RT721_JD_CTRL, + RT721_JD_1PIN_GAT_CTRL2, 0xc002); + /* RC calibration -1 */ + rt721_sdca_index_write(rt721, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b00); + /* RC calibration -2 */ + rt721_sdca_index_write(rt721, RT721_RC_CALIB_CTRL, + RT721_RC_CALIB_CTRL0, 0x0b40); + /* pow_clk_12p288mhz_dre03 change to register mode */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON14, 0x3333); + /* Tune calibration timing control */ + regmap_write(rt721->mbq_regmap, 0x5810035, 0x0036); + /* calibration HP amp output select control from Efuse */ + regmap_write(rt721->mbq_regmap, 0x5810030, 0xee00); + /* FSM related control */ + rt721_sdca_index_write(rt721, RT721_CAP_PORT_CTRL, + RT721_HP_AMP_2CH_CAL1, 0x0140); + /* HP calibration related control */ + regmap_write(rt721->mbq_regmap, 0x5810000, 0x0021); + /* W1C HP calibration*/ + regmap_write(rt721->mbq_regmap, 0x5810000, 0x8021); + /* reg_sel_cin_hp_0010/0011 */ + rt721_sdca_index_write(rt721, RT721_CAP_PORT_CTRL, + RT721_HP_AMP_2CH_CAL18, 0x5522); + regmap_write(rt721->mbq_regmap, 0x5b10007, 0x2000); + /* sel_sensing_lr_hp */ + regmap_write(rt721->mbq_regmap, 0x5B10017, 0x1b0f); + /* Release HP-JD */ + rt721_sdca_index_write(rt721, RT721_CBJ_CTRL, + RT721_CBJ_A0_GAT_CTRL1, 0x2a02); + /* en_osw gating auto done bit */ + rt721_sdca_index_write(rt721, RT721_CAP_PORT_CTRL, + RT721_HP_AMP_2CH_CAL4, 0xa105); + /* pow_clk_en_sw_amp_detect_sel to register mode */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON14, 0x3b33); + /* cp_sw_hp to auto mode */ + regmap_write(rt721->mbq_regmap, 0x310400, 0x3023); + /* pow_clk_en_sw_amp_detect power up */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON14, 0x3f33); + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON13, 0x6048); + /* switch size detect threshold */ + regmap_write(rt721->mbq_regmap, 0x310401, 0x3000); + regmap_write(rt721->mbq_regmap, 0x310402, 0x1b00); + /* en_hp_amp_detect auto mode */ + regmap_write(rt721->mbq_regmap, 0x310300, 0x000f); + /* amp detect threshold */ + regmap_write(rt721->mbq_regmap, 0x310301, 0x3000); + regmap_write(rt721->mbq_regmap, 0x310302, 0x1b00); + /* gating_sdw_link_rst_n_1_cbj_reg */ + rt721_sdca_index_write(rt721, RT721_VENDOR_ANA_CTL, + RT721_UAJ_TOP_TCON17, 0x0008); + /* CKXEN_SDAC chopper function */ + rt721_sdca_index_write(rt721, RT721_DAC_CTRL, + RT721_DAC_2CH_CTRL3, 0x55ff); + /* CKXSEL_SDAC chopper frequency */ + rt721_sdca_index_write(rt721, RT721_DAC_CTRL, + RT721_DAC_2CH_CTRL4, 0xcc00); + /* Bias current for SDAC */ + rt721_sdca_index_write(rt721, RT721_ANA_POW_PART, + RT721_MBIAS_LV_CTRL2, 0x6677); + /* VREF2 level selection */ + rt721_sdca_index_write(rt721, RT721_ANA_POW_PART, + RT721_VREF2_LV_CTRL1, 0x7600); + /* ADC09/MIC2 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL2, 0x1234); + /* LINE2 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL3, 0x3512); + /* DAC03/HP power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL1, 0x4040); + /* ADC27 power entity floating control */ + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_ENT_FLOAT_CTL4, 0x1201); + /* Fine tune PDE40 latency */ + regmap_write(rt721->regmap, 0x2f58, 0x07); +} + +int rt721_sdca_index_write(struct rt721_sdca_priv *rt721, + unsigned int nid, unsigned int reg, unsigned int value) +{ + struct regmap *regmap = rt721->mbq_regmap; + unsigned int addr = (nid << 20) | reg; + int ret; + + ret = regmap_write(regmap, addr, value); + if (ret < 0) + dev_err(&rt721->slave->dev, + "Failed to set private value: %06x <= %04x ret=%d\n", + addr, value, ret); + + return ret; +} + +int rt721_sdca_index_read(struct rt721_sdca_priv *rt721, + unsigned int nid, unsigned int reg, unsigned int *value) +{ + int ret; + struct regmap *regmap = rt721->mbq_regmap; + unsigned int addr = (nid << 20) | reg; + + ret = regmap_read(regmap, addr, value); + if (ret < 0) + dev_err(&rt721->slave->dev, + "Failed to get private value: %06x => %04x ret=%d\n", + addr, *value, ret); + + return ret; +} + +static int rt721_sdca_index_update_bits(struct rt721_sdca_priv *rt721, + unsigned int nid, unsigned int reg, unsigned int mask, unsigned int val) +{ + unsigned int tmp; + int ret; + + ret = rt721_sdca_index_read(rt721, nid, reg, &tmp); + if (ret < 0) + return ret; + + set_mask_bits(&tmp, mask, val); + return rt721_sdca_index_write(rt721, nid, reg, tmp); +} + +static int rt721_sdca_btn_type(unsigned char *buffer) +{ + if ((*buffer & 0xf0) == 0x10 || (*buffer & 0x0f) == 0x01 || + (*(buffer + 1) == 0x01) || (*(buffer + 1) == 0x10)) + return SND_JACK_BTN_2; + else if ((*buffer & 0xf0) == 0x20 || (*buffer & 0x0f) == 0x02 || + (*(buffer + 1) == 0x02) || (*(buffer + 1) == 0x20)) + return SND_JACK_BTN_3; + else if ((*buffer & 0xf0) == 0x40 || (*buffer & 0x0f) == 0x04 || + (*(buffer + 1) == 0x04) || (*(buffer + 1) == 0x40)) + return SND_JACK_BTN_0; + else if ((*buffer & 0xf0) == 0x80 || (*buffer & 0x0f) == 0x08 || + (*(buffer + 1) == 0x08) || (*(buffer + 1) == 0x80)) + return SND_JACK_BTN_1; + + return 0; +} + +static unsigned int rt721_sdca_button_detect(struct rt721_sdca_priv *rt721) +{ + unsigned int btn_type = 0, offset, idx, val, owner; + int ret; + unsigned char buf[3]; + + /* get current UMP message owner */ + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), &owner); + if (ret < 0) + return 0; + + /* if owner is device then there is no button event from device */ + if (owner == 1) + return 0; + + /* read UMP message offset */ + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), &offset); + if (ret < 0) + goto _end_btn_det_; + + for (idx = 0; idx < sizeof(buf); idx++) { + ret = regmap_read(rt721->regmap, + RT721_BUF_ADDR_HID1 + offset + idx, &val); + if (ret < 0) + goto _end_btn_det_; + buf[idx] = val & 0xff; + } + + if (buf[0] == 0x11) + btn_type = rt721_sdca_btn_type(&buf[1]); + +_end_btn_det_: + /* Host is owner, so set back to device */ + if (owner == 0) + /* set owner to device */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_CURRENT_OWNER, 0), 0x01); + + return btn_type; +} + +static int rt721_sdca_headset_detect(struct rt721_sdca_priv *rt721) +{ + unsigned int det_mode; + int ret; + + /* get detected_mode */ + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0), &det_mode); + + if (ret < 0) + goto io_error; + + switch (det_mode) { + case 0x00: + rt721->jack_type = 0; + break; + case 0x03: + rt721->jack_type = SND_JACK_HEADPHONE; + break; + case 0x05: + rt721->jack_type = SND_JACK_HEADSET; + break; + } + + /* write selected_mode */ + if (det_mode) { + ret = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_SELECTED_MODE, 0), det_mode); + if (ret < 0) + goto io_error; + } + + dev_dbg(&rt721->slave->dev, + "%s, detected_mode=0x%x\n", __func__, det_mode); + + return 0; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); + return ret; +} + +static void rt721_sdca_jack_detect_handler(struct work_struct *work) +{ + struct rt721_sdca_priv *rt721 = + container_of(work, struct rt721_sdca_priv, jack_detect_work.work); + int btn_type = 0, ret; + + if (!rt721->hs_jack) + return; + + if (!rt721->component->card || !rt721->component->card->instantiated) + return; + + /* SDW_SCP_SDCA_INT_SDCA_6 is used for jack detection */ + if (rt721->scp_sdca_stat1 & SDW_SCP_SDCA_INT_SDCA_6) { + ret = rt721_sdca_headset_detect(rt721); + if (ret < 0) + return; + } + + /* SDW_SCP_SDCA_INT_SDCA_8 is used for button detection */ + if (rt721->scp_sdca_stat2 & SDW_SCP_SDCA_INT_SDCA_8) + btn_type = rt721_sdca_button_detect(rt721); + + if (rt721->jack_type == 0) + btn_type = 0; + + dev_dbg(&rt721->slave->dev, + "in %s, jack_type=%d\n", __func__, rt721->jack_type); + dev_dbg(&rt721->slave->dev, + "in %s, btn_type=0x%x\n", __func__, btn_type); + pr_info("rtk debug, jack_type=%d\n", rt721->jack_type); + pr_info("rtk debug, btn_type=0x%x\n", btn_type); + dev_dbg(&rt721->slave->dev, + "in %s, scp_sdca_stat1=0x%x, scp_sdca_stat2=0x%x\n", __func__, + rt721->scp_sdca_stat1, rt721->scp_sdca_stat2); + + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt721->jack_btn_check_work, msecs_to_jiffies(200)); + } +} + +static void rt721_sdca_btn_check_handler(struct work_struct *work) +{ + struct rt721_sdca_priv *rt721 = + container_of(work, struct rt721_sdca_priv, jack_btn_check_work.work); + int btn_type = 0, ret, idx; + unsigned int det_mode, offset, val; + unsigned char buf[3]; + + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_GE49, + RT721_SDCA_CTL_DETECTED_MODE, 0), &det_mode); + if (ret < 0) + goto io_error; + + /* pin attached */ + if (det_mode) { + /* read UMP message offset */ + ret = regmap_read(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_HID, RT721_SDCA_ENT_HID01, + RT721_SDCA_CTL_HIDTX_MESSAGE_OFFSET, 0), &offset); + if (ret < 0) + goto io_error; + + for (idx = 0; idx < sizeof(buf); idx++) { + ret = regmap_read(rt721->regmap, + RT721_BUF_ADDR_HID1 + offset + idx, &val); + if (ret < 0) + goto io_error; + buf[idx] = val & 0xff; + } + + if (buf[0] == 0x11) + btn_type = rt721_sdca_btn_type(&buf[1]); + } else + rt721->jack_type = 0; + + dev_dbg(&rt721->slave->dev, "%s, btn_type=0x%x\n", __func__, btn_type); + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type | btn_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (btn_type) { + /* button released */ + snd_soc_jack_report(rt721->hs_jack, rt721->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + mod_delayed_work(system_power_efficient_wq, + &rt721->jack_btn_check_work, msecs_to_jiffies(200)); + } + + return; + +io_error: + pr_err_ratelimited("IO error in %s, ret %d\n", __func__, ret); +} + +static void rt721_sdca_jack_init(struct rt721_sdca_priv *rt721) +{ + mutex_lock(&rt721->calibrate_mutex); + if (rt721->hs_jack) { + /* set SCP_SDCA_IntMask1[0]=1 */ + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK1, + SDW_SCP_SDCA_INTMASK_SDCA_0 | SDW_SCP_SDCA_INTMASK_SDCA_6); + /* set SCP_SDCA_IntMask2[0]=1 */ + sdw_write_no_pm(rt721->slave, SDW_SCP_SDCA_INTMASK2, + SDW_SCP_SDCA_INTMASK_SDCA_8); + dev_dbg(&rt721->slave->dev, "in %s enable\n", __func__); + rt721_sdca_index_write(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_UAJ_CTL, 0x036E); + /* set XU(et03h) & XU(et0Dh) to Not bypassed */ + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XU03, + RT721_SDCA_CTL_SELECTED_MODE, 0), 0); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_XU0D, + RT721_SDCA_CTL_SELECTED_MODE, 0), 0); + /* trigger GE interrupt */ + rt721_sdca_index_update_bits(rt721, RT721_HDA_SDCA_FLOAT, + RT721_GE_REL_CTRL1, 0x4000, 0x4000); + } + mutex_unlock(&rt721->calibrate_mutex); +} + +static int rt721_sdca_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + int ret; + + rt721->hs_jack = hs_jack; + + ret = pm_runtime_resume_and_get(component->dev); + if (ret < 0) { + if (ret != -EACCES) { + dev_err(component->dev, "%s: failed to resume %d\n", __func__, ret); + return ret; + } + /* pm_runtime not enabled yet */ + dev_dbg(component->dev, "%s: skipping jack init for now\n", __func__); + return 0; + } + + rt721_sdca_jack_init(rt721); + + pm_runtime_mark_last_busy(component->dev); + pm_runtime_put_autosuspend(component->dev); + + return 0; +} + +/* For SDCA control DAC/ADC Gain */ +static int rt721_sdca_set_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned int read_l, read_r, gain_l_val, gain_r_val; + unsigned int adc_vol_flag = 0, changed = 0; + unsigned int lvalue, rvalue; + const unsigned int interval_offset = 0xc0; + const unsigned int tendA = 0x200; + const unsigned int tendB = 0xa00; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume") || + strstr(ucontrol->id.name, "FU0F Capture Volume")) + adc_vol_flag = 1; + + regmap_read(rt721->mbq_regmap, mc->reg, &lvalue); + regmap_read(rt721->mbq_regmap, mc->rreg, &rvalue); + + /* L Channel */ + gain_l_val = ucontrol->value.integer.value[0]; + if (gain_l_val > mc->max) + gain_l_val = mc->max; + + if (mc->shift == 8) { + /* boost gain */ + gain_l_val = gain_l_val * tendB; + } else if (mc->shift == 1) { + /* FU33 boost gain */ + if (gain_l_val == 0) + gain_l_val = 0x8000; + else + gain_l_val = (gain_l_val - 1) * tendA; + } else { + /* ADC/DAC gain */ + if (adc_vol_flag) + gain_l_val = 0x1e00 - ((mc->max - gain_l_val) * interval_offset); + else + gain_l_val = 0 - ((mc->max - gain_l_val) * interval_offset); + gain_l_val &= 0xffff; + } + + /* R Channel */ + gain_r_val = ucontrol->value.integer.value[1]; + if (gain_r_val > mc->max) + gain_r_val = mc->max; + + if (mc->shift == 8) { + /* boost gain */ + gain_r_val = gain_r_val * tendB; + } else if (mc->shift == 1) { + /* FU33 boost gain */ + if (gain_r_val == 0) + gain_r_val = 0x8000; + else + gain_r_val = (gain_r_val - 1) * tendA; + } else { + /* ADC/DAC gain */ + if (adc_vol_flag) + gain_r_val = 0x1e00 - ((mc->max - gain_r_val) * interval_offset); + else + gain_r_val = 0 - ((mc->max - gain_r_val) * interval_offset); + gain_r_val &= 0xffff; + } + + if (lvalue != gain_l_val || rvalue != gain_r_val) + changed = 1; + else + return 0; + + /* Lch*/ + regmap_write(rt721->mbq_regmap, mc->reg, gain_l_val); + + /* Rch */ + regmap_write(rt721->mbq_regmap, mc->rreg, gain_r_val); + + regmap_read(rt721->mbq_regmap, mc->reg, &read_l); + regmap_read(rt721->mbq_regmap, mc->rreg, &read_r); + if (read_r == gain_r_val && read_l == gain_l_val) + return changed; + + return -EIO; +} + +static int rt721_sdca_set_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + unsigned int read_l, read_r, ctl_l = 0, ctl_r = 0; + unsigned int adc_vol_flag = 0; + const unsigned int interval_offset = 0xc0; + const unsigned int tendB = 0xa00; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume") || + strstr(ucontrol->id.name, "FU0F Capture Volume")) + adc_vol_flag = 1; + + regmap_read(rt721->mbq_regmap, mc->reg, &read_l); + regmap_read(rt721->mbq_regmap, mc->rreg, &read_r); + + if (mc->shift == 8) /* boost gain */ + ctl_l = read_l / tendB; + else { + if (adc_vol_flag) + ctl_l = mc->max - (((0x1e00 - read_l) & 0xffff) / interval_offset); + else + ctl_l = mc->max - (((0 - read_l) & 0xffff) / interval_offset); + } + + if (read_l != read_r) { + if (mc->shift == 8) /* boost gain */ + ctl_r = read_r / tendB; + else { /* ADC/DAC gain */ + if (adc_vol_flag) + ctl_r = mc->max - (((0x1e00 - read_r) & 0xffff) / interval_offset); + else + ctl_r = mc->max - (((0 - read_r) & 0xffff) / interval_offset); + } + } else { + ctl_r = ctl_l; + } + + ucontrol->value.integer.value[0] = ctl_l; + ucontrol->value.integer.value[1] = ctl_r; + + return 0; +} + +static int rt721_sdca_set_fu1e_capture_ctl(struct rt721_sdca_priv *rt721) +{ + int err, i; + unsigned int ch_mute; + + for (i = 0; i < ARRAY_SIZE(rt721->fu1e_mixer_mute); i++) { + ch_mute = rt721->fu1e_dapm_mute || rt721->fu1e_mixer_mute[i]; + err = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_01) + i, ch_mute); + if (err < 0) + return err; + } + + return 0; +} + +static int rt721_sdca_fu1e_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + unsigned int i; + + for (i = 0; i < p->count; i++) + ucontrol->value.integer.value[i] = !rt721->fu1e_mixer_mute[i]; + + return 0; +} + +static int rt721_sdca_fu1e_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + int err, changed = 0, i; + + for (i = 0; i < p->count; i++) { + if (rt721->fu1e_mixer_mute[i] != !ucontrol->value.integer.value[i]) + changed = 1; + rt721->fu1e_mixer_mute[i] = !ucontrol->value.integer.value[i]; + } + + err = rt721_sdca_set_fu1e_capture_ctl(rt721); + if (err < 0) + return err; + + return changed; +} + +static int rt721_sdca_set_fu0f_capture_ctl(struct rt721_sdca_priv *rt721) +{ + int err; + unsigned int ch_l, ch_r; + + ch_l = (rt721->fu0f_dapm_mute || rt721->fu0f_mixer_l_mute) ? 0x01 : 0x00; + ch_r = (rt721->fu0f_dapm_mute || rt721->fu0f_mixer_r_mute) ? 0x01 : 0x00; + + err = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_L), ch_l); + if (err < 0) + return err; + + err = regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_MUTE, CH_R), ch_r); + if (err < 0) + return err; + + return 0; +} + +static int rt721_sdca_fu0f_capture_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = !rt721->fu0f_mixer_l_mute; + ucontrol->value.integer.value[1] = !rt721->fu0f_mixer_r_mute; + return 0; +} + +static int rt721_sdca_fu0f_capture_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + int err, changed = 0; + + if (rt721->fu0f_mixer_l_mute != !ucontrol->value.integer.value[0] || + rt721->fu0f_mixer_r_mute != !ucontrol->value.integer.value[1]) + changed = 1; + + rt721->fu0f_mixer_l_mute = !ucontrol->value.integer.value[0]; + rt721->fu0f_mixer_r_mute = !ucontrol->value.integer.value[1]; + err = rt721_sdca_set_fu0f_capture_ctl(rt721); + if (err < 0) + return err; + + return changed; +} + +static int rt721_sdca_fu_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + + if (p->max == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = p->count; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = p->max; + return 0; +} + +static int rt721_sdca_dmic_set_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + unsigned int boost_step = 0x0a00; + unsigned int vol_max = 0x1e00; + unsigned int regvalue, ctl, i; + unsigned int adc_vol_flag = 0; + const unsigned int interval_offset = 0xc0; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume")) + adc_vol_flag = 1; + + /* check all channels */ + for (i = 0; i < p->count; i++) { + regmap_read(rt721->mbq_regmap, p->reg_base + i, &regvalue); + + if (!adc_vol_flag) /* boost gain */ + ctl = regvalue / boost_step; + else { /* ADC gain */ + if (adc_vol_flag) + ctl = p->max - (((vol_max - regvalue) & 0xffff) / interval_offset); + else + ctl = p->max - (((0 - regvalue) & 0xffff) / interval_offset); + } + + ucontrol->value.integer.value[i] = ctl; + } + + return 0; +} + +static int rt721_sdca_dmic_set_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); + struct rt721_sdca_dmic_kctrl_priv *p = + (struct rt721_sdca_dmic_kctrl_priv *)kcontrol->private_value; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned int boost_step = 0x0a00; + unsigned int vol_max = 0x1e00; + unsigned int gain_val[4]; + unsigned int i, adc_vol_flag = 0, changed = 0; + unsigned int regvalue[4]; + const unsigned int interval_offset = 0xc0; + int err; + + if (strstr(ucontrol->id.name, "FU1E Capture Volume")) + adc_vol_flag = 1; + + /* check all channels */ + for (i = 0; i < p->count; i++) { + regmap_read(rt721->mbq_regmap, p->reg_base + i, &regvalue[i]); + + gain_val[i] = ucontrol->value.integer.value[i]; + if (gain_val[i] > p->max) + gain_val[i] = p->max; + + if (!adc_vol_flag) /* boost gain */ + gain_val[i] = gain_val[i] * boost_step; + else { /* ADC gain */ + gain_val[i] = vol_max - ((p->max - gain_val[i]) * interval_offset); + gain_val[i] &= 0xffff; + } + + if (regvalue[i] != gain_val[i]) + changed = 1; + } + + if (!changed) + return 0; + + for (i = 0; i < p->count; i++) { + err = regmap_write(rt721->mbq_regmap, p->reg_base + i, gain_val[i]); + if (err < 0) + dev_err(&rt721->slave->dev, "%#08x can't be set\n", p->reg_base + i); + } + + return changed; +} + +#define RT721_SDCA_PR_VALUE(xreg_base, xcount, xmax, xinvert) \ + ((unsigned long)&(struct rt721_sdca_dmic_kctrl_priv) \ + {.reg_base = xreg_base, .count = xcount, .max = xmax, \ + .invert = xinvert}) + +#define RT721_SDCA_FU_CTRL(xname, reg_base, xmax, xinvert, xcount) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = rt721_sdca_fu_info, \ + .get = rt721_sdca_fu1e_capture_get, \ + .put = rt721_sdca_fu1e_capture_put, \ + .private_value = RT721_SDCA_PR_VALUE(reg_base, xcount, xmax, xinvert)} + +#define RT721_SDCA_EXT_TLV(xname, reg_base, xhandler_get,\ + xhandler_put, xcount, xmax, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ + SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .tlv.p = (tlv_array), \ + .info = rt721_sdca_fu_info, \ + .get = xhandler_get, .put = xhandler_put, \ + .private_value = RT721_SDCA_PR_VALUE(reg_base, xcount, xmax, 0) } + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6525, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -1725, 75, 0); +static const DECLARE_TLV_DB_SCALE(boost_vol_tlv, 0, 1000, 0); +static const DECLARE_TLV_DB_SCALE(mic2_boost_vol_tlv, -200, 200, 0); + +static const struct snd_kcontrol_new rt721_sdca_controls[] = { + /* Headphone playback settings */ + SOC_DOUBLE_R_EXT_TLV("FU05 Playback Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_VOLUME, CH_R), 0, 0x57, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, out_vol_tlv), + /* Headset mic capture settings */ + SOC_DOUBLE_EXT("FU0F Capture Switch", SND_SOC_NOPM, 0, 1, 1, 0, + rt721_sdca_fu0f_capture_get, rt721_sdca_fu0f_capture_put), + SOC_DOUBLE_R_EXT_TLV("FU0F Capture Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU0F, + RT721_SDCA_CTL_FU_VOLUME, CH_R), 0, 0x3f, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, mic_vol_tlv), + SOC_DOUBLE_R_EXT_TLV("FU33 Boost Volume", + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_L), + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PLATFORM_FU44, + RT721_SDCA_CTL_FU_CH_GAIN, CH_R), 1, 0x15, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, mic2_boost_vol_tlv), + /* AMP playback settings */ + SOC_DOUBLE_R_EXT_TLV("FU06 Playback Volume", + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_VOLUME, CH_L), + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_VOLUME, CH_R), 0, 0x57, 0, + rt721_sdca_set_gain_get, rt721_sdca_set_gain_put, out_vol_tlv), + /* DMIC capture settings */ + RT721_SDCA_FU_CTRL("FU1E Capture Switch", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_MUTE, CH_01), 1, 1, 4), + RT721_SDCA_EXT_TLV("FU1E Capture Volume", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_USER_FU1E, + RT721_SDCA_CTL_FU_VOLUME, CH_01), + rt721_sdca_dmic_set_gain_get, rt721_sdca_dmic_set_gain_put, + 4, 0x3f, mic_vol_tlv), + RT721_SDCA_EXT_TLV("FU15 Boost Volume", + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_FU15, + RT721_SDCA_CTL_FU_CH_GAIN, CH_01), + rt721_sdca_dmic_set_gain_get, rt721_sdca_dmic_set_gain_put, + 4, 3, boost_vol_tlv), +}; + +static int rt721_sdca_adc_mux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned int val = 0, mask_sft, mask; + + if (strstr(ucontrol->id.name, "ADC 09 Mux")) { + mask_sft = 12; + mask = 0x7; + } else if (strstr(ucontrol->id.name, "ADC 08 R Mux")) { + mask_sft = 10; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 08 L Mux")) { + mask_sft = 8; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 R Mux")) { + mask_sft = 6; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 L Mux")) { + mask_sft = 4; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 R Mux")) { + mask_sft = 2; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 L Mux")) { + mask_sft = 0; + mask = 0x3; + } else + return -EINVAL; + + rt721_sdca_index_read(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, &val); + + ucontrol->value.enumerated.item[0] = (val >> mask_sft) & mask; + + return 0; +} + +static int rt721_sdca_adc_mux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_dapm_kcontrol_component(kcontrol); + struct snd_soc_dapm_context *dapm = + snd_soc_dapm_kcontrol_dapm(kcontrol); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int val, val2 = 0, change, mask_sft, mask; + unsigned int check; + + if (item[0] >= e->items) + return -EINVAL; + + if (strstr(ucontrol->id.name, "ADC 09 Mux")) { + mask_sft = 12; + mask = 0x7; + } else if (strstr(ucontrol->id.name, "ADC 08 R Mux")) { + mask_sft = 10; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 08 L Mux")) { + mask_sft = 8; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 R Mux")) { + mask_sft = 6; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 10 L Mux")) { + mask_sft = 4; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 R Mux")) { + mask_sft = 2; + mask = 0x3; + } else if (strstr(ucontrol->id.name, "ADC 07 L Mux")) { + mask_sft = 0; + mask = 0x3; + } else + return -EINVAL; + + val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + rt721_sdca_index_read(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, &val2); + + if (strstr(ucontrol->id.name, "ADC 09 Mux")) + val2 = (val2 >> mask_sft) & 0x7; + else + val2 = (val2 >> mask_sft) & 0x3; + + if (val == val2) + change = 0; + else + change = 1; + + if (change) { + rt721_sdca_index_read(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, &check); + rt721_sdca_index_update_bits(rt721, RT721_HDA_SDCA_FLOAT, + RT721_HDA_LEGACY_MUX_CTL0, mask << mask_sft, + val << mask_sft); + } + + snd_soc_dapm_mux_update_power(dapm, kcontrol, + item[0], e, NULL); + + return change; +} + +static const char * const adc09_mux_text[] = { + "MIC2", + "LINE1", + "LINE2", +}; +static const char * const adc07_10_mux_text[] = { + "DMIC1 RE", + "DMIC1 FE", + "DMIC2 RE", + "DMIC2 FE", +}; + +static SOC_ENUM_SINGLE_DECL( + rt721_adc09_enum, SND_SOC_NOPM, 0, adc09_mux_text); +static SOC_ENUM_SINGLE_DECL( + rt721_dmic_enum, SND_SOC_NOPM, 0, adc07_10_mux_text); + +static const struct snd_kcontrol_new rt721_sdca_adc09_mux = + SOC_DAPM_ENUM_EXT("ADC 09 Mux", rt721_adc09_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc08_r_mux = + SOC_DAPM_ENUM_EXT("ADC 08 R Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc08_l_mux = + SOC_DAPM_ENUM_EXT("ADC 08 L Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc10_r_mux = + SOC_DAPM_ENUM_EXT("ADC 10 R Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc10_l_mux = + SOC_DAPM_ENUM_EXT("ADC 10 L Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc07_r_mux = + SOC_DAPM_ENUM_EXT("ADC 07 R Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); +static const struct snd_kcontrol_new rt721_sdca_adc07_l_mux = + SOC_DAPM_ENUM_EXT("ADC 07 L Mux", rt721_dmic_enum, + rt721_sdca_adc_mux_get, rt721_sdca_adc_mux_put); + + +static int rt721_sdca_fu42_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(100); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_L), unmute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_R), unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_L), mute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_USER_FU05, + RT721_SDCA_CTL_FU_MUTE, CH_R), mute); + break; + } + return 0; +} + +static int rt721_sdca_fu21_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_L), unmute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_R), unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_L), mute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_USER_FU06, + RT721_SDCA_CTL_FU_MUTE, CH_R), mute); + break; + } + return 0; +} + +static int rt721_sdca_fu23_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char unmute = 0x0, mute = 0x1; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_L), unmute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_R), unmute); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_L), mute); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE23, + RT721_SDCA_CTL_FU_MUTE, CH_R), mute); + break; + } + return 0; +} + +static int rt721_sdca_fu113_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt721->fu1e_dapm_mute = false; + rt721_sdca_set_fu1e_capture_ctl(rt721); + break; + case SND_SOC_DAPM_PRE_PMD: + rt721->fu1e_dapm_mute = true; + rt721_sdca_set_fu1e_capture_ctl(rt721); + break; + } + return 0; +} + +static int rt721_sdca_fu36_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt721->fu0f_dapm_mute = false; + rt721_sdca_set_fu0f_capture_ctl(rt721); + break; + case SND_SOC_DAPM_PRE_PMD: + rt721->fu0f_dapm_mute = true; + rt721_sdca_set_fu0f_capture_ctl(rt721); + break; + } + return 0; +} + +static int rt721_sdca_pde47_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE40, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE40, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static int rt721_sdca_pde41_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE41, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_PDE41, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static int rt721_sdca_pde11_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_PDE2A, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_PDE2A, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static int rt721_sdca_pde34_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + unsigned char ps0 = 0x0, ps3 = 0x3; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE12, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps0); + break; + case SND_SOC_DAPM_PRE_PMD: + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_PDE12, + RT721_SDCA_CTL_REQ_POWER_STATE, 0), ps3); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget rt721_sdca_dapm_widgets[] = { + SND_SOC_DAPM_OUTPUT("HP"), + SND_SOC_DAPM_OUTPUT("SPK"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("LINE2"), + SND_SOC_DAPM_INPUT("DMIC1_2"), + SND_SOC_DAPM_INPUT("DMIC3_4"), + + SND_SOC_DAPM_SUPPLY("PDE 41", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde41_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 47", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde47_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 11", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde11_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_SUPPLY("PDE 34", SND_SOC_NOPM, 0, 0, + rt721_sdca_pde34_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_DAC_E("FU 21", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu21_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("FU 23", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu23_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("FU 42", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu42_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("FU 36", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu36_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_ADC_E("FU 113", NULL, SND_SOC_NOPM, 0, 0, + rt721_sdca_fu113_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MUX("ADC 09 Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc09_mux), + SND_SOC_DAPM_MUX("ADC 08 R Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc08_r_mux), + SND_SOC_DAPM_MUX("ADC 08 L Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc08_l_mux), + SND_SOC_DAPM_MUX("ADC 10 R Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc10_r_mux), + SND_SOC_DAPM_MUX("ADC 10 L Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc10_l_mux), + SND_SOC_DAPM_MUX("ADC 07 R Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc07_r_mux), + SND_SOC_DAPM_MUX("ADC 07 L Mux", SND_SOC_NOPM, 0, 0, + &rt721_sdca_adc07_l_mux), + + SND_SOC_DAPM_AIF_IN("DP1RX", "DP1 Headphone Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP2TX", "DP2 Headset Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DP3RX", "DP3 Speaker Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("DP6TX", "DP6 DMic Capture", 0, SND_SOC_NOPM, 0, 0), +}; + +static const struct snd_soc_dapm_route rt721_sdca_audio_map[] = { + {"FU 42", NULL, "DP1RX"}, + {"FU 21", NULL, "DP3RX"}, + {"FU 23", NULL, "DP3RX"}, + + {"ADC 09 Mux", "MIC2", "MIC2"}, + {"ADC 09 Mux", "LINE1", "LINE1"}, + {"ADC 09 Mux", "LINE2", "LINE2"}, + {"ADC 07 R Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 07 R Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 07 R Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 07 R Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 07 L Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 07 L Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 07 L Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 07 L Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 08 R Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 08 R Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 08 R Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 08 R Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 08 L Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 08 L Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 08 L Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 08 L Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 10 R Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 10 R Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 10 R Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 10 R Mux", "DMIC2 FE", "DMIC3_4"}, + {"ADC 10 L Mux", "DMIC1 RE", "DMIC1_2"}, + {"ADC 10 L Mux", "DMIC1 FE", "DMIC1_2"}, + {"ADC 10 L Mux", "DMIC2 RE", "DMIC3_4"}, + {"ADC 10 L Mux", "DMIC2 FE", "DMIC3_4"}, + {"FU 36", NULL, "PDE 34"}, + {"FU 36", NULL, "ADC 09 Mux"}, + {"FU 113", NULL, "PDE 11"}, + {"FU 113", NULL, "ADC 07 R Mux"}, + {"FU 113", NULL, "ADC 07 L Mux"}, + {"FU 113", NULL, "ADC 10 R Mux"}, + {"FU 113", NULL, "ADC 10 L Mux"}, + {"DP2TX", NULL, "FU 36"}, + {"DP6TX", NULL, "FU 113"}, + + {"HP", NULL, "PDE 47"}, + {"HP", NULL, "FU 42"}, + {"SPK", NULL, "PDE 41"}, + {"SPK", NULL, "FU 21"}, + {"SPK", NULL, "FU 23"}, +}; + +static int rt721_sdca_parse_dt(struct rt721_sdca_priv *rt721, struct device *dev) +{ + device_property_read_u32(dev, "realtek,jd-src", &rt721->jd_src); + + return 0; +} + +static int rt721_sdca_probe(struct snd_soc_component *component) +{ + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + int ret; + + rt721_sdca_parse_dt(rt721, &rt721->slave->dev); + rt721->component = component; + + ret = pm_runtime_resume(component->dev); + if (ret < 0 && ret != -EACCES) + return ret; + + return 0; +} + +static const struct snd_soc_component_driver soc_sdca_dev_rt721 = { + .probe = rt721_sdca_probe, + .controls = rt721_sdca_controls, + .num_controls = ARRAY_SIZE(rt721_sdca_controls), + .dapm_widgets = rt721_sdca_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt721_sdca_dapm_widgets), + .dapm_routes = rt721_sdca_audio_map, + .num_dapm_routes = ARRAY_SIZE(rt721_sdca_audio_map), + .set_jack = rt721_sdca_set_jack_detect, + .endianness = 1, +}; + +static int rt721_sdca_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void rt721_sdca_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int rt721_sdca_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_runtime *sdw_stream; + int retval, port, num_channels; + unsigned int sampling_rate; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!rt721->slave) + return -EINVAL; + + /* + * RT721_AIF1 with port = 1 for headphone playback + * RT721_AIF1 with port = 2 for headset-mic capture + * RT721_AIF2 with port = 3 for speaker playback + * RT721_AIF3 with port = 6 for digital-mic capture + */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + if (dai->id == RT721_AIF1) + port = 1; + else if (dai->id == RT721_AIF2) + port = 3; + else + return -EINVAL; + } else { + direction = SDW_DATA_DIR_TX; + if (dai->id == RT721_AIF1) + port = 2; + else if (dai->id == RT721_AIF3) + port = 6; + else + return -EINVAL; + } + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = GENMASK(num_channels - 1, 0); + port_config.num = port; + + retval = sdw_stream_add_slave(rt721->slave, &stream_config, + &port_config, 1, sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + if (params_channels(params) > 16) { + dev_err(component->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + /* sampling rate configuration */ + switch (params_rate(params)) { + case 8000: + sampling_rate = RT721_SDCA_RATE_8000HZ; + break; + case 16000: + sampling_rate = RT721_SDCA_RATE_16000HZ; + break; + case 24000: + sampling_rate = RT721_SDCA_RATE_24000HZ; + break; + case 32000: + sampling_rate = RT721_SDCA_RATE_32000HZ; + break; + case 44100: + sampling_rate = RT721_SDCA_RATE_44100HZ; + break; + case 48000: + sampling_rate = RT721_SDCA_RATE_48000HZ; + break; + case 96000: + sampling_rate = RT721_SDCA_RATE_96000HZ; + break; + case 192000: + sampling_rate = RT721_SDCA_RATE_192000HZ; + break; + case 384000: + sampling_rate = RT721_SDCA_RATE_384000HZ; + break; + case 768000: + sampling_rate = RT721_SDCA_RATE_768000HZ; + break; + default: + dev_err(component->dev, "Rate %d is not supported\n", + params_rate(params)); + return -EINVAL; + } + + /* set sampling frequency */ + if (dai->id == RT721_AIF1) { + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS01, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_JACK_CODEC, RT721_SDCA_ENT_CS11, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + } + + if (dai->id == RT721_AIF2) + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_AMP, RT721_SDCA_ENT_CS31, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + + if (dai->id == RT721_AIF3) + regmap_write(rt721->regmap, + SDW_SDCA_CTL(FUNC_NUM_MIC_ARRAY, RT721_SDCA_ENT_CS1F, + RT721_SDCA_CTL_SAMPLE_FREQ_INDEX, 0), sampling_rate); + + return 0; +} + +static int rt721_sdca_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt721_sdca_priv *rt721 = snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt721->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt721->slave, sdw_stream); + return 0; +} + +#define RT721_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000) +#define RT721_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + +static const struct snd_soc_dai_ops rt721_sdca_ops = { + .hw_params = rt721_sdca_pcm_hw_params, + .hw_free = rt721_sdca_pcm_hw_free, + .set_stream = rt721_sdca_set_sdw_stream, + .shutdown = rt721_sdca_shutdown, +}; + +static struct snd_soc_dai_driver rt721_sdca_dai[] = { + { + .name = "rt721-sdca-aif1", + .id = RT721_AIF1, + .playback = { + .stream_name = "DP1 Headphone Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .capture = { + .stream_name = "DP2 Headset Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .ops = &rt721_sdca_ops, + }, + { + .name = "rt721-sdca-aif2", + .id = RT721_AIF2, + .playback = { + .stream_name = "DP3 Speaker Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .ops = &rt721_sdca_ops, + }, + { + .name = "rt721-sdca-aif3", + .id = RT721_AIF3, + .capture = { + .stream_name = "DP6 DMic Capture", + .channels_min = 1, + .channels_max = 4, + .rates = RT721_STEREO_RATES, + .formats = RT721_FORMATS, + }, + .ops = &rt721_sdca_ops, + } +}; + +int rt721_sdca_init(struct device *dev, struct regmap *regmap, + struct regmap *mbq_regmap, struct sdw_slave *slave) +{ + struct rt721_sdca_priv *rt721; + + rt721 = devm_kzalloc(dev, sizeof(*rt721), GFP_KERNEL); + if (!rt721) + return -ENOMEM; + + dev_set_drvdata(dev, rt721); + rt721->slave = slave; + rt721->regmap = regmap; + rt721->mbq_regmap = mbq_regmap; + + regcache_cache_only(rt721->regmap, true); + regcache_cache_only(rt721->mbq_regmap, true); + + mutex_init(&rt721->calibrate_mutex); + mutex_init(&rt721->disable_irq_lock); + + INIT_DELAYED_WORK(&rt721->jack_detect_work, rt721_sdca_jack_detect_handler); + INIT_DELAYED_WORK(&rt721->jack_btn_check_work, rt721_sdca_btn_check_handler); + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt721->hw_init = false; + rt721->first_hw_init = false; + rt721->fu1e_dapm_mute = true; + rt721->fu0f_dapm_mute = true; + rt721->fu0f_mixer_l_mute = rt721->fu0f_mixer_r_mute = true; + rt721->fu1e_mixer_mute[0] = rt721->fu1e_mixer_mute[1] = + rt721->fu1e_mixer_mute[2] = rt721->fu1e_mixer_mute[3] = true; + + return devm_snd_soc_register_component(dev, + &soc_sdca_dev_rt721, rt721_sdca_dai, ARRAY_SIZE(rt721_sdca_dai)); +} + +int rt721_sdca_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt721_sdca_priv *rt721 = dev_get_drvdata(dev); + + rt721->disable_irq = false; + + if (rt721->hw_init) + return 0; + + regcache_cache_only(rt721->regmap, false); + regcache_cache_only(rt721->mbq_regmap, false); + if (rt721->first_hw_init) { + regcache_cache_bypass(rt721->regmap, true); + regcache_cache_bypass(rt721->mbq_regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + rt721_sdca_dmic_preset(rt721); + rt721_sdca_amp_preset(rt721); + rt721_sdca_jack_preset(rt721); + if (rt721->first_hw_init) { + regcache_cache_bypass(rt721->regmap, false); + regcache_mark_dirty(rt721->regmap); + regcache_cache_bypass(rt721->mbq_regmap, false); + regcache_mark_dirty(rt721->mbq_regmap); + } else + rt721->first_hw_init = true; + + /* Mark Slave initialization complete */ + rt721->hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + return 0; +} + +MODULE_DESCRIPTION("ASoC RT721 SDCA SDW driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt721-sdca.h b/sound/soc/codecs/rt721-sdca.h new file mode 100644 index 000000000000..69cf51bb6c95 --- /dev/null +++ b/sound/soc/codecs/rt721-sdca.h @@ -0,0 +1,277 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * rt721-sdca.h -- RT721 SDCA ALSA SoC audio driver header + * + * Copyright(c) 2024 Realtek Semiconductor Corp. + */ + +#ifndef __RT721_H__ +#define __RT721_H__ + +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/soc.h> +#include <linux/workqueue.h> + +struct rt721_sdca_priv { + struct regmap *regmap; + struct regmap *mbq_regmap; + struct snd_soc_component *component; + struct sdw_slave *slave; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; + struct mutex calibrate_mutex; + struct mutex disable_irq_lock; + bool disable_irq; + /* For Headset jack & Headphone */ + unsigned int scp_sdca_stat1; + unsigned int scp_sdca_stat2; + struct snd_soc_jack *hs_jack; + struct delayed_work jack_detect_work; + struct delayed_work jack_btn_check_work; + int jack_type; + int jd_src; + bool fu0f_dapm_mute; + bool fu0f_mixer_l_mute; + bool fu0f_mixer_r_mute; + /* For DMIC */ + bool fu1e_dapm_mute; + bool fu1e_mixer_mute[4]; +}; + +struct rt721_sdca_dmic_kctrl_priv { + unsigned int reg_base; + unsigned int count; + unsigned int max; + unsigned int invert; +}; + +/* NID */ +#define RT721_ANA_POW_PART 0x01 +#define RT721_DAC_CTRL 0x04 +#define RT721_JD_CTRL 0x09 +#define RT721_CBJ_CTRL 0x0a +#define RT721_CAP_PORT_CTRL 0x0c +#define RT721_CLASD_AMP_CTRL 0x0d +#define RT721_VENDOR_REG 0x20 +#define RT721_RC_CALIB_CTRL 0x40 +#define RT721_VENDOR_EQ_L 0x53 +#define RT721_VENDOR_EQ_R 0x54 +#define RT721_VENDOR_HP_CALI 0x56 +#define RT721_VENDOR_CHARGE_PUMP 0x57 +#define RT721_VENDOR_CLASD_CALI 0x58 +#define RT721_VENDOR_IMS_DRE 0x5b +#define RT721_VENDOR_SPK_EFUSE 0x5c +#define RT721_VENDOR_LEVEL_CTRL 0x5d +#define RT721_VENDOR_ANA_CTL 0x5f +#define RT721_HDA_SDCA_FLOAT 0x61 + +/* Index (NID:01h) */ +#define RT721_MBIAS_LV_CTRL2 0x07 +#define RT721_VREF1_HV_CTRL1 0x0a +#define RT721_VREF2_LV_CTRL1 0x0b + +/* Index (NID:04h) */ +#define RT721_DAC_2CH_CTRL3 0x02 +#define RT721_DAC_2CH_CTRL4 0x03 + +/* Index (NID:09h) */ +#define RT721_JD_1PIN_GAT_CTRL2 0x07 + +/* Index (NID:0ah) */ +#define RT721_CBJ_A0_GAT_CTRL1 0x04 +#define RT721_CBJ_A0_GAT_CTRL2 0x05 + +/* Index (NID:0Ch) */ +#define RT721_HP_AMP_2CH_CAL1 0x05 +#define RT721_HP_AMP_2CH_CAL4 0x08 +#define RT721_HP_AMP_2CH_CAL18 0x1b + +/* Index (NID:0dh) */ +#define RT721_CLASD_AMP_2CH_CAL 0x14 + +/* Index (NID:20h) */ +#define RT721_JD_PRODUCT_NUM 0x00 +#define RT721_ANALOG_BIAS_CTL3 0x04 +#define RT721_JD_CTRL1 0x09 +#define RT721_LDO2_3_CTL1 0x0e +#define RT721_GPIO_PAD_CTRL5 0x13 +#define RT721_LDO1_CTL 0x1a +#define RT721_HP_JD_CTRL 0x24 +#define RT721_VD_HIDDEN_CTRL 0x26 +#define RT721_CLSD_CTRL6 0x3c +#define RT721_COMBO_JACK_AUTO_CTL1 0x45 +#define RT721_COMBO_JACK_AUTO_CTL2 0x46 +#define RT721_COMBO_JACK_AUTO_CTL3 0x47 +#define RT721_DIGITAL_MISC_CTRL4 0x4a +#define RT721_VREFO_GAT 0x63 +#define RT721_FSM_CTL 0x67 +#define RT721_SDCA_INTR_REC 0x82 +#define RT721_SW_CONFIG1 0x8a +#define RT721_SW_CONFIG2 0x8b + +/* Index (NID:40h) */ +#define RT721_RC_CALIB_CTRL0 0x00 + +/* Index (NID:58h) */ +#define RT721_DAC_DC_CALI_CTL1 0x01 +#define RT721_DAC_DC_CALI_CTL2 0x02 +#define RT721_DAC_DC_CALI_CTL3 0x03 + +/* Index (NID:5fh) */ +#define RT721_MISC_POWER_CTL0 0x00 +#define RT721_MISC_POWER_CTL31 0x31 +#define RT721_UAJ_TOP_TCON13 0x44 +#define RT721_UAJ_TOP_TCON14 0x45 +#define RT721_UAJ_TOP_TCON17 0x48 + +/* Index (NID:61h) */ +#define RT721_HDA_LEGACY_MUX_CTL0 0x00 +#define RT721_HDA_LEGACY_UAJ_CTL 0x02 +#define RT721_HDA_LEGACY_CTL1 0x05 +#define RT721_HDA_LEGACY_RESET_CTL 0x06 +#define RT721_GE_REL_CTRL1 0x0d +#define RT721_HDA_LEGACY_GPIO_WAKE_EN_CTL 0x0e +#define RT721_GE_SDCA_RST_CTRL 0x10 +#define RT721_INT_RST_EN_CTRL 0x11 +#define RT721_XU_EVENT_EN 0x13 +#define RT721_INLINE_CTL2 0x17 +#define RT721_UMP_HID_CTRL1 0x18 +#define RT721_UMP_HID_CTRL2 0x19 +#define RT721_UMP_HID_CTRL3 0x1a +#define RT721_UMP_HID_CTRL4 0x1b +#define RT721_UMP_HID_CTRL5 0x1c +#define RT721_FUNC_FLOAT_CTL0 0x22 +#define RT721_FUNC_FLOAT_CTL1 0x23 +#define RT721_FUNC_FLOAT_CTL2 0x24 +#define RT721_FUNC_FLOAT_CTL3 0x25 +#define RT721_ENT_FLOAT_CTL0 0x29 +#define RT721_ENT_FLOAT_CTL1 0x2c +#define RT721_ENT_FLOAT_CTL2 0x2d +#define RT721_ENT_FLOAT_CTL3 0x2e +#define RT721_ENT_FLOAT_CTL4 0x2f +#define RT721_CH_FLOAT_CTL1 0x45 +#define RT721_CH_FLOAT_CTL2 0x46 +#define RT721_ENT_FLOAT_CTL5 0x53 +#define RT721_ENT_FLOAT_CTL6 0x54 +#define RT721_ENT_FLOAT_CTL7 0x55 +#define RT721_ENT_FLOAT_CTL8 0x57 +#define RT721_ENT_FLOAT_CTL9 0x5a +#define RT721_ENT_FLOAT_CTL10 0x5b +#define RT721_CH_FLOAT_CTL3 0x6a +#define RT721_CH_FLOAT_CTL4 0x6d +#define RT721_CH_FLOAT_CTL5 0x70 +#define RT721_CH_FLOAT_CTL6 0x92 + +/* Parameter & Verb control 01 (0x26)(NID:20h) */ +#define RT721_HIDDEN_REG_SW_RESET (0x1 << 14) + +/* Buffer address for HID */ +#define RT721_BUF_ADDR_HID1 0x44030000 +#define RT721_BUF_ADDR_HID2 0x44030020 + +/* RT721 SDCA Control - function number */ +#define FUNC_NUM_JACK_CODEC 0x01 +#define FUNC_NUM_MIC_ARRAY 0x02 +#define FUNC_NUM_HID 0x03 +#define FUNC_NUM_AMP 0x04 + +/* RT721 SDCA entity */ +#define RT721_SDCA_ENT_HID01 0x01 +#define RT721_SDCA_ENT_XUV 0x03 +#define RT721_SDCA_ENT_GE49 0x49 +#define RT721_SDCA_ENT_USER_FU05 0x05 +#define RT721_SDCA_ENT_USER_FU06 0x06 +#define RT721_SDCA_ENT_USER_FU0F 0x0f +#define RT721_SDCA_ENT_USER_FU10 0x19 +#define RT721_SDCA_ENT_USER_FU1E 0x1e +#define RT721_SDCA_ENT_FU15 0x15 +#define RT721_SDCA_ENT_PDE23 0x23 +#define RT721_SDCA_ENT_PDE40 0x40 +#define RT721_SDCA_ENT_PDE41 0x41 +#define RT721_SDCA_ENT_PDE11 0x11 +#define RT721_SDCA_ENT_PDE12 0x12 +#define RT721_SDCA_ENT_PDE2A 0x2a +#define RT721_SDCA_ENT_CS01 0x01 +#define RT721_SDCA_ENT_CS11 0x11 +#define RT721_SDCA_ENT_CS1F 0x1f +#define RT721_SDCA_ENT_CS1C 0x1c +#define RT721_SDCA_ENT_CS31 0x31 +#define RT721_SDCA_ENT_OT23 0x42 +#define RT721_SDCA_ENT_IT26 0x26 +#define RT721_SDCA_ENT_IT09 0x09 +#define RT721_SDCA_ENT_PLATFORM_FU15 0x15 +#define RT721_SDCA_ENT_PLATFORM_FU44 0x44 +#define RT721_SDCA_ENT_XU03 0x03 +#define RT721_SDCA_ENT_XU0D 0x0d +#define RT721_SDCA_ENT_FU55 0x55 + +/* RT721 SDCA control */ +#define RT721_SDCA_CTL_SAMPLE_FREQ_INDEX 0x10 +#define RT721_SDCA_CTL_FU_MUTE 0x01 +#define RT721_SDCA_CTL_FU_VOLUME 0x02 +#define RT721_SDCA_CTL_HIDTX_CURRENT_OWNER 0x10 +#define RT721_SDCA_CTL_HIDTX_SET_OWNER_TO_DEVICE 0x11 +#define RT721_SDCA_CTL_HIDTX_MESSAGE_OFFSET 0x12 +#define RT721_SDCA_CTL_HIDTX_MESSAGE_LENGTH 0x13 +#define RT721_SDCA_CTL_SELECTED_MODE 0x01 +#define RT721_SDCA_CTL_DETECTED_MODE 0x02 +#define RT721_SDCA_CTL_REQ_POWER_STATE 0x01 +#define RT721_SDCA_CTL_VENDOR_DEF 0x30 +#define RT721_SDCA_CTL_XUV 0x34 +#define RT721_SDCA_CTL_FU_CH_GAIN 0x0b + +/* RT721 SDCA channel */ +#define CH_L 0x01 +#define CH_R 0x02 +#define CH_01 0x01 +#define CH_02 0x02 +#define CH_03 0x03 +#define CH_04 0x04 +#define CH_08 0x08 +#define CH_09 0x09 +#define CH_0A 0x0a + +/* sample frequency index */ +#define RT721_SDCA_RATE_8000HZ 0x01 +#define RT721_SDCA_RATE_11025HZ 0x02 +#define RT721_SDCA_RATE_12000HZ 0x03 +#define RT721_SDCA_RATE_16000HZ 0x04 +#define RT721_SDCA_RATE_22050HZ 0x05 +#define RT721_SDCA_RATE_24000HZ 0x06 +#define RT721_SDCA_RATE_32000HZ 0x07 +#define RT721_SDCA_RATE_44100HZ 0x08 +#define RT721_SDCA_RATE_48000HZ 0x09 +#define RT721_SDCA_RATE_88200HZ 0x0a +#define RT721_SDCA_RATE_96000HZ 0x0b +#define RT721_SDCA_RATE_176400HZ 0x0c +#define RT721_SDCA_RATE_192000HZ 0x0d +#define RT721_SDCA_RATE_384000HZ 0x0e +#define RT721_SDCA_RATE_768000HZ 0x0f + +enum { + RT721_AIF1, /* For headset mic and headphone */ + RT721_AIF2, /* For speaker */ + RT721_AIF3, /* For dmic */ + RT721_AIFS, +}; + +enum rt721_sdca_jd_src { + RT721_JD_NULL, + RT721_JD1, + RT721_JD2, +}; + +int rt721_sdca_io_init(struct device *dev, struct sdw_slave *slave); +int rt721_sdca_init(struct device *dev, struct regmap *regmap, + struct regmap *mbq_regmap, struct sdw_slave *slave); +int rt721_sdca_index_write(struct rt721_sdca_priv *rt721, + unsigned int nid, unsigned int reg, unsigned int value); +int rt721_sdca_index_read(struct rt721_sdca_priv *rt721, + unsigned int nid, unsigned int reg, unsigned int *value); + +int rt721_sdca_jack_detect(struct rt721_sdca_priv *rt721, bool *hp, bool *mic); +#endif /* __RT721_H__ */ -- 2.34.1

2 2

[PATCH v1] ALSA: Fix typos in comments across various files
by Yu Jiaoliang 24 Sep '24

24 Sep '24

This patch fixes typos in comments in the atomisp driver. These changes improve code readability without affecting functionality. Signed-off-by: Yu Jiaoliang <yujiaoliang(a)vivo.com> --- sound/core/compress_offload.c | 2 +- sound/core/oss/rate.c | 2 +- sound/core/pcm_native.c | 2 +- sound/core/sound.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index b8c0d6edbdd1..bdf1d78de833 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -288,7 +288,7 @@ static ssize_t snd_compr_write(struct file *f, const char __user *buf, stream = &data->stream; guard(mutex)(&stream->device->lock); - /* write is allowed when stream is running or has been steup */ + /* write is allowed when stream is running or has been setup */ switch (stream->runtime->state) { case SNDRV_PCM_STATE_SETUP: case SNDRV_PCM_STATE_PREPARED: diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 98269119347f..b56eeda5e30e 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -294,7 +294,7 @@ static int rate_action(struct snd_pcm_plugin *plugin, default: break; } - return 0; /* silenty ignore other actions */ + return 0; /* silently ignore other actions */ } int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug, diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 99e39b5359cc..b49bc89c0758 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3115,7 +3115,7 @@ struct snd_pcm_sync_ptr32 { } c; } __packed; -/* recalcuate the boundary within 32bit */ +/* recalculate the boundary within 32bit */ static snd_pcm_uframes_t recalculate_boundary(struct snd_pcm_runtime *runtime) { snd_pcm_uframes_t boundary; diff --git a/sound/core/sound.c b/sound/core/sound.c index b9db9aa0bfcb..6531a67f13b3 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -133,7 +133,7 @@ static struct snd_minor *autoload_device(unsigned int minor) /* /dev/aloadSEQ */ snd_request_other(minor); } - mutex_lock(&sound_mutex); /* reacuire lock */ + mutex_lock(&sound_mutex); /* reacquire lock */ return snd_minors[minor]; } #else /* !CONFIG_MODULES */ -- 2.34.1

1 0

Plugin for windows sound api
by GitHub issues - opened 22 Sep '24

22 Sep '24

alsa-project/alsa-plugins issue #61 was opened from uis246: There is already OSS plugin for alsa to support BSD, there are many sound server plugins for alsa. Is it possible to make such plugin for sounds api used in windows? Not sure what is better place to start this discussion. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/61 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH] ALSA: Drop explicit initialization of struct i2c_device_id::driver_data to 0
by Uwe Kleine-König 20 Sep '24

20 Sep '24

These drivers don't use the driver_data member of struct i2c_device_id, so don't explicitly initialize this member. This prepares putting driver_data in an anonymous union which requires either no initialization or named designators. But it's also a nice cleanup on its own. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)baylibre.com> --- sound/aoa/codecs/onyx.c | 2 +- sound/aoa/codecs/tas.c | 2 +- sound/pci/hda/cs35l41_hda_i2c.c | 2 +- sound/pci/hda/tas2781_hda_i2c.c | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index e90e03bb0dc0..ac347a14f282 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -1040,7 +1040,7 @@ static void onyx_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id onyx_i2c_id[] = { - { "MAC,pcm3052", 0 }, + { "MAC,pcm3052" }, { } }; MODULE_DEVICE_TABLE(i2c,onyx_i2c_id); diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c index be9822ebf9f8..804b2ebbe28f 100644 --- a/sound/aoa/codecs/tas.c +++ b/sound/aoa/codecs/tas.c @@ -927,7 +927,7 @@ static void tas_i2c_remove(struct i2c_client *client) } static const struct i2c_device_id tas_i2c_id[] = { - { "MAC,tas3004", 0 }, + { "MAC,tas3004" }, { } }; MODULE_DEVICE_TABLE(i2c,tas_i2c_id); diff --git a/sound/pci/hda/cs35l41_hda_i2c.c b/sound/pci/hda/cs35l41_hda_i2c.c index 603e9bff3a71..bb84740c8520 100644 --- a/sound/pci/hda/cs35l41_hda_i2c.c +++ b/sound/pci/hda/cs35l41_hda_i2c.c @@ -39,7 +39,7 @@ static void cs35l41_hda_i2c_remove(struct i2c_client *clt) } static const struct i2c_device_id cs35l41_hda_i2c_id[] = { - { "cs35l41-hda", 0 }, + { "cs35l41-hda" }, {} }; diff --git a/sound/pci/hda/tas2781_hda_i2c.c b/sound/pci/hda/tas2781_hda_i2c.c index f58f434e7110..4b9dc84ce6bb 100644 --- a/sound/pci/hda/tas2781_hda_i2c.c +++ b/sound/pci/hda/tas2781_hda_i2c.c @@ -951,7 +951,7 @@ static const struct dev_pm_ops tas2781_hda_pm_ops = { }; static const struct i2c_device_id tas2781_hda_i2c_id[] = { - { "tas2781-hda", 0 }, + { "tas2781-hda" }, {} }; base-commit: 62f92d634458a1e308bb699986b9147a6d670457 -- 2.45.2

2 1

[PATCH] ASoC: cs35l45: Corrects cs35l45_get_clk_freq_id function data type
by Ricardo Rivera-Matos 20 Sep '24

20 Sep '24

Changes cs35l45_get_clk_freq_id() function data type from unsigned int to int. This function is returns a positive index value if successful or a negative error code if unsuccessful. Functionally there should be no difference as long as the unsigned int return is interpreted as an int, however it should be corrected for readability. Signed-off-by: Ricardo Rivera-Matos <rriveram(a)opensource.cirrus.com> --- sound/soc/codecs/cs35l45-tables.c | 2 +- sound/soc/codecs/cs35l45.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l45-tables.c b/sound/soc/codecs/cs35l45-tables.c index e1cebb9e4dc6..405dab137b3b 100644 --- a/sound/soc/codecs/cs35l45-tables.c +++ b/sound/soc/codecs/cs35l45-tables.c @@ -315,7 +315,7 @@ static const struct { { 0x3B, 24576000 }, }; -unsigned int cs35l45_get_clk_freq_id(unsigned int freq) +int cs35l45_get_clk_freq_id(unsigned int freq) { int i; diff --git a/sound/soc/codecs/cs35l45.h b/sound/soc/codecs/cs35l45.h index e2ebcf58d7e0..7a790d2acac7 100644 --- a/sound/soc/codecs/cs35l45.h +++ b/sound/soc/codecs/cs35l45.h @@ -507,7 +507,7 @@ extern const struct dev_pm_ops cs35l45_pm_ops; extern const struct regmap_config cs35l45_i2c_regmap; extern const struct regmap_config cs35l45_spi_regmap; int cs35l45_apply_patch(struct cs35l45_private *cs35l45); -unsigned int cs35l45_get_clk_freq_id(unsigned int freq); +int cs35l45_get_clk_freq_id(unsigned int freq); int cs35l45_probe(struct cs35l45_private *cs35l45); void cs35l45_remove(struct cs35l45_private *cs35l45); -- 2.43.0

3 2

Handling values
by GitHub issues - opened 19 Sep '24

19 Sep '24

alsa-project/alsa-ucm-conf issue #442 was opened from thenameisluk: i have similar config to https://github.com/alsa-project/alsa-ucm-conf/blob/master/ucm2/Qualcomm/sc7… for [homestar](https://wiki.postmarketos.org/wiki/Lenovo_IdeaPad_Duet_5_(google-… ``` # Use case configuration for ALC5682+MAX98357A on SC7180 SectionVerb { Value { TQ "HiFi" } } SectionDevice."Speaker" { Comment "Speaker" Value { PlaybackChannels 4 PlaybackPCM "hw:${CardId},1" PlaybackPriority 100 PlaybackMixerElem "Speaker" PlaybackVolume "Speaker Playback Volume" } } SectionDevice."Mic" { Comment "Internal Microphone" EnableSequence [ cset "name='IF1 01 ADC Swap Mux' 1" cset "name='Stereo1 ADC L2 Mux' 1" cset "name='Stereo1 ADC R2 Mux' 1" cset "name='Stereo1 ADC MIXL ADC2 Switch' 1" cset "name='Stereo1 ADC MIXR ADC2 Switch' 1" ] DisableSequence [ cset "name='Stereo1 ADC L2 Mux' 0" cset "name='Stereo1 ADC R2 Mux' 0" cset "name='Stereo1 ADC MIXL ADC2 Switch' 0" cset "name='Stereo1 ADC MIXR ADC2 Switch' 0" ] Value { CapturePCM "hw:${CardId},0" CapturePriority 100 CaptureMixerElem "STO1 ADC" CaptureVolume "STO1 ADC Capture Volume" } } SectionDevice."HDMI" { Comment "HDMI Jack" ConflictingDevice [ "Speaker" ] EnableSequence [ ] DisableSequence [ ] Value { PlaybackPriority 200 PlaybackPCM "hw:${CardId},2" JackControl "HDMI Jack" } } ``` the speaker and mic section is irrelevant here hdmi is causing issues without ``` ConflictingDevice [ "Speaker" ] ``` pipewire doesn't want to use the ucm at all unless the hdmi is plugged in with it pipewire will only configure hdmi if hdmi is plugged in (when you restart it or at boot) it's all on the same card and the interface never goes away, only switches between ``` # amixer -c 0 contents numid=9,iface=CARD,name='HDMI Jack' ; type=BOOLEAN,access=r-------,values=1 : values=off ... ``` without hdmi and ``` # amixer -c 0 contents numid=9,iface=CARD,name='HDMI Jack' ; type=BOOLEAN,access=r-------,values=1 : values=on ``` with hdmi ```sudo alsactl monitor```also shows ``` node hw:0, #9 (0,0,0,HDMI Jack,0) VALUE ``` when plugging in and unplugging knowing all this i am unable to figure out how to handle it with ucm2 and would appreciate some help getting it working without that ConflictingDevice Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/442 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ALSA: hda/tas2781: Add new quirk for Lenovo Y990 Laptop
by Baojun Xu 19 Sep '24

19 Sep '24

Add new vendor_id and subsystem_id in quirk for Lenovo Y990 Laptop. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 480e82df7a4c..4f69fd30fdce 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10612,6 +10612,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38cd, "Y790 VECO DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38d2, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), SND_PCI_QUIRK(0x17aa, 0x38d7, "Lenovo Yoga 9 14IMH9", ALC287_FIXUP_YOGA9_14IMH9_BASS_SPK_PIN), + SND_PCI_QUIRK(0x17aa, 0x38df, "Y990 YG DUAL", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38f9, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38fa, "Thinkbook 16P Gen5", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), -- 2.43.0

2 1

[PATCH] ASoC: amd: acp: don't set card long_name
by Vijendar Mukunda 18 Sep '24

18 Sep '24

UCM can load a board-specific file based on the card long_name. Remove the constant "AMD Soundwire SOF" long_name so that the ASoC core can set the long_name based on DMI information. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-sdw-sof-mach.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 6c50c8276538..306854fb08e3 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -400,9 +400,6 @@ static int sof_card_dai_links_create(struct snd_soc_card *card) return ret; } -/* SoC card */ -static const char sdw_card_long_name[] = "AMD Soundwire SOF"; - static int mc_probe(struct platform_device *pdev) { struct snd_soc_acpi_mach *mach = dev_get_platdata(&pdev->dev); @@ -463,8 +460,6 @@ static int mc_probe(struct platform_device *pdev) if (!card->components) return -ENOMEM; - card->long_name = sdw_card_long_name; - /* Register the card */ ret = devm_snd_soc_register_card(card->dev, card); if (ret) { -- 2.34.1

3 3

Re: [PATCH] [RFC] ASoC: Conditional PCM support
by Pierre-Louis Bossart 17 Sep '24

17 Sep '24

On 8/21/24 12:18, Cezary Rojewski wrote: > Conditional PCM (condpcm) helps facilitate modern audio usecases such as > Echo Cancellations and Noise Reduction. These are not invoked by the > means of userspace application opening an endpoint (FrontEnd) but are a > "side effect" of selected PCMs running simultaneously e.g.: if both > Speaker (source) and Microphone Array (sink) are running, reference > data from the Speaker and take it into account when processing capture > for better voice command detection ratio. The point about dependencies between capture/playback usages is certainly valid, and we've faced it multiple times for SOF - and even before in the mobile phone days. I am not convinced however that the graph management suggested here solves the well-known DPCM routing problems? See notes in no specific order below. I am not following what the 'source' and 'sink' concepts refer to in this context. It looks like you are referring to regular PCM devices, i.e. Front Ends in soc-pcm parlance but examples and code make references to Back Ends. There are also complicated cases where the amplifiers can provide an echo reference for AEC and I/V sensing for speaker protection. You would want to capture both even if there's no capture happening at the userspace level. This is a well-know DPCM routing issue where we have to rely on a Front-End being opened and some tags in UCM to deal with loose coupling. It would help if you added precisions on your assumptions of where the processing takes place. In some cases Echo cancellation is handled in userspace, others in SOC firmware and others externally in a codec. The notion of source/sink is also problematic when the same BE provides two sources of information that will be split, again same problem with amplifier feedback being used for two separate functions. What happens if you have multiple sinks for one source? Same for the cases where the mic input is split multiple ways with different processing added on different PCM capture devices, e.g. for WebRTC there's an ask for a raw input, an AEC-processed input and AEC+NS-processed input. That's typically implemented with two splitters, the echo reference would be used by an intermediate node inside a firmware graph, not at the DAI/BE or host/FE levels, and such intermediate nodes are usually not handled by soc-pcm. We really need more than the notion of FE and BE, a two-layer solution is no longer sufficient. The other thing that looks weird is the dependency on both sink and source sharing a common state. For a noise reduction there are cases where you'd want the mic input to be stored in a history buffer so that the noise parameters can be estimated as soon as the actual capture starts. > Which PCMs are needed for given conditional PCM to be spawned is > determinated by the driver when registering the condpcm. Presumably such links should be described by a topology file? It would be odd for a driver to have to guess when to connect processing elements. > The functionality was initially proposed for the avs-driver [1] and, > depending on feedback and review may either go back into avs -or- become > a ASoC-core feature. Implementation present here is an example of how > such functionality could look and work on the ASoC side. Compared to > what was provided initially, the patch carries simplified version of the > feature: no priority/overriding for already running conditional PCMs. > Whatever is spawned is treated as a non-conflicting entity. > > Assumptions and design decisions: > > - existence and outcome of condpcm operations is entirely optional and > shall not impact the runtime flow of PCMs that spawned given condpcm, > e.g.: fail in cpcm->hw_params() shall not impact fe->hw_params() or > be->hw_params() negatively. Think of it as of debugfs. Useful? Yes. > Required for system to operate? No. that's debatable, if the AEC setup isn't successful then is the functionality implemented correctly? My take is no, don't fail silently if the AEC doesn't work. If this functionality is listed as a product requirement then it cannot be treated as a debugfs optional thing. Exhibit A for this is the countless cases where validation reported a problem with a path remaining active or conversely not being setup, or a voice quality issue. Those are not optional... > - a condpcm is a runtime entity that's audio format independent - since > certain FE/BEs are its dependencies already, that's no need to do > format ruling twice. Driver may still do custom checks thanks to > ->match() operation. > > - a condpcm allows for additional processing of data that flows from > data-source - a substream instance acting as data provider - > to sink - a substream acting as data consumer. At the same time, > regardless of substream->stream, given substream may act as data > source for one condpcm and data sink for another, simultaneously. > > - while condpcm's behaviour mimics standard PCM one, there is no > ->open() and ->close() - FE/BEs are treated as operational starting > with successful ->hw_params(), when hw_ruling is done and hardware is > configured. > > - cpcm->prepare() gets called only when both data source and sink are > prepared > - cpcm->trigger(START) gets called only when both data source and sink > are running > - cpcm->trigger(STOP) gets called when either data source or sink is > stopped > > Simplified state machine: > > | > register_condpcm() | > v > +--+-------------+ > | DISCONNECTED |<-+ > +--+-------------+ | > | | > condpcm_hw_params() | | > v | > +--+-------------+ | > | SETUP | | condpcm_hw_free() > +--+-------------+ | > | | > condpcm_prepare() | | > v | > +--+----+--------+ | > | PREPARED |--+ > +--+----------+--+ > | ^ > condpcm_start() | | condpcm_stop() > v | > +--+----------+--+ > | RUNNING | > +----------------+ > > What's missing? > I've not covered the locking part yet. While some operations are covered > by default thanks to snd_soc_dpcm_mutex(), it is insufficient. > If feature goes back to the avs-driver, then I'm set due to path->mutex. > > The locking is one of the reasons I'm leaning towards leaving the > condpcm within the avs-driver. For soc_condpcm_find_match() to be > precise and do no harm, a lock must prepend the list_for_each_entry(). > Entries (substreams) of that list may be part of number of different > components and the search may negatively impact runtime flow of > substreams that do not care about condpcms at all. > > Has this been tested? > > Unit-like only. Typical case below with avs_condpcm_ops representing > bunch of stubs with printfs. > > static struct snd_soc_condpcm_pred pred1 = { > .card_name = "ssp0-loopback", > .link_name = "SSP0-Codec", /* BE */ > .direction = SNDRV_PCM_STREAM_PLAYBACK, > }; > > static struct snd_soc_condpcm_pred pred2 = { > .card_name = "hdaudioB0D2", > .link_name = "HDMI1", /* FE */ > .direction = SNDRV_PCM_STREAM_PLAYBACK, > }; It's not intuitive to follow what HDMI and SSP might have to do with each other, nor why one is a BE and one is an FE? If I follow the code below, the SSP loopback is a source feeds into an HDMI sink, and SSP is a BE and HDMI an FE? Confusing example... > static void avs_condpcms_register(struct avs_dev *adev) > { > (...) > snd_soc_register_condpcm(&pred1, &pred2, &avs_condpcm_ops, adev); > } > > [1]: https://lore.kernel.org/alsa-devel/20211208111301.1817725-21-cezary.rojewsk… > > Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> > --- > include/sound/pcm.h | 1 + > include/sound/soc.h | 65 ++++++++ > sound/core/pcm.c | 1 + > sound/soc/Makefile | 2 +- > sound/soc/soc-condpcm.c | 348 ++++++++++++++++++++++++++++++++++++++++ > sound/soc/soc-condpcm.h | 17 ++ > sound/soc/soc-core.c | 2 + > sound/soc/soc-pcm.c | 11 ++ > 8 files changed, 446 insertions(+), 1 deletion(-) > create mode 100644 sound/soc/soc-condpcm.c > create mode 100644 sound/soc/soc-condpcm.h > > diff --git a/include/sound/pcm.h b/include/sound/pcm.h > index ac8f3aef9205..7e635b3103a2 100644 > --- a/include/sound/pcm.h > +++ b/include/sound/pcm.h > @@ -482,6 +482,7 @@ struct snd_pcm_substream { > struct list_head link_list; /* linked list member */ > struct snd_pcm_group self_group; /* fake group for non linked substream (with substream lock inside) */ > struct snd_pcm_group *group; /* pointer to current group */ > + struct list_head cpcm_candidate_node; It wouldn't hurt to describe what 'candidate' might mean here? > /* -- assigned files -- */ > int ref_count; > atomic_t mmap_count; > diff --git a/include/sound/soc.h b/include/sound/soc.h > index e844f6afc5b5..32a6b5092192 100644 > --- a/include/sound/soc.h > +++ b/include/sound/soc.h > @@ -426,6 +426,7 @@ struct snd_jack; > struct snd_soc_card; > struct snd_soc_pcm_stream; > struct snd_soc_ops; > +struct snd_soc_condpcm; > struct snd_soc_pcm_runtime; > struct snd_soc_dai; > struct snd_soc_dai_driver; > @@ -1154,6 +1155,64 @@ struct snd_soc_card { > #define for_each_card_widgets_safe(card, w, _w) \ > list_for_each_entry_safe(w, _w, &card->widgets, list) > > +/* Conditional PCM operations called by soc-pcm.c. */ > +struct snd_soc_condpcm_ops { > + int (*match)(struct snd_soc_condpcm *, struct snd_pcm_substream *, > + struct snd_pcm_substream *); > + int (*hw_params)(struct snd_soc_condpcm *, struct snd_pcm_hw_params *); > + int (*hw_free)(struct snd_soc_condpcm *); > + int (*prepare)(struct snd_soc_condpcm *, struct snd_pcm_substream *); > + int (*trigger)(struct snd_soc_condpcm *, struct snd_pcm_substream *, int); > +}; > + > +/** > + * struct snd_soc_condpcm_pred - Predicate, describes source or sink (substream) > + * dependency for given conditional PCM. > + * > + * @card_name: Name of card owning substream to find. > + * @link_name: Name of DAI LINK owning substream to find. > + * @direction: Whether its SNDRV_PCM_STREAM_PLAYBACK or CAPTURE. > + */ > +struct snd_soc_condpcm_pred { > + const char *card_name; Please tell me the runtimes and links are in the same card... If not, there's all kinds of power management and probe/remove issues... > + const char *link_name; dai link name? > + int direction; > +}; > + > +/** > + * struct snd_soc_condpcm - Conditional PCM descriptor. > + * > + * @ops: custom PCM operations. > + * @preds: predicates for identifying source and sink for given conditional PCM. predicate is a verb and a noun, not clear what you are trying to document. > + * > + * @source: substreaming acting as a data source, assigned at runtime. > + * @sink: substreaming acting as a data sink, assigned at runtime. > + * @state: current runtime state. presumably this state is already defined that the state of sink/source? > + * > + * @source_node: list navigation for rtd->source_list. > + * @sink_node: list navigation for rtd->sink_list. > + * @node: list navigation for condpcm_list (soc-condpcm.c). > + */ > +struct snd_soc_condpcm { > + const struct snd_soc_condpcm_ops *ops; > + struct snd_soc_condpcm_pred preds[2]; > + > + struct snd_pcm_substream *source; > + struct snd_pcm_substream *sink; > + snd_pcm_state_t state; > + void *private_data; > + > + /* Condpcms navigation for the pcm runtime. */ > + struct list_head source_node; > + struct list_head sink_node; > + struct list_head node; > +}; > + > +int snd_soc_unregister_condpcm(struct snd_soc_condpcm *cpcm); > +struct snd_soc_condpcm *snd_soc_register_condpcm(struct snd_soc_condpcm_pred *source, > + struct snd_soc_condpcm_pred *sink, > + const struct snd_soc_condpcm_ops *ops, > + void *private_data); > > static inline int snd_soc_card_is_instantiated(struct snd_soc_card *card) > { > @@ -1161,6 +1220,10 @@ static inline int snd_soc_card_is_instantiated(struct snd_soc_card *card) > } > > /* SoC machine DAI configuration, glues a codec and cpu DAI together */ > +/* > + * @source_cpcm_list: List of condpcms this pcm is source for. > + * @sink_cpcm_list: List of condpcms this pcm is sink for. > + */ > struct snd_soc_pcm_runtime { > struct device *dev; > struct snd_soc_card *card; > @@ -1172,6 +1235,8 @@ struct snd_soc_pcm_runtime { > /* Dynamic PCM BE runtime data */ > struct snd_soc_dpcm_runtime dpcm[SNDRV_PCM_STREAM_LAST + 1]; > struct snd_soc_dapm_widget *c2c_widget[SNDRV_PCM_STREAM_LAST + 1]; > + struct list_head cpcm_source_list; > + struct list_head cpcm_sink_list; > > long pmdown_time; > > diff --git a/sound/core/pcm.c b/sound/core/pcm.c > index dc37f3508dc7..12243cecaa11 100644 > --- a/sound/core/pcm.c > +++ b/sound/core/pcm.c > @@ -663,6 +663,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) > substream->pstr = pstr; > substream->number = idx; > substream->stream = stream; > + INIT_LIST_HEAD(&substream->cpcm_candidate_node); > sprintf(substream->name, "subdevice #%i", idx); > substream->buffer_bytes_max = UINT_MAX; > if (prev == NULL) > diff --git a/sound/soc/Makefile b/sound/soc/Makefile > index 775bb38c2ed4..8004de7c500c 100644 > --- a/sound/soc/Makefile > +++ b/sound/soc/Makefile > @@ -1,6 +1,6 @@ > # SPDX-License-Identifier: GPL-2.0 > snd-soc-core-y := soc-core.o soc-dapm.o soc-jack.o soc-utils.o soc-dai.o soc-component.o > -snd-soc-core-y += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o > +snd-soc-core-y += soc-pcm.o soc-devres.o soc-ops.o soc-link.o soc-card.o soc-condpcm.o > snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o > > ifneq ($(CONFIG_SND_SOC_TOPOLOGY),) > diff --git a/sound/soc/soc-condpcm.c b/sound/soc/soc-condpcm.c > new file mode 100644 > index 000000000000..786b3fafd714 > --- /dev/null > +++ b/sound/soc/soc-condpcm.c > @@ -0,0 +1,348 @@ > +// SPDX-License-Identifier: GPL-2.0-only > +// > +// Conditional-PCM management > +// > +// Copyright(c) 2024 Intel Corporation > +// > +// Author: Cezary Rojewski <cezary.rojewski(a)intel.com> > +// > + > +#include <linux/export.h> > +#include <linux/list.h> > +#include <linux/slab.h> > +#include <sound/pcm.h> > +#include <sound/soc.h> > +#include "soc-condpcm.h" > + > +/* > + * condpcm_list - Stores all registered conditional pcms. > + * condpcm_candidate_list - Stores all substreams that passed hw_params() step. > + */ > +static LIST_HEAD(condpcm_list); > +static LIST_HEAD(condpcm_candidate_list); > + > +static bool dpcm_prepare_done(const struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + enum snd_soc_dpcm_state state = rtd->dpcm[substream->stream].state; > + > + return state == SND_SOC_DPCM_STATE_PREPARE || > + state == SND_SOC_DPCM_STATE_START || > + state == SND_SOC_DPCM_STATE_PAUSED; > +} > + > +static bool dpcm_hw_params_done(const struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + enum snd_soc_dpcm_state state = rtd->dpcm[substream->stream].state; > + > + return state >= SND_SOC_DPCM_STATE_HW_PARAMS && > + state < SND_SOC_DPCM_STATE_HW_FREE; > +} > + > +static bool dpcm_start_done(const struct snd_pcm_substream *substream) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + enum snd_soc_dpcm_state state = rtd->dpcm[substream->stream].state; > + > + return state == SND_SOC_DPCM_STATE_START || > + state == SND_SOC_DPCM_STATE_PAUSED; > +} > + > +static int soc_condpcm_hw_params(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_hw_params *params) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(cpcm->source); > + struct snd_soc_pcm_runtime *rtd2 = snd_soc_substream_to_rtd(cpcm->sink); how are the 'params' defined? I read above " a condpcm is a runtime entity that's audio format independent - since certain FE/BEs are its dependencies already, that's no need to do format ruling twice. " That doesn't tell us how this 'params' is determined. This is important for cases where the speaker output is e.g. 2ch 48kHz and the mic input is 4ch 96kHz. If this condpcm is not managed by any usersapce action, then what is the logic for selecting the settings in 'params'? > + int ret; > + > + ret = cpcm->ops->hw_params(cpcm, params); > + if (ret) > + return ret; > + > + list_add_tail(&cpcm->source_node, &rtd->cpcm_source_list); > + list_add_tail(&cpcm->sink_node, &rtd2->cpcm_sink_list); > + cpcm->state = SNDRV_PCM_STATE_SETUP; > + return 0; > +} There's also the well-known problem that hw_params can be called multiple times. Here this wouldn't work with the same source/sink added multiple times in a list. > + > +static void soc_condpcm_hw_free(struct snd_soc_condpcm *cpcm) > +{ > + cpcm->ops->hw_free(cpcm); > + list_del(&cpcm->source_node); > + list_del(&cpcm->sink_node); > + cpcm->state = SNDRV_PCM_STATE_DISCONNECTED; > +} > + > +static void soc_condpcm_prepare(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream) > +{ > + int ret; > + > + ret = cpcm->ops->prepare(cpcm, substream); > + if (!ret) > + cpcm->state = SNDRV_PCM_STATE_PREPARED; > +} you probably need to look at the xruns and resume cases, where prepare() is used for vastly different purposes. > + > +static void soc_condpcm_start(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream, int cmd) > +{ > + int ret; > + > + ret = cpcm->ops->trigger(cpcm, substream, cmd); > + if (ret) > + cpcm->state = SNDRV_PCM_STATE_SETUP; > + else > + cpcm->state = SNDRV_PCM_STATE_RUNNING; > +} > + > +static void soc_condpcm_stop(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream, int cmd) > +{ > + int ret; > + > + ret = cpcm->ops->trigger(cpcm, substream, cmd); > + if (ret || cmd != SNDRV_PCM_TRIGGER_PAUSE_PUSH) > + cpcm->state = SNDRV_PCM_STATE_SETUP; > + else > + cpcm->state = SNDRV_PCM_STATE_PREPARED; > +} > + > +void snd_soc_condpcms_prepare(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream) > +{ > + struct snd_soc_condpcm *cpcm; > + > + /* Prepare conditional pcms only if source and sink are both preprared. */ > + list_for_each_entry(cpcm, &rtd->cpcm_source_list, source_node) { > + if (cpcm->state == SNDRV_PCM_STATE_SETUP && > + dpcm_prepare_done(cpcm->sink)) > + soc_condpcm_prepare(cpcm, substream); > + } > + > + list_for_each_entry(cpcm, &rtd->cpcm_sink_list, sink_node) { > + if (cpcm->state == SNDRV_PCM_STATE_SETUP && > + dpcm_prepare_done(cpcm->source)) > + soc_condpcm_prepare(cpcm, substream); > + } > +} > + > +static void soc_condpcms_start(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, int cmd) > +{ > + struct snd_soc_condpcm *cpcm; > + > + /* Start conditional pcms only if source and sink are both running. */ > + list_for_each_entry(cpcm, &rtd->cpcm_source_list, source_node) { > + if (cpcm->state == SNDRV_PCM_STATE_PREPARED && > + dpcm_start_done(cpcm->sink)) > + soc_condpcm_start(cpcm, substream, cmd); > + } > + > + list_for_each_entry(cpcm, &rtd->cpcm_sink_list, sink_node) { > + if (cpcm->state == SNDRV_PCM_STATE_PREPARED && > + dpcm_start_done(cpcm->source)) > + soc_condpcm_start(cpcm, substream, cmd); > + } > +} > + > +static void soc_condpcms_stop(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, int cmd) > +{ > + struct snd_soc_condpcm *cpcm; > + > + /* Stop conditional pcms if either source or sink is not running. */ > + list_for_each_entry(cpcm, &rtd->cpcm_source_list, source_node) > + if (cpcm->state == SNDRV_PCM_STATE_RUNNING) > + soc_condpcm_stop(cpcm, substream, cmd); > + > + list_for_each_entry(cpcm, &rtd->cpcm_sink_list, sink_node) > + if (cpcm->state == SNDRV_PCM_STATE_RUNNING) > + soc_condpcm_stop(cpcm, substream, cmd); > +} > + > +void snd_soc_condpcms_trigger(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, int cmd) > +{ > + switch (cmd) { > + case SNDRV_PCM_TRIGGER_START: > + case SNDRV_PCM_TRIGGER_RESUME: > + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: > + soc_condpcms_start(rtd, substream, cmd); > + break; > + case SNDRV_PCM_TRIGGER_STOP: > + case SNDRV_PCM_TRIGGER_SUSPEND: > + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: > + soc_condpcms_stop(rtd, substream, cmd); > + break; > + } > +} > + > +void snd_soc_condpcms_teardown(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream) > +{ > + struct snd_soc_condpcm *cpcm, *save; > + > + list_del(&substream->cpcm_candidate_node); > + > + /* Free all condpcms this substream spawned. */ > + list_for_each_entry_safe(cpcm, save, &rtd->cpcm_source_list, source_node) > + soc_condpcm_hw_free(cpcm); > + list_for_each_entry_safe(cpcm, save, &rtd->cpcm_sink_list, sink_node) > + soc_condpcm_hw_free(cpcm); > +} > + > +static bool soc_condpcm_test(struct snd_soc_condpcm *cpcm, > + struct snd_pcm_substream *substream, int dir) > +{ > + struct snd_soc_pcm_runtime *rtd = snd_soc_substream_to_rtd(substream); > + struct snd_soc_condpcm_pred *pred = &cpcm->preds[dir]; > + > + return pred->direction == substream->stream && > + !strcmp(pred->card_name, rtd->card->name) && > + !strcmp(pred->link_name, rtd->dai_link->name); > +} > + > +static struct snd_pcm_substream * > +soc_condpcm_find_match(struct snd_soc_condpcm *cpcm, struct snd_pcm_substream *ignore, int dir) > +{ > + struct snd_pcm_substream *substream; > + > + list_for_each_entry(substream, &condpcm_candidate_list, cpcm_candidate_node) { > + if (substream == ignore) > + continue; > + > + /* Only substreams that passed hw_params() are valid candidates. */ > + if (!dpcm_hw_params_done(substream)) > + continue; > + > + if (soc_condpcm_test(cpcm, substream, dir)) > + return substream; > + } > + > + return NULL; > +} > + > +static int soc_condpcm_walk(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params, int dir) > +{ > + /* Temporary source/sink cache. */ > + struct snd_pcm_substream *substreams[2]; > + struct snd_soc_condpcm *cpcm; > + int ret; > + > + substreams[dir] = substream; > + > + list_for_each_entry(cpcm, &condpcm_list, node) { > + if (cpcm->state != SNDRV_PCM_STATE_DISCONNECTED) > + continue; > + > + /* Does this cpcm match @substream? */ > + if (!soc_condpcm_test(cpcm, substream, dir)) > + continue; > + > + /* Find pair for the @substream. */ > + substreams[!dir] = soc_condpcm_find_match(cpcm, substream, !dir); > + if (!substreams[!dir]) > + continue; > + > + /* Allow driver to have the final word. */ > + ret = cpcm->ops->match(cpcm, substreams[0], substreams[1]); > + if (ret) > + continue; > + cpcm->source = substreams[0]; > + cpcm->sink = substreams[1]; > + > + ret = soc_condpcm_hw_params(cpcm, params); > + if (ret) { > + cpcm->source = NULL; > + cpcm->sink = NULL; > + return ret; > + } > + } > + > + return 0; > +} > + > +/* Called by soc-pcm.c after each successful hw_params(). */ > +int snd_soc_condpcms_walk_all(struct snd_soc_pcm_runtime *rtd, > + struct snd_pcm_substream *substream, > + struct snd_pcm_hw_params *params) > +{ > + int ret; > + > + list_add_tail(&substream->cpcm_candidate_node, &condpcm_candidate_list); > + > + /* Spawn all condpcms this substream is the missing source of. */ > + ret = soc_condpcm_walk(rtd, substream, params, SNDRV_PCM_STREAM_CAPTURE); > + if (ret) > + return ret; > + > + /* Spawn all condpcms this substream is the missing sink of. */ > + return soc_condpcm_walk(rtd, substream, params, SNDRV_PCM_STREAM_PLAYBACK); > +} Are loops supported? Is the order between capture and playback intentional? Is the notion of playback/capture even relevant when trying to add loopbacks? Lots of questions...

4 7

[PATCH 000/113] ALSA: add snd_pcm_is_playback/capture() macro
by Kuninori Morimoto 17 Sep '24

17 Sep '24

Hi ALSA-ML Cc Staging-ML, USB-ML Many drivers are using below code to know the Sound direction. if (direction == SNDRV_PCM_STREAM_PLAYBACK) This patch-set add snd_pcm_is_playback/capture() macro to handle it. Thank you for your help !! Best regards --- Kuninori Morimoto

4 119

[PATCH 0/2] Updates for Atmel SSC DAI
by Andrei Simion 16 Sep '24

16 Sep '24

This patch set includes two updates for the Atmel SSC DAI driver: - Address the limitation with the S24_LE format. - Add stream names for DPCM and future use-cases. Codrin Ciubotariu (2): ASoC: atmel: atmel_ssc_dai: Add stream names ASoC: atmel: atmel_ssc_dai: Drop S24_LE support due to single channel limitation sound/soc/atmel/atmel_ssc_dai.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) base-commit: 5acd9952f95fb4b7da6d09a3be39195a80845eb6 -- 2.34.1

2 6

[PATCH stable-6.10 regression] Revert "soundwire: stream: fix programming slave ports for non-continous port maps"
by Peter Ujfalusi 16 Sep '24

16 Sep '24

The prop->src_dpn_prop and prop.sink_dpn_prop is allocated for the _number_ of ports and it is forced as 0 index based. The original code was correct while the change to walk the bits and use their position as index into the arrays is not correct. For exmple we can have the prop.source_ports=0x2, which means we have one port, but the prop.src_dpn_prop[1] is accessing outside of the allocated memory. This reverts commit 6fa78e9c41471fe43052cd6feba6eae1b0277ae3. Cc: stable(a)vger.kernel.org # 6.10.y Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- Hi, The reverted patch causes major regression on soundwire causing all audio to fail. Interestingly the patch is only in 6.10.8 and 6.10.9, not in mainline or linux-next. soundwire sdw-master-0-1: Program transport params failed: -22 soundwire sdw-master-0-1: Program params failed: -22 SDW1-Playback: ASoC: error at snd_soc_link_prepare on SDW1-Playback: -22 Regards, Peter drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 00191b1d2260..4e9e7d2a942d 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1286,18 +1286,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - unsigned long mask; + u8 num_ports; int i; if (direction == SDW_DATA_DIR_TX) { - mask = slave->prop.source_ports; + num_ports = hweight32(slave->prop.source_ports); dpn_prop = slave->prop.src_dpn_prop; } else { - mask = slave->prop.sink_ports; + num_ports = hweight32(slave->prop.sink_ports); dpn_prop = slave->prop.sink_dpn_prop; } - for_each_set_bit(i, &mask, 32) { + for (i = 0; i < num_ports; i++) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.46.0

4 8

[GIT PULL] ASoC updates for v6.12
by Mark Brown 16 Sep '24

16 Sep '24

The following changes since commit 839a4ec06f75cec8fec2cc5fc14e921d0c3f7369: ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict (2024-08-23 11:02:53 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.12 for you to fetch changes up to 2772ee6de6cf94e5f2a0c0ce6067d0796a4170ba: ASoC: topology: Fix redundant logical jump (2024-09-13 19:40:38 +0100) ---------------------------------------------------------------- ASoC: Updates for v6.12 This is a very large set of changes, almost all in drivers rather than the core. Even with the addition of several quite large drivers the overall diffstat is negative thanks to the removal of some old Intel board support which has been obsoleted by the AVS driver, helped a bit by some factoring out into helpers (especially around the Soundwire machine drivers for x86). Highlights include: - More simplifications and cleanups throughout the subsystem from Morimoto-san. - Extensive cleanups and refactoring of the Soundwire drivers to make better use of helpers. - Removal of Intel machine support obsoleted by the AVS driver. - Lots of DT schema conversions. - Machine support for many AMD and Intel x86 platforms. - Support for AMD ACP 7.1, Mediatek MT6367 and MT8365, Realtek RTL1320 SoundWire and rev C, and Texas Instruments TAS2563 ---------------------------------------------------------------- Alexandre Mergnat (11): ASoC: dt-bindings: mediatek,mt8365-afe: Add audio afe document ASoC: dt-bindings: mediatek,mt8365-mt6357: Add audio sound card document dt-bindings: mfd: mediatek: Add codec property for MT6357 PMIC ASoC: mediatek: mt8365: Add common header ASoC: mediatek: mt8365: Add audio clock control support ASoC: mediatek: mt8365: Add I2S DAI support ASoC: mediatek: mt8365: Add ADDA DAI support ASoC: mediatek: mt8365: Add DMIC DAI support ASoC: mediatek: mt8365: Add PCM DAI support ASoC: mediatek: mt8365: Add the AFE driver support ASoC: mediatek: Add MT8365 support Andrei Simion (1): ASoC: dt-bindings: microchip,sama7g5-spdifrx: Add common DAI reference Andrew Kreimer (1): ASoC: tlv320aic31xx: Fix typos Animesh Agarwal (2): ASoC: dt-bindings: ti,pcm512x: Convert to dtschema ASoC: dt-bindings: fsl,imx-audio-es8328: Convert to dtschema Arnd Bergmann (2): ASoC: mediatek: mt8365: include linux/bitfield.h ASoC: mediatek: mt8365: remove unused mt8365_i2s_hd_str Balamurugan C (4): ASoC: Intel: soc-acpi: Add entry for sof_es8336 in ARL match table. ASoC: Intel: soc-acpi: Add entry for HDMI_In capture support in ARL match table ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for ARL. ASoC: Intel: ARL: Add entry for HDMI-In capture support to non-I2S codec boards. Baojun Xu (1): ASoC: tas2781: Remove unnecessary line feed for tasdevice_codec_remove Bard Liao (6): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support ASoC: Intel: sof_sdw: add rt1320 amp support ASoC: SOF: Intel: hda: refactoring topology name fixup for HDA mach ASoC: Intel: sof_sdw: move ignore_internal_dmic check earlier ASoC: Intel: sof_sdw: overwrite mach_params->dmic_num Biju Das (2): ASoC: dt-bindings: renesas,rz-ssi: Document port property ASoC: sh: rz-ssi: Add full duplex support Binbin Zhou (7): ASoC: loongson: Use BIT() macro ASoC: loongson: Simplify probe() with local dev variable ASoC: loongson: Simplify with dev_err_probe() ASoC: loongson: Simplify if statment in loongson_card_hw_params() ASoC: loongson: Replace if with ternary operator ASoC: loongson: Factor out loongson_card_acpi_find_device() function ASoC: loongson: Factor out loongson i2s enable clock functions Brent Lu (10): ASoC: SOF: Intel: hda: refactoring topology name fixup for SDW mach ASoC: SOF: Intel: hda: support BT link mask in mach_params ASoC: Intel: skl_hda_dsp_generic: support BT audio offload ASoC: Intel: skl_hda_dsp_generic: remove hdac-hdmi support ASoC: Intel: skl_hda_dsp_generic: use sof_hdmi_private to init HDMI ASoC: Intel: board_helpers: support HDA link initialization ASoC: Intel: skl_hda_dsp_generic: use common module for DAI links ASoC: Intel: ehl_rt5660: do not check common_hdmi_codec_drv ASoC: Intel: sof_pcm512x: do not check common_hdmi_codec_drv ASoC: SOF: Intel: hda: remove common_hdmi_codec_drv Cezary Rojewski (14): ALSA: hda: Move SST device entries to AVS ASoC: Intel: Drop skl_machine_pdata usage ASoC: Intel: Remove bxt_rt298 board driver ASoC: Intel: Remove bxt_da7219_max98357a board driver ASoC: Intel: Remove kbl_rt5663_rt5514_max98927 board driver ASoC: Intel: Remove kbl_rt5663_max98927 board driver ASoC: Intel: Remove kbl_rt5660 board driver ASoC: Intel: Remove kbl_da7219_max98927 board driver ASoC: Intel: Remove kbl_da7219_max98357a board driver ASoC: Intel: Remove skl_rt286 board driver ASoC: Intel: Remove skl_nau88l25_ssm4567 board driver ASoC: Intel: Remove skl_nau88l25_max98357a board driver ASoC: Intel: Remove skylake driver ASoC: Intel: avs: Enable by default for all SST configurations Charles Keepax (3): ASoC: Intel: soc-acpi: arl: Add match entries for new cs42l43 laptops ASoC: Intel: soc-acpi: adl: Add match entries for new cs42l43 laptops ASoC: Intel: soc-acpi: lnl: Add match entries for new cs42l43 laptops Chen Ni (4): ASoC: mediatek: mt8192: remove redundant null pointer check before of_node_put ASoC: cs42l42: Convert comma to semicolon ASoC: topology-test: Convert comma to semicolon ASoC: Intel: skl_hda_dsp_generic: convert comma to semicolon Christophe JAILLET (4): ASoC: sti-sas: Constify snd_soc_component_driver struct ASoC: cs43130: Constify snd_soc_component_driver struct ASoC: cs43130: Constify struct reg_sequence and reg_sequences ASoC: rt1318: Constify struct reg_sequence Codrin Ciubotariu (5): ASoC: atmel: mchp-i2s-mcc: Improve maxburst calculation for better performance ASoC: atmel: mchp-i2s-mcc: Remove interface name from stream_name ASoC: atmel: mchp-pdmc: Improve maxburst calculation for better performance ASoC: atmel: mchp-pdmc: Add snd_soc_dai_driver name ASoC: atmel: mchp-pdmc: Retain Non-Runtime Controls Colin Ian King (1): ASoC: codecs: wcd934x: make read-only array minCode_param static const Dmitry Torokhov (1): ASoC: tas5086: use sleeping variants of gpiod API Fabio Estevam (7): ASoC: fsl_audmix: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() ASoC: fsl_mqs: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() ASoC: fsl_rpmsg: Switch to RUNTIME_PM_OPS() ASoC: fsl_spdif: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() ASoC: fsl_ssi: Switch to SYSTEM_SLEEP_PM_OPS ASoC: imx-audmux: Switch to SYSTEM_SLEEP_PM_OPS ASoC: imx-pcm-rpmsg: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() Frank Li (2): ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml ASoC: dt-bindings: Convert tpa6130a2.txt to yaml Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Geert Uytterhoeven (1): ASoC: dt-bindings: dlg,da7213: Convert to json-schema Gustavo A. R. Silva (1): ASoC: SOF: sof-audio: Avoid -Wflex-array-member-not-at-end warnings Heiko Stuebner (2): ASoC: dt-bindings: realtek,rt5616: document mclk clock ASoC: dt-bindings: realtek,rt5616: Document audio graph port Hongbo Li (2): ASoC: dapm: Use IS_ERR_OR_NULL() helper function ASoC: adi: Use str_enabled_disabled() helper Jens Reidel (1): ASoC: qcom: sm8250: enable primary mi2s Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Krzysztof Kozlowski (16): ASoC: codecs: wsa881x: Drop unused version readout ASoC: codecs: wsa883x: Handle reading version failure ASoC: codecs: wsa883x: Simplify handling variant/version ASoC: codecs: wsa884x: Simplify handling variant ASoC: dt-bindings: qcom,apq8016-sbc-sndcard: move to separate binding ASoC: codecs: lpass-wsa-macro: Do not hard-code dai in VI mixer ASoC: codecs: wsa881x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa883x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa884x: Use designator array initializers for Soundwire ports ASoC: codecs: wcd938x: Drop unused defines and enums ASoC: codecs: wcd937x: Move max port number defines to enum ASoC: codecs: wcd938x: Move max port number defines to enum ASoC: codecs: wcd939x: Move max port number defines to enum ASoC: dt-bindings: samsung,odroid: drop stale clocks ASoC: MAINTAINERS: Drop incorrect tlv320aic31xx.txt path ASoC: codecs: wsa884x: Implement temperature reading and hwmon Kuninori Morimoto (19): ASoC: rsnd: remove rsnd_mod_confirm_ssi() under DEBUG ASoC: rsnd: use pcm_dmaengine code ASoC: stm: use snd_pcm_direction_name() ASoC: sof: pcm: use snd_pcm_direction_name() ASoC: sof: intel: use snd_pcm_direction_name() ASoC: fsl: lpc3xxx-i2s: use snd_pcm_direction_name() ASoC: tegra: use snd_pcm_direction_name() ASoC: soc-pcm: use snd_pcm_direction_name() ASoC: soc-dapm: use snd_pcm_direction_name() ASoC: remove bespoke trigger support ASoC: soc-pcm: remove snd_soc_dpcm_stream_{lock/unlock}_irq() ASoC: soc-pcm: remove snd_soc_dpcm_stream_lock_irqsave_nested() ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() ASoC: Intel: sof_sdw: use playback/capture_only flags ASoC: soc-pcm: move snd_soc_dpcm_can_be_xxx() to top ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function ASoC: makes rtd->initialized bit field ASoC: dt-bindings: renesas,rsnd: add post-init-providers property Linus Walleij (3): ASoC: tas2781-i2c: Drop weird GPIO code ASoC: tas2781-i2c: Get the right GPIO line ASoC: tas*: Drop unused GPIO includes Liu Jing (1): ASoC: mediatek: mt2701-cs42448: Optimize redundant code in mt2701_cs42448_machine_probe Luca Ceresoli (3): ASoC: dapm-graph: remove the "ROOT" cluster ASoC: dapm-graph: visualize component On/Off bias level ASoC: dapm-graph: show path name for non-static routes Lukasz Majewski (1): ASoC: dt-bindings: Convert mxs-saif.txt to fsl,saif.yaml (imx28 saif) Ma Ke (2): ASoC: rt5682s: Return devm_of_clk_add_hw_provider to transfer the error ASoC: rt5682: Return devm_of_clk_add_hw_provider to transfer the error Maciej Strozek (1): ASoC: Intel: sof_sdw: Add quirks from some new Dell laptops Marek Vasut (1): ASoC: tlv320aic32x4: Add multi endpoint support Mark Brown (37): Add HDMI Audio support ASoC: codecs: wsa88xx: Few cleanups ASoC: codecs: wsa88xx and wcd93xx: Soundwire port AMD SOF based generic SoundWire machine driver ALSA/ASoC: use snd_pcm_direction_name() ASoC: SOF: reshuffle and optimize structures ASoC/SOF/PCI/Intel: add PantherLake support ASoC: Merge fixes ASoC: tas*: Fix up GPIO usage ASoC: Intel: Remove skylake driver ALSA/ASoC/SoundWire: Intel: update maximum number ASoC: Merge up fixes ASoC: grace time for DPCM cleanup ASoC: Merge up fixes ASoC: dapm-graph: add component on/off and route names Add SOF support for ACP7.0 based platform ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local ASoC: Intel: boards: updates for 6.12 ASoC: dt-bindings: amlogic-sound-cards: document Add audio support for the MediaTek Genio 350-evk Fixes for the audio setup on the rk3588-nanopc-t6 Add i2s/tdm support for acp7.0 and acp7.1 platforms Some clean up with helper fucntion Add audio support for the MediaTek Genio 350-evk ASoC: mt8365: Open code BIT() to avoid spurious warnings ASoC: mt8365: Remove spurious unsigned long casts ASoC: mt8365: Remove unused prototype for mt8365_afe_clk_group_48k() ASoC: mt8365: Make non-exported functions static ASoC: mt8365: Remove unused variables ASoC: mt8365: Remove unused DMIC IIR coefficient configuration ASoC: mt8365: Allow build coverage ASoC: loongson: Simplify code formatting ASoC: mt8365: Fix -Werror builds Add support for primary mi2s on SM8250 ASoC: Improvements for mchp-pdmc ASoC: Intel: boards: updates for 6.12 - part 2 AMD SoundWire machine driver code refactor Muhammad Usama Anjum (3): ASoC: amd: acp: Return in-case of error ASoC: mediatek: mt8365: check validity before usage of i2s_data ASoc: mediatek: mt8365: Remove unneeded assignment Neil Armstrong (2): ASoC: dt-bindings: amlogic,axg-sound-card: document clocks property ASoC: dt-bindings: amlogic,gx-sound-card: document clocks property Nicolas Belin (2): ASoc: mediatek: mt8365: Add a specific soundcard for EVK ASoC: codecs: add MT6357 support Nikita Shubin (1): ASoC: dt-bindings: cirrus,cs4271: Convert to dtschema Pierre-Louis Bossart (11): ASoC: Intel: soc-acpi: add PTL match tables ASoC: SOF: Intel: add initial support for PTL ASoC: SOF: sof-priv.h: optimize snd_sof_platform_stream_params ASoC: SOF: sof-priv.h: optimize snd_sof_mailbox ASoC: SOF: sof-priv.h: optimize snd_sof_ipc_msg ASoC: SOF: sof-audio.h: optimize snd_sof_pcm_stream_pipeline_list ALSA/ASoC/SoundWire: Intel: use single definition for SDW_INTEL_MAX_LINKS soundwire: intel: add probe-time check on link id soundwire: intel: increase maximum number of links ASoC: Intel: sof_sdw: make sof_sdw_quirk static ASoC: Intel: boards: always check the result of acpi_dev_get_first_match_dev() Richard Fitzgerald (2): ASoC: cs35l56: Make struct regmap_config const ASoC: cs-amp-lib: Add KUnit test case for empty calibration entries Rob Herring (Arm) (2): ASoC: Use of_property_present() ASoC: Use of_property_read_bool() Romain Gantois (1): ASoC: tlv320aic31xx: Add support for loading filter coefficients Shen Lichuan (1): ASoC: SOF: topology: Use kmemdup_array instead of kmemdup for multiple allocation Shenghao Ding (7): ASoC: tas2781: Add TAS2563 into the Header ASoC: tas2781: Fix a compiling warning reported by robot kernel test due to adding tas2563_dvc_table ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781 ASoc: tas2781: Remove unnecessary line feed and space ASoC: tas2781: mark const variables tas2563_dvc_table as __maybe_unused ASoC: tas2781: replace devm_kzalloc and scnprintf with devm_kstrdup ASoC: tas2781: Add Calibration Kcontrols for Chromebook Shuming Fan (1): ASoC: rt1320: Add support for version C Simon Trimmer (1): ASoC: cs35l56: Use regmap_read_bypassed() to wake the device Srinivas Kandagatla (1): ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250 Takashi Iwai (1): ASoC: Use __counted_by() annotation for snd_soc_pcm_runtime Tang Bin (6): ASoC: loongson: remove redundant variable assignments ASoC: codecs: fix the right check and simplify code ASoC: soc-ac97: Fix the incorrect description ASoC: loongson: Add the correct judgement return ASoC: mediatek: mt7986-afe-pcm: Remove redundant error message ASoC: topology: Fix redundant logical jump Uwe Kleine-König (1): ASoC: Switch back to struct platform_driver::remove() Venkata Prasad Potturu (12): ASoC: amd: acp: Refactor TDM slots selction based on acp revision id ASoC: amd: acp: Refactor I2S dai driver ASoC: amd: acp: Update pcm hardware capabilities for acp6.3 platform ASoC: amd: acp: Add I2S TDM support for acp6.3 platform ASoC: amd: acp: Update pcm hardware capabilities for acp7.0 platform ASoC: amd: acp: Add I2S master clock generation support for acp7.0 platform ASoC: amd: acp: Set i2s clock for acp7.0 platform ASoC: amd: acp: Modify max channels and sample rate support for acp70 dai driver ASoC: amd: acp: Add I2S TDM support for acp7.0 platform ASoC: amd: acp: Add pte configuration for ACP7.0 platform ASoC: amd: acp: Add i2s master clock generation support for acp7.1 platform ASoC: amd: acp: Add I2S TDM support for acp7.1 platform Vijendar Mukunda (47): ASoC: intel: rename codec_info and dai_info structures names ASoC: intel: rename soundwire common header macros ASoC: intel: rename soundwire machine driver soc ops ASoC: intel: rename soundwire codec helper functions ASoC: intel: rename maxim codec macros ASoC: intel: rename ignore_pch_dmic variable name ASoC: intel/sdw-utils: move soundwire machine driver soc ops ASoC: intel: move soundwire machine driver common structures ASoC: intel/sdw_utils: move soundwire machine driver helper functions ASoC: intel/sdw_utils: move dmic codec helper function ASoC: intel/sdw_utils: move rtk dmic helper functions ASoC: intel/sdw_utils: move rt712 sdca helper functions ASoC: intel/sdw_utils: move rt722 sdca helper functions ASoC: intel: split soundwire machine driver private data ASoC: intel/sdw_utils: move rt5682 codec helper function ASoC: intel/sdw_utils: move rtk jack common helper functions ASoC: intel/sdw_utils: move rt700 and rt711 codec helper functions ASoC: intel/sdw_utils: move rtk amp codec helper functions ASoC: intel/sdw_utils: move cirrus soundwire codec helper functions ASoC: intel/sdw_utils: move maxim codec helper functions ASoC: intel/sdw_utils: move dai id common macros ASoC: intel/sdw_utils: move soundwire dai type macros ASoC: intel/sdw_utils: move soundwire codec_info_list structure ASoC: intel/sdw_utils: move machine driver dai link helper functions ASoC: sdw_utils: refactor sof_sdw_card_late_probe function ASoC: intel/sdw_utils: refactor init_dai_link() and init_simple_dai_link() ASoC: soc-acpi: add pci revision id field in mach params structure ASoC: amd: acp: add soundwire machines for acp6.3 based platform ASoC: SOF: amd: add alternate machines for acp6.3 based platform ASoC: SOF: amd: update mach params subsystem_rev variable ASoC: amd/sdw_utils: add sof based soundwire generic machine driver ASoC: amd: acp: remove MODULE_ALIAS for SoundWire machine driver ASoC: amd: acp: add ZSC control register programming sequence ASoC: SOF: amd: update conditional check for cache register update ASoC: amd: acp: Convert comma to semicolon ASoC: amd: acp: add legacy driver support for ACP7.1 based platforms ASoC: amd: acp: replace desc->rev check with acp pci revision id ASoC: SOF: amd: remove unused variable from sof_amd_acp_desc structure ASoC: amd: Add acpi machine id for acp7.0 version based platform ASoC: SOF: amd: add support for acp7.0 based platform ASoC: amd: acp: remove MODULE_ALIAS for legacy machine driver ASoC: amd: acp: remove MODULE_ALIAS for sof based generic machine driver ASoC: intel: sof_sdw: rename soundwire endpoint and dailink structures ASoC: intel: sof_sdw: rename soundwire parsing helper functions ASoC: sdw_util/intel: move soundwire endpoint and dai link structures ASoC: sdw_utils/intel: move soundwire endpoint parsing helper functions ASoC: amd: acp: refactor SoundWire machine driver code Wu Bo (1): ASoC: dwc: change to use devm_clk_get_enabled() helpers Yue Haibing (2): ASoC: fsl: lpc3xxx-i2s: Remove set but not used variable 'savedbitclkrate' ASoC: fsl: lpc3xxx: Make some symbols static Zhang Yi (2): ASoC: codecs: ES8326: suspend issue ASoC: codecs: ES8326: input issue after init Zhang Zekun (3): ASoC: audio-graph-card: Use for_each_child_of_node_scoped() to simplify code ASoC: audio-graph-card2: Use helper function of_get_child_count() ASoC: meson: Remove unused declartion in header file tangbin (3): ASoC: loongson: Remove useless variable definitions ASoC: loongson: remove unnecessary assignment in i2s_resume() ASoC: loongson: fix error release ying zuxin (1): ASoC: codecs: Use devm_clk_get_enabled() helpers .mailmap | 1 + .../testing/sysfs-bus-i2c-devices-turris-omnia-mcu | 6 +- Documentation/admin-guide/cifs/usage.rst | 2 +- Documentation/admin-guide/kernel-parameters.txt | 4 +- Documentation/arch/arm64/silicon-errata.rst | 18 + .../devicetree/bindings/ata/rockchip,dwc-ahci.yaml | 3 + .../bindings/display/panel/samsung,atna33xc20.yaml | 9 +- .../devicetree/bindings/mfd/mediatek,mt6357.yaml | 21 + .../bindings/sound/amlogic,axg-sound-card.yaml | 7 + .../bindings/sound/amlogic,gx-sound-card.yaml | 7 + .../devicetree/bindings/sound/cirrus,cs4271.yaml | 101 + Documentation/devicetree/bindings/sound/cs4271.txt | 57 - Documentation/devicetree/bindings/sound/da7213.txt | 45 - .../devicetree/bindings/sound/dlg,da7213.yaml | 103 + .../bindings/sound/fsl,imx-audio-es8328.yaml | 111 + .../devicetree/bindings/sound/fsl,saif.yaml | 83 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 60 - 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include/sound/tas2781.h | 71 +- include/sound/ump_convert.h | 1 + include/trace/events/btrfs.h | 8 + include/trace/events/mptcp.h | 2 +- include/uapi/asm-generic/unistd.h | 5 +- include/ufs/ufshcd.h | 1 + include/ufs/ufshci.h | 1 + init/Kconfig | 1 + io_uring/napi.c | 2 - io_uring/net.c | 7 +- io_uring/poll.c | 1 + kernel/dma/debug.c | 5 +- kernel/irq/irqdesc.c | 1 + kernel/jump_label.c | 4 +- kernel/kcov.c | 15 +- kernel/kprobes.c | 4 +- kernel/ksysfs.c | 7 + kernel/locking/lockdep.c | 6 + kernel/locking/qspinlock_paravirt.h | 2 +- kernel/module/main.c | 41 +- kernel/padata.c | 7 + kernel/profile.c | 242 +- kernel/sched/core.c | 68 +- kernel/sched/cputime.c | 6 + kernel/sched/stats.c | 10 - kernel/task_work.c | 6 + kernel/time/clocksource.c | 2 +- kernel/time/ntp.c | 9 +- kernel/time/tick-broadcast.c | 3 +- kernel/time/timekeeping.c | 2 +- kernel/trace/fgraph.c | 2 +- kernel/trace/preemptirq_delay_test.c | 2 - kernel/trace/ring_buffer.c | 12 - kernel/trace/trace.h | 23 + 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net/mac80211/util.c | 2 +- net/mptcp/mib.c | 2 + net/mptcp/mib.h | 2 + net/mptcp/options.c | 5 +- net/mptcp/pm.c | 12 + net/mptcp/pm_netlink.c | 93 +- net/mptcp/pm_userspace.c | 18 + net/mptcp/protocol.c | 18 +- net/mptcp/protocol.h | 4 + net/mptcp/subflow.c | 26 +- net/sched/act_ct.c | 4 +- net/sctp/input.c | 19 +- net/smc/af_smc.c | 7 +- net/smc/smc_stats.h | 2 +- net/sunrpc/svc.c | 2 +- net/wireless/scan.c | 11 +- net/wireless/sme.c | 1 + rust/kernel/firmware.rs | 2 +- scripts/syscall.tbl | 6 +- sound/core/seq/seq_ports.h | 14 +- sound/core/seq/seq_ump_convert.c | 132 +- sound/core/ump_convert.c | 60 +- sound/firewire/amdtp-stream.c | 38 +- sound/firewire/amdtp-stream.h | 1 + sound/hda/intel-dsp-config.c | 57 +- sound/hda/intel-sdw-acpi.c | 5 +- sound/pci/hda/cs35l41_hda.c | 13 - sound/pci/hda/cs35l56_hda.c | 38 +- sound/pci/hda/cs35l56_hda.h | 1 - sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_generic.c | 63 + sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/hda_intel.c | 10 +- sound/pci/hda/patch_conexant.c | 56 +- sound/pci/hda/patch_hdmi.c | 2 + sound/pci/hda/patch_realtek.c | 2 + sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/Kconfig | 2 + sound/soc/Makefile | 1 + sound/soc/adi/axi-i2s.c | 6 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp-config.c | 12 + sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/acp/Kconfig | 22 + sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-i2s.c | 188 +- sound/soc/amd/acp/acp-legacy-common.c | 40 +- sound/soc/amd/acp/acp-legacy-mach.c | 7 - sound/soc/amd/acp/acp-mach-common.c | 2 +- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-pci.c | 4 + sound/soc/amd/acp/acp-pdm.c | 7 +- sound/soc/amd/acp/acp-platform.c | 117 +- sound/soc/amd/acp/acp-rembrandt.c | 2 +- sound/soc/amd/acp/acp-renoir.c | 2 +- sound/soc/amd/acp/acp-sdw-sof-mach.c | 509 +++ sound/soc/amd/acp/acp-sof-mach.c | 7 - sound/soc/amd/acp/acp63.c | 2 +- sound/soc/amd/acp/acp70.c | 87 +- sound/soc/amd/acp/amd-acp63-acpi-match.c | 90 + sound/soc/amd/acp/amd.h | 27 +- sound/soc/amd/acp/chip_offset_byte.h | 84 +- sound/soc/amd/acp/soc_amd_sdw_common.h | 44 + sound/soc/amd/mach-config.h | 2 + sound/soc/amd/ps/ps-pdm-dma.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 2 +- sound/soc/apple/mca.c | 2 +- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 44 +- sound/soc/atmel/mchp-pdmc.c | 99 +- sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/i2sc.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/bcm/bcm63xx-i2s-whistler.c | 2 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4613.c | 4 +- sound/soc/codecs/cs-amp-lib-test.c | 44 + sound/soc/codecs/cs35l56-shared.c | 25 +- sound/soc/codecs/cs42l42-sdw.c | 12 +- sound/soc/codecs/cs42l43.c | 2 +- sound/soc/codecs/cs43130.c | 113 +- sound/soc/codecs/cs47l15.c | 2 +- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/cs47l35.c | 2 +- sound/soc/codecs/cs47l85.c | 2 +- sound/soc/codecs/cs47l90.c | 2 +- sound/soc/codecs/cs47l92.c | 2 +- sound/soc/codecs/es8326.c | 6 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/codecs/lpass-tx-macro.c | 2 +- sound/soc/codecs/lpass-va-macro.c | 2 +- sound/soc/codecs/lpass-wsa-macro.c | 27 +- sound/soc/codecs/msm8916-wcd-digital.c | 2 +- sound/soc/codecs/mt6357.c | 1855 ++++++++++ sound/soc/codecs/mt6357.h | 660 ++++ sound/soc/codecs/peb2466.c | 14 +- sound/soc/codecs/rk817_codec.c | 2 +- sound/soc/codecs/rt1318.c | 3 +- sound/soc/codecs/rt1320-sdw.c | 2147 +++++++++++- sound/soc/codecs/rt1320-sdw.h | 3 + sound/soc/codecs/rt5682.c | 4 +- sound/soc/codecs/rt5682s.c | 4 +- sound/soc/codecs/sti-sas.c | 21 +- sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - sound/soc/codecs/tas2781-comlib.c | 35 +- sound/soc/codecs/tas2781-fmwlib.c | 61 +- sound/soc/codecs/tas2781-i2c.c | 953 +++++- sound/soc/codecs/tas5086.c | 9 +- sound/soc/codecs/tlv320aic31xx.c | 102 +- sound/soc/codecs/tlv320aic32x4.c | 9 + sound/soc/codecs/wcd934x.c | 4 +- sound/soc/codecs/wcd937x.c | 2 +- sound/soc/codecs/wcd937x.h | 34 +- sound/soc/codecs/wcd938x.c | 20 +- sound/soc/codecs/wcd938x.h | 4 +- sound/soc/codecs/wcd939x.c | 2 +- sound/soc/codecs/wcd939x.h | 6 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- sound/soc/codecs/wsa881x.c | 44 +- sound/soc/codecs/wsa883x.c | 75 +- sound/soc/codecs/wsa884x.c | 240 +- sound/soc/dwc/dwc-i2s.c | 18 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_audmix.c | 14 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- sound/soc/fsl/fsl_mqs.c | 13 +- sound/soc/fsl/fsl_rpmsg.c | 11 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 13 +- sound/soc/fsl/fsl_ssi.c | 8 +- sound/soc/fsl/fsl_xcvr.c | 2 +- sound/soc/fsl/imx-audmux.c | 8 +- sound/soc/fsl/imx-card.c | 3 - sound/soc/fsl/imx-pcm-rpmsg.c | 16 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/lpc3xxx-i2s.c | 11 +- sound/soc/fsl/lpc3xxx-pcm.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 4 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- sound/soc/generic/audio-graph-card.c | 16 +- .../soc/generic/audio-graph-card2-custom-sample.c | 2 +- sound/soc/generic/audio-graph-card2.c | 15 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/generic/test-component.c | 11 +- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/img/img-parallel-out.c | 2 +- sound/soc/img/img-spdif-in.c | 2 +- sound/soc/img/img-spdif-out.c | 2 +- sound/soc/img/pistachio-internal-dac.c | 2 +- sound/soc/intel/Kconfig | 120 - sound/soc/intel/Makefile | 1 - sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/avs/core.c | 8 +- sound/soc/intel/boards/Kconfig | 156 +- sound/soc/intel/boards/Makefile | 31 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 720 ---- sound/soc/intel/boards/bxt_rt298.c | 670 ---- sound/soc/intel/boards/bytcht_cx2072x.c | 4 + sound/soc/intel/boards/bytcht_da7213.c | 4 + sound/soc/intel/boards/bytcht_es8316.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 + sound/soc/intel/boards/cht_bsw_rt5672.c | 4 + sound/soc/intel/boards/ehl_rt5660.c | 3 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 688 ---- sound/soc/intel/boards/kbl_da7219_max98927.c | 1175 ------- sound/soc/intel/boards/kbl_rt5660.c | 567 --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 1073 ------ .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 869 ----- sound/soc/intel/boards/skl_hda_dsp_common.c | 168 - sound/soc/intel/boards/skl_hda_dsp_common.h | 67 - sound/soc/intel/boards/skl_hda_dsp_generic.c | 242 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 704 ---- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 751 ---- sound/soc/intel/boards/skl_rt286.c | 568 --- sound/soc/intel/boards/sof_board_helpers.c | 152 + sound/soc/intel/boards/sof_board_helpers.h | 3 + sound/soc/intel/boards/sof_es8336.c | 14 +- sound/soc/intel/boards/sof_pcm512x.c | 5 +- sound/soc/intel/boards/sof_rt5682.c | 7 + sound/soc/intel/boards/sof_sdw.c | 1384 +------- sound/soc/intel/boards/sof_sdw_common.h | 172 +- sound/soc/intel/boards/sof_sdw_hdmi.c | 16 +- sound/soc/intel/boards/sof_ssp_amp.c | 6 + sound/soc/intel/boards/sof_wm8804.c | 6 +- sound/soc/intel/catpt/device.c | 2 +- sound/soc/intel/common/Makefile | 1 + sound/soc/intel/common/soc-acpi-intel-adl-match.c | 105 + sound/soc/intel/common/soc-acpi-intel-arl-match.c | 263 ++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 6 - sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 1 - sound/soc/intel/common/soc-acpi-intel-hda-match.c | 18 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 6 - sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 11 - sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 104 + sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 145 + sound/soc/intel/common/soc-acpi-intel-skl-match.c | 5 - sound/soc/intel/skylake/Makefile | 15 - sound/soc/intel/skylake/bxt-sst.c | 629 ---- sound/soc/intel/skylake/cnl-sst-dsp.c | 266 -- sound/soc/intel/skylake/cnl-sst-dsp.h | 103 - sound/soc/intel/skylake/cnl-sst.c | 508 --- sound/soc/intel/skylake/skl-debug.c | 248 -- sound/soc/intel/skylake/skl-i2s.h | 87 - sound/soc/intel/skylake/skl-messages.c | 1419 -------- sound/soc/intel/skylake/skl-nhlt.c | 269 -- sound/soc/intel/skylake/skl-pcm.c | 1507 -------- sound/soc/intel/skylake/skl-ssp-clk.c | 428 --- sound/soc/intel/skylake/skl-ssp-clk.h | 108 - sound/soc/intel/skylake/skl-sst-cldma.c | 373 -- sound/soc/intel/skylake/skl-sst-cldma.h | 243 -- sound/soc/intel/skylake/skl-sst-dsp.c | 462 --- sound/soc/intel/skylake/skl-sst-dsp.h | 256 -- sound/soc/intel/skylake/skl-sst-ipc.c | 1071 ------ sound/soc/intel/skylake/skl-sst-ipc.h | 169 - sound/soc/intel/skylake/skl-sst-utils.c | 425 --- sound/soc/intel/skylake/skl-sst.c | 599 ---- sound/soc/intel/skylake/skl-topology.c | 3605 -------------------- sound/soc/intel/skylake/skl-topology.h | 524 --- sound/soc/intel/skylake/skl.c | 1177 ------- sound/soc/intel/skylake/skl.h | 207 -- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/loongson/loongson_card.c | 125 +- sound/soc/loongson/loongson_dma.c | 25 +- sound/soc/loongson/loongson_i2s.c | 115 +- sound/soc/loongson/loongson_i2s.h | 24 +- sound/soc/loongson/loongson_i2s_pci.c | 51 +- sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 16 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 9 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 +- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 2 +- sound/soc/mediatek/mt8365/Makefile | 15 + sound/soc/mediatek/mt8365/mt8365-afe-clk.c | 421 +++ sound/soc/mediatek/mt8365/mt8365-afe-clk.h | 32 + sound/soc/mediatek/mt8365/mt8365-afe-common.h | 448 +++ sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 2274 ++++++++++++ sound/soc/mediatek/mt8365/mt8365-dai-adda.c | 311 ++ sound/soc/mediatek/mt8365/mt8365-dai-dmic.c | 310 ++ sound/soc/mediatek/mt8365/mt8365-dai-i2s.c | 846 +++++ sound/soc/mediatek/mt8365/mt8365-dai-pcm.c | 293 ++ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 343 ++ sound/soc/mediatek/mt8365/mt8365-reg.h | 993 ++++++ sound/soc/meson/aiu-fifo.h | 2 - sound/soc/meson/aiu.c | 2 +- sound/soc/meson/axg-card.c | 13 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 2 +- sound/soc/qcom/common.c | 1 - sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sm8250.c | 8 + sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/arndale.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 2 +- sound/soc/sdw_utils/Kconfig | 6 + sound/soc/sdw_utils/Makefile | 11 + .../soc_sdw_bridge_cs35l56.c} | 56 +- .../soc_sdw_cs42l42.c} | 13 +- .../soc_sdw_cs42l43.c} | 38 +- .../soc_sdw_cs_amp.c} | 18 +- .../sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} | 10 +- .../sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} | 56 +- .../soc_sdw_rt5682.c} | 12 +- .../sof_sdw_rt700.c => sdw_utils/soc_sdw_rt700.c} | 12 +- .../sof_sdw_rt711.c => sdw_utils/soc_sdw_rt711.c} | 38 +- .../soc_sdw_rt712_sdca.c} | 10 +- .../soc_sdw_rt722_sdca.c} | 10 +- .../soc_sdw_rt_amp.c} | 43 +- .../soc_sdw_rt_amp_coeff_tables.h} | 6 +- .../soc_sdw_rt_dmic.c} | 11 +- .../soc_sdw_rt_sdca_jack_common.c} | 40 +- sound/soc/sdw_utils/soc_sdw_utils.c | 1170 +++++++ sound/soc/sh/Kconfig | 1 + sound/soc/sh/fsi.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/rcar/adg.c | 4 +- sound/soc/sh/rcar/core.c | 19 +- sound/soc/sh/rcar/dma.c | 75 +- sound/soc/sh/rcar/rsnd.h | 10 - sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/sh/rz-ssi.c | 259 +- sound/soc/sh/siu_dai.c | 2 +- sound/soc/soc-ac97.c | 4 +- sound/soc/soc-core.c | 21 +- sound/soc/soc-dai.c | 58 - sound/soc/soc-dapm.c | 5 +- sound/soc/soc-pcm.c | 362 +- sound/soc/soc-topology-test.c | 132 +- sound/soc/soc-topology.c | 9 +- sound/soc/sof/amd/Kconfig | 11 + sound/soc/sof/amd/Makefile | 4 +- sound/soc/sof/amd/acp-common.c | 3 + sound/soc/sof/amd/acp-dsp-offset.h | 24 +- sound/soc/sof/amd/acp-loader.c | 2 +- sound/soc/sof/amd/acp.c | 88 +- sound/soc/sof/amd/acp.h | 11 +- sound/soc/sof/amd/acp70.c | 142 + sound/soc/sof/amd/pci-acp63.c | 2 +- sound/soc/sof/amd/pci-acp70.c | 112 + sound/soc/sof/amd/pci-rmb.c | 1 - sound/soc/sof/amd/pci-rn.c | 1 - sound/soc/sof/amd/pci-vangogh.c | 1 - sound/soc/sof/imx/imx8.c | 2 +- sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/imx/imx8ulp.c | 2 +- sound/soc/sof/intel/Kconfig | 17 + sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda-stream.c | 4 +- sound/soc/sof/intel/hda.c | 282 +- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 + sound/soc/sof/intel/mtl.c | 16 +- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 + sound/soc/sof/intel/shim.h | 1 + sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- sound/soc/sof/pcm.c | 2 +- sound/soc/sof/sof-audio.h | 8 +- sound/soc/sof/sof-priv.h | 16 +- sound/soc/sof/topology.c | 8 +- sound/soc/sprd/sprd-mcdt.c | 2 +- sound/soc/starfive/jh7110_pwmdac.c | 2 +- sound/soc/starfive/jh7110_tdm.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/stm/stm32_i2s.c | 6 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/stm/stm32_spdifrx.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra186_asrc.c | 2 +- sound/soc/tegra/tegra186_dspk.c | 2 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra210_admaif.c | 2 +- sound/soc/tegra/tegra210_adx.c | 2 +- sound/soc/tegra/tegra210_ahub.c | 2 +- sound/soc/tegra/tegra210_amx.c | 2 +- sound/soc/tegra/tegra210_dmic.c | 2 +- sound/soc/tegra/tegra210_i2s.c | 18 +- sound/soc/tegra/tegra210_mixer.c | 2 +- sound/soc/tegra/tegra210_mvc.c | 2 +- sound/soc/tegra/tegra210_ope.c | 2 +- sound/soc/tegra/tegra210_sfc.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 2 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/tegra/tegra_audio_graph_card.c | 2 +- sound/soc/tegra/tegra_pcm.c | 2 +- sound/soc/ti/ams-delta.c | 2 +- sound/soc/ti/davinci-i2s.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-pxs2.c | 2 +- sound/soc/uniphier/evea.c | 2 +- sound/soc/ux500/mop500.c | 2 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- sound/soc/xilinx/xlnx_spdif.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- sound/usb/line6/driver.c | 5 + sound/usb/quirks-table.h | 4 + sound/usb/stream.c | 4 +- tools/build/feature/Makefile | 53 +- tools/perf/Documentation/Build.txt | 28 + tools/perf/Makefile.config | 20 +- tools/perf/Makefile.perf | 27 +- .../pmu-events/arch/riscv/andes/ax45/firmware.json | 2 +- .../pmu-events/arch/riscv/riscv-sbi-firmware.json | 2 +- .../pmu-events/arch/riscv/sifive/u74/firmware.json | 2 +- .../arch/riscv/starfive/dubhe-80/firmware.json | 2 +- .../arch/riscv/thead/c900-legacy/firmware.json | 2 +- tools/perf/util/callchain.c | 2 +- tools/sound/dapm-graph | 44 +- tools/testing/selftests/bpf/Makefile | 2 +- .../selftests/bpf/prog_tests/uprobe_syscall.c | 4 +- tools/testing/selftests/dmabuf-heaps/dmabuf-heap.c | 4 +- tools/testing/selftests/drivers/net/hw/rss_ctx.py | 37 +- tools/testing/selftests/hid/hid_bpf.c | 26 + tools/testing/selftests/hid/progs/hid.c | 2 +- .../testing/selftests/hid/progs/hid_bpf_helpers.h | 2 +- tools/testing/selftests/kselftest/ksft.py | 2 +- tools/testing/selftests/kvm/riscv/get-reg-list.c | 8 +- tools/testing/selftests/mm/Makefile | 2 +- tools/testing/selftests/mm/mremap_test.c | 2 + tools/testing/selftests/net/mptcp/mptcp_connect.c | 8 +- tools/testing/selftests/net/mptcp/mptcp_join.sh | 158 +- tools/testing/selftests/seccomp/seccomp_bpf.c | 2 + virt/kvm/Kconfig | 4 +- virt/kvm/guest_memfd.c | 227 +- virt/kvm/kvm_main.c | 49 +- 984 files changed, 25292 insertions(+), 30662 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs4271.txt delete mode 100644 Documentation/devicetree/bindings/sound/da7213.txt create mode 100644 Documentation/devicetree/bindings/sound/dlg,da7213.yaml create mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml create mode 100644 Documentation/devicetree/bindings/sound/fsl,saif.yaml delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8365-mt6357.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mxs-saif.txt delete mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm512x.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt delete mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt delete mode 100644 arch/s390/kernel/alternative.h create mode 100644 include/sound/soc_sdw_utils.h create mode 100644 include/sound/tas2563-tlv.h create mode 100644 sound/soc/amd/acp/acp-sdw-sof-mach.c create mode 100644 sound/soc/amd/acp/amd-acp63-acpi-match.c create mode 100644 sound/soc/amd/acp/soc_amd_sdw_common.h create mode 100644 sound/soc/codecs/mt6357.c create mode 100644 sound/soc/codecs/mt6357.h delete mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/bxt_rt298.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5660.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c delete mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c delete mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h delete mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c delete mode 100644 sound/soc/intel/boards/skl_rt286.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c delete mode 100644 sound/soc/intel/skylake/Makefile delete mode 100644 sound/soc/intel/skylake/bxt-sst.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/cnl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-debug.c delete mode 100644 sound/soc/intel/skylake/skl-i2s.h delete mode 100644 sound/soc/intel/skylake/skl-messages.c delete mode 100644 sound/soc/intel/skylake/skl-nhlt.c delete mode 100644 sound/soc/intel/skylake/skl-pcm.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h delete mode 100644 sound/soc/intel/skylake/skl-sst-utils.c delete mode 100644 sound/soc/intel/skylake/skl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-topology.c delete mode 100644 sound/soc/intel/skylake/skl-topology.h delete mode 100644 sound/soc/intel/skylake/skl.c delete mode 100644 sound/soc/intel/skylake/skl.h create mode 100644 sound/soc/mediatek/mt8365/Makefile create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-clk.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-clk.h create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-common.h create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-adda.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-dmic.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-mt6357.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-reg.h create mode 100644 sound/soc/sdw_utils/Kconfig create mode 100644 sound/soc/sdw_utils/Makefile rename sound/soc/{intel/boards/bridge_cs35l56.c => sdw_utils/soc_sdw_bridge_cs35l56.c} (63%) rename sound/soc/{intel/boards/sof_sdw_cs42l42.c => sdw_utils/soc_sdw_cs42l42.c} (82%) rename sound/soc/{intel/boards/sof_sdw_cs42l43.c => sdw_utils/soc_sdw_cs42l43.c} (73%) rename sound/soc/{intel/boards/sof_sdw_cs_amp.c => sdw_utils/soc_sdw_cs_amp.c} (66%) rename sound/soc/{intel/boards/sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} (71%) rename sound/soc/{intel/boards/sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} (63%) rename sound/soc/{intel/boards/sof_sdw_rt5682.c => sdw_utils/soc_sdw_rt5682.c} (82%) rename sound/soc/{intel/boards/sof_sdw_rt700.c => sdw_utils/soc_sdw_rt700.c} (81%) rename sound/soc/{intel/boards/sof_sdw_rt711.c => sdw_utils/soc_sdw_rt711.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt712_sdca.c => sdw_utils/soc_sdw_rt712_sdca.c} (74%) rename sound/soc/{intel/boards/sof_sdw_rt722_sdca.c => sdw_utils/soc_sdw_rt722_sdca.c} (68%) rename sound/soc/{intel/boards/sof_sdw_rt_amp.c => sdw_utils/soc_sdw_rt_amp.c} (84%) rename sound/soc/{intel/boards/sof_sdw_amp_coeff_tables.h => sdw_utils/soc_sdw_rt_amp_coeff_tables.h} (97%) rename sound/soc/{intel/boards/sof_sdw_rt_dmic.c => sdw_utils/soc_sdw_rt_dmic.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt_sdca_jack_common.c => sdw_utils/soc_sdw_rt_sdca_jack_common.c} (78%) create mode 100644 sound/soc/sdw_utils/soc_sdw_utils.c create mode 100644 sound/soc/sof/amd/acp70.c create mode 100644 sound/soc/sof/amd/pci-acp70.c create mode 100644 sound/soc/sof/intel/pci-ptl.c

1 1

[GIT PULL] ASoC updates for v6.12
by Mark Brown 14 Sep '24

14 Sep '24

The following changes since commit 839a4ec06f75cec8fec2cc5fc14e921d0c3f7369: ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict (2024-08-23 11:02:53 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.12 for you to fetch changes up to 2772ee6de6cf94e5f2a0c0ce6067d0796a4170ba: ASoC: topology: Fix redundant logical jump (2024-09-13 19:40:38 +0100) ---------------------------------------------------------------- ASoC: Updates for v6.12 This is a very large set of changes, almost all in drivers rather than the core. Even with the addition of several quite large drivers the overall diffstat is negative thanks to the removal of some old Intel board support which has been obsoleted by the AVS driver, helped a bit by some factoring out into helpers (especially around the Soundwire machine drivers for x86). Highlights include: - More simplifications and cleanups throughout the subsystem from Morimoto-san. - Extensive cleanups and refactoring of the Soundwire drivers to make better use of helpers. - Removal of Intel machine support obsoleted by the AVS driver. - Lots of DT schema conversions. - Machine support for many AMD and Intel x86 platforms. - Support for AMD ACP 7.1, Mediatek MT6367 and MT8365, Realtek RTL1320 SoundWire and rev C, and Texas Instruments TAS2563 ---------------------------------------------------------------- Alexandre Mergnat (11): ASoC: dt-bindings: mediatek,mt8365-afe: Add audio afe document ASoC: dt-bindings: mediatek,mt8365-mt6357: Add audio sound card document dt-bindings: mfd: mediatek: Add codec property for MT6357 PMIC ASoC: mediatek: mt8365: Add common header ASoC: mediatek: mt8365: Add audio clock control support ASoC: mediatek: mt8365: Add I2S DAI support ASoC: mediatek: mt8365: Add ADDA DAI support ASoC: mediatek: mt8365: Add DMIC DAI support ASoC: mediatek: mt8365: Add PCM DAI support ASoC: mediatek: mt8365: Add the AFE driver support ASoC: mediatek: Add MT8365 support Andrei Simion (1): ASoC: dt-bindings: microchip,sama7g5-spdifrx: Add common DAI reference Andrew Kreimer (1): ASoC: tlv320aic31xx: Fix typos Animesh Agarwal (2): ASoC: dt-bindings: ti,pcm512x: Convert to dtschema ASoC: dt-bindings: fsl,imx-audio-es8328: Convert to dtschema Arnd Bergmann (2): ASoC: mediatek: mt8365: include linux/bitfield.h ASoC: mediatek: mt8365: remove unused mt8365_i2s_hd_str Balamurugan C (4): ASoC: Intel: soc-acpi: Add entry for sof_es8336 in ARL match table. ASoC: Intel: soc-acpi: Add entry for HDMI_In capture support in ARL match table ASoC: Intel: sof_rt5682: Add HDMI-In capture with rt5682 support for ARL. ASoC: Intel: ARL: Add entry for HDMI-In capture support to non-I2S codec boards. Baojun Xu (1): ASoC: tas2781: Remove unnecessary line feed for tasdevice_codec_remove Bard Liao (6): ASoC: Intel: soc-acpi-intel-ptl-match: add rt711-sdca table ASoC: Intel: soc-acpi-intel-ptl-match: Add rt722 support ASoC: Intel: sof_sdw: add rt1320 amp support ASoC: SOF: Intel: hda: refactoring topology name fixup for HDA mach ASoC: Intel: sof_sdw: move ignore_internal_dmic check earlier ASoC: Intel: sof_sdw: overwrite mach_params->dmic_num Biju Das (2): ASoC: dt-bindings: renesas,rz-ssi: Document port property ASoC: sh: rz-ssi: Add full duplex support Binbin Zhou (7): ASoC: loongson: Use BIT() macro ASoC: loongson: Simplify probe() with local dev variable ASoC: loongson: Simplify with dev_err_probe() ASoC: loongson: Simplify if statment in loongson_card_hw_params() ASoC: loongson: Replace if with ternary operator ASoC: loongson: Factor out loongson_card_acpi_find_device() function ASoC: loongson: Factor out loongson i2s enable clock functions Brent Lu (10): ASoC: SOF: Intel: hda: refactoring topology name fixup for SDW mach ASoC: SOF: Intel: hda: support BT link mask in mach_params ASoC: Intel: skl_hda_dsp_generic: support BT audio offload ASoC: Intel: skl_hda_dsp_generic: remove hdac-hdmi support ASoC: Intel: skl_hda_dsp_generic: use sof_hdmi_private to init HDMI ASoC: Intel: board_helpers: support HDA link initialization ASoC: Intel: skl_hda_dsp_generic: use common module for DAI links ASoC: Intel: ehl_rt5660: do not check common_hdmi_codec_drv ASoC: Intel: sof_pcm512x: do not check common_hdmi_codec_drv ASoC: SOF: Intel: hda: remove common_hdmi_codec_drv Cezary Rojewski (14): ALSA: hda: Move SST device entries to AVS ASoC: Intel: Drop skl_machine_pdata usage ASoC: Intel: Remove bxt_rt298 board driver ASoC: Intel: Remove bxt_da7219_max98357a board driver ASoC: Intel: Remove kbl_rt5663_rt5514_max98927 board driver ASoC: Intel: Remove kbl_rt5663_max98927 board driver ASoC: Intel: Remove kbl_rt5660 board driver ASoC: Intel: Remove kbl_da7219_max98927 board driver ASoC: Intel: Remove kbl_da7219_max98357a board driver ASoC: Intel: Remove skl_rt286 board driver ASoC: Intel: Remove skl_nau88l25_ssm4567 board driver ASoC: Intel: Remove skl_nau88l25_max98357a board driver ASoC: Intel: Remove skylake driver ASoC: Intel: avs: Enable by default for all SST configurations Charles Keepax (3): ASoC: Intel: soc-acpi: arl: Add match entries for new cs42l43 laptops ASoC: Intel: soc-acpi: adl: Add match entries for new cs42l43 laptops ASoC: Intel: soc-acpi: lnl: Add match entries for new cs42l43 laptops Chen Ni (4): ASoC: mediatek: mt8192: remove redundant null pointer check before of_node_put ASoC: cs42l42: Convert comma to semicolon ASoC: topology-test: Convert comma to semicolon ASoC: Intel: skl_hda_dsp_generic: convert comma to semicolon Christophe JAILLET (4): ASoC: sti-sas: Constify snd_soc_component_driver struct ASoC: cs43130: Constify snd_soc_component_driver struct ASoC: cs43130: Constify struct reg_sequence and reg_sequences ASoC: rt1318: Constify struct reg_sequence Codrin Ciubotariu (5): ASoC: atmel: mchp-i2s-mcc: Improve maxburst calculation for better performance ASoC: atmel: mchp-i2s-mcc: Remove interface name from stream_name ASoC: atmel: mchp-pdmc: Improve maxburst calculation for better performance ASoC: atmel: mchp-pdmc: Add snd_soc_dai_driver name ASoC: atmel: mchp-pdmc: Retain Non-Runtime Controls Colin Ian King (1): ASoC: codecs: wcd934x: make read-only array minCode_param static const Dmitry Torokhov (1): ASoC: tas5086: use sleeping variants of gpiod API Fabio Estevam (7): ASoC: fsl_audmix: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() ASoC: fsl_mqs: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() ASoC: fsl_rpmsg: Switch to RUNTIME_PM_OPS() ASoC: fsl_spdif: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() ASoC: fsl_ssi: Switch to SYSTEM_SLEEP_PM_OPS ASoC: imx-audmux: Switch to SYSTEM_SLEEP_PM_OPS ASoC: imx-pcm-rpmsg: Switch to RUNTIME/SYSTEM_SLEEP_PM_OPS() Frank Li (2): ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml ASoC: dt-bindings: Convert tpa6130a2.txt to yaml Fred Oh (1): ASoC: SOF: Intel: add PTL specific power control register Geert Uytterhoeven (1): ASoC: dt-bindings: dlg,da7213: Convert to json-schema Gustavo A. R. Silva (1): ASoC: SOF: sof-audio: Avoid -Wflex-array-member-not-at-end warnings Heiko Stuebner (2): ASoC: dt-bindings: realtek,rt5616: document mclk clock ASoC: dt-bindings: realtek,rt5616: Document audio graph port Hongbo Li (2): ASoC: dapm: Use IS_ERR_OR_NULL() helper function ASoC: adi: Use str_enabled_disabled() helper Jens Reidel (1): ASoC: qcom: sm8250: enable primary mi2s Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Krzysztof Kozlowski (16): ASoC: codecs: wsa881x: Drop unused version readout ASoC: codecs: wsa883x: Handle reading version failure ASoC: codecs: wsa883x: Simplify handling variant/version ASoC: codecs: wsa884x: Simplify handling variant ASoC: dt-bindings: qcom,apq8016-sbc-sndcard: move to separate binding ASoC: codecs: lpass-wsa-macro: Do not hard-code dai in VI mixer ASoC: codecs: wsa881x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa883x: Use designator array initializers for Soundwire ports ASoC: codecs: wsa884x: Use designator array initializers for Soundwire ports ASoC: codecs: wcd938x: Drop unused defines and enums ASoC: codecs: wcd937x: Move max port number defines to enum ASoC: codecs: wcd938x: Move max port number defines to enum ASoC: codecs: wcd939x: Move max port number defines to enum ASoC: dt-bindings: samsung,odroid: drop stale clocks ASoC: MAINTAINERS: Drop incorrect tlv320aic31xx.txt path ASoC: codecs: wsa884x: Implement temperature reading and hwmon Kuninori Morimoto (19): ASoC: rsnd: remove rsnd_mod_confirm_ssi() under DEBUG ASoC: rsnd: use pcm_dmaengine code ASoC: stm: use snd_pcm_direction_name() ASoC: sof: pcm: use snd_pcm_direction_name() ASoC: sof: intel: use snd_pcm_direction_name() ASoC: fsl: lpc3xxx-i2s: use snd_pcm_direction_name() ASoC: tegra: use snd_pcm_direction_name() ASoC: soc-pcm: use snd_pcm_direction_name() ASoC: soc-dapm: use snd_pcm_direction_name() ASoC: remove bespoke trigger support ASoC: soc-pcm: remove snd_soc_dpcm_stream_{lock/unlock}_irq() ASoC: soc-pcm: remove snd_soc_dpcm_stream_lock_irqsave_nested() ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() ASoC: Intel: sof_sdw: use playback/capture_only flags ASoC: soc-pcm: move snd_soc_dpcm_can_be_xxx() to top ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function ASoC: makes rtd->initialized bit field ASoC: dt-bindings: renesas,rsnd: add post-init-providers property Linus Walleij (3): ASoC: tas2781-i2c: Drop weird GPIO code ASoC: tas2781-i2c: Get the right GPIO line ASoC: tas*: Drop unused GPIO includes Liu Jing (1): ASoC: mediatek: mt2701-cs42448: Optimize redundant code in mt2701_cs42448_machine_probe Luca Ceresoli (3): ASoC: dapm-graph: remove the "ROOT" cluster ASoC: dapm-graph: visualize component On/Off bias level ASoC: dapm-graph: show path name for non-static routes Lukasz Majewski (1): ASoC: dt-bindings: Convert mxs-saif.txt to fsl,saif.yaml (imx28 saif) Ma Ke (2): ASoC: rt5682s: Return devm_of_clk_add_hw_provider to transfer the error ASoC: rt5682: Return devm_of_clk_add_hw_provider to transfer the error Maciej Strozek (1): ASoC: Intel: sof_sdw: Add quirks from some new Dell laptops Marek Vasut (1): ASoC: tlv320aic32x4: Add multi endpoint support Mark Brown (37): Add HDMI Audio support ASoC: codecs: wsa88xx: Few cleanups ASoC: codecs: wsa88xx and wcd93xx: Soundwire port AMD SOF based generic SoundWire machine driver ALSA/ASoC: use snd_pcm_direction_name() ASoC: SOF: reshuffle and optimize structures ASoC/SOF/PCI/Intel: add PantherLake support ASoC: Merge fixes ASoC: tas*: Fix up GPIO usage ASoC: Intel: Remove skylake driver ALSA/ASoC/SoundWire: Intel: update maximum number ASoC: Merge up fixes ASoC: grace time for DPCM cleanup ASoC: Merge up fixes ASoC: dapm-graph: add component on/off and route names Add SOF support for ACP7.0 based platform ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local ASoC: Intel: boards: updates for 6.12 ASoC: dt-bindings: amlogic-sound-cards: document Add audio support for the MediaTek Genio 350-evk Fixes for the audio setup on the rk3588-nanopc-t6 Add i2s/tdm support for acp7.0 and acp7.1 platforms Some clean up with helper fucntion Add audio support for the MediaTek Genio 350-evk ASoC: mt8365: Open code BIT() to avoid spurious warnings ASoC: mt8365: Remove spurious unsigned long casts ASoC: mt8365: Remove unused prototype for mt8365_afe_clk_group_48k() ASoC: mt8365: Make non-exported functions static ASoC: mt8365: Remove unused variables ASoC: mt8365: Remove unused DMIC IIR coefficient configuration ASoC: mt8365: Allow build coverage ASoC: loongson: Simplify code formatting ASoC: mt8365: Fix -Werror builds Add support for primary mi2s on SM8250 ASoC: Improvements for mchp-pdmc ASoC: Intel: boards: updates for 6.12 - part 2 AMD SoundWire machine driver code refactor Muhammad Usama Anjum (3): ASoC: amd: acp: Return in-case of error ASoC: mediatek: mt8365: check validity before usage of i2s_data ASoc: mediatek: mt8365: Remove unneeded assignment Neil Armstrong (2): ASoC: dt-bindings: amlogic,axg-sound-card: document clocks property ASoC: dt-bindings: amlogic,gx-sound-card: document clocks property Nicolas Belin (2): ASoc: mediatek: mt8365: Add a specific soundcard for EVK ASoC: codecs: add MT6357 support Nikita Shubin (1): ASoC: dt-bindings: cirrus,cs4271: Convert to dtschema Pierre-Louis Bossart (11): ASoC: Intel: soc-acpi: add PTL match tables ASoC: SOF: Intel: add initial support for PTL ASoC: SOF: sof-priv.h: optimize snd_sof_platform_stream_params ASoC: SOF: sof-priv.h: optimize snd_sof_mailbox ASoC: SOF: sof-priv.h: optimize snd_sof_ipc_msg ASoC: SOF: sof-audio.h: optimize snd_sof_pcm_stream_pipeline_list ALSA/ASoC/SoundWire: Intel: use single definition for SDW_INTEL_MAX_LINKS soundwire: intel: add probe-time check on link id soundwire: intel: increase maximum number of links ASoC: Intel: sof_sdw: make sof_sdw_quirk static ASoC: Intel: boards: always check the result of acpi_dev_get_first_match_dev() Richard Fitzgerald (2): ASoC: cs35l56: Make struct regmap_config const ASoC: cs-amp-lib: Add KUnit test case for empty calibration entries Rob Herring (Arm) (2): ASoC: Use of_property_present() ASoC: Use of_property_read_bool() Romain Gantois (1): ASoC: tlv320aic31xx: Add support for loading filter coefficients Shen Lichuan (1): ASoC: SOF: topology: Use kmemdup_array instead of kmemdup for multiple allocation Shenghao Ding (7): ASoC: tas2781: Add TAS2563 into the Header ASoC: tas2781: Fix a compiling warning reported by robot kernel test due to adding tas2563_dvc_table ASoc: tas2781: Rename dai_driver name to unify the name between TAS2563 and TAS2781 ASoc: tas2781: Remove unnecessary line feed and space ASoC: tas2781: mark const variables tas2563_dvc_table as __maybe_unused ASoC: tas2781: replace devm_kzalloc and scnprintf with devm_kstrdup ASoC: tas2781: Add Calibration Kcontrols for Chromebook Shuming Fan (1): ASoC: rt1320: Add support for version C Simon Trimmer (1): ASoC: cs35l56: Use regmap_read_bypassed() to wake the device Srinivas Kandagatla (1): ASoC: dt-bindings: qcom,lpass-wsa-macro: correct clocks on SM8250 Takashi Iwai (1): ASoC: Use __counted_by() annotation for snd_soc_pcm_runtime Tang Bin (6): ASoC: loongson: remove redundant variable assignments ASoC: codecs: fix the right check and simplify code ASoC: soc-ac97: Fix the incorrect description ASoC: loongson: Add the correct judgement return ASoC: mediatek: mt7986-afe-pcm: Remove redundant error message ASoC: topology: Fix redundant logical jump Uwe Kleine-König (1): ASoC: Switch back to struct platform_driver::remove() Venkata Prasad Potturu (12): ASoC: amd: acp: Refactor TDM slots selction based on acp revision id ASoC: amd: acp: Refactor I2S dai driver ASoC: amd: acp: Update pcm hardware capabilities for acp6.3 platform ASoC: amd: acp: Add I2S TDM support for acp6.3 platform ASoC: amd: acp: Update pcm hardware capabilities for acp7.0 platform ASoC: amd: acp: Add I2S master clock generation support for acp7.0 platform ASoC: amd: acp: Set i2s clock for acp7.0 platform ASoC: amd: acp: Modify max channels and sample rate support for acp70 dai driver ASoC: amd: acp: Add I2S TDM support for acp7.0 platform ASoC: amd: acp: Add pte configuration for ACP7.0 platform ASoC: amd: acp: Add i2s master clock generation support for acp7.1 platform ASoC: amd: acp: Add I2S TDM support for acp7.1 platform Vijendar Mukunda (47): ASoC: intel: rename codec_info and dai_info structures names ASoC: intel: rename soundwire common header macros ASoC: intel: rename soundwire machine driver soc ops ASoC: intel: rename soundwire codec helper functions ASoC: intel: rename maxim codec macros ASoC: intel: rename ignore_pch_dmic variable name ASoC: intel/sdw-utils: move soundwire machine driver soc ops ASoC: intel: move soundwire machine driver common structures ASoC: intel/sdw_utils: move soundwire machine driver helper functions ASoC: intel/sdw_utils: move dmic codec helper function ASoC: intel/sdw_utils: move rtk dmic helper functions ASoC: intel/sdw_utils: move rt712 sdca helper functions ASoC: intel/sdw_utils: move rt722 sdca helper functions ASoC: intel: split soundwire machine driver private data ASoC: intel/sdw_utils: move rt5682 codec helper function ASoC: intel/sdw_utils: move rtk jack common helper functions ASoC: intel/sdw_utils: move rt700 and rt711 codec helper functions ASoC: intel/sdw_utils: move rtk amp codec helper functions ASoC: intel/sdw_utils: move cirrus soundwire codec helper functions ASoC: intel/sdw_utils: move maxim codec helper functions ASoC: intel/sdw_utils: move dai id common macros ASoC: intel/sdw_utils: move soundwire dai type macros ASoC: intel/sdw_utils: move soundwire codec_info_list structure ASoC: intel/sdw_utils: move machine driver dai link helper functions ASoC: sdw_utils: refactor sof_sdw_card_late_probe function ASoC: intel/sdw_utils: refactor init_dai_link() and init_simple_dai_link() ASoC: soc-acpi: add pci revision id field in mach params structure ASoC: amd: acp: add soundwire machines for acp6.3 based platform ASoC: SOF: amd: add alternate machines for acp6.3 based platform ASoC: SOF: amd: update mach params subsystem_rev variable ASoC: amd/sdw_utils: add sof based soundwire generic machine driver ASoC: amd: acp: remove MODULE_ALIAS for SoundWire machine driver ASoC: amd: acp: add ZSC control register programming sequence ASoC: SOF: amd: update conditional check for cache register update ASoC: amd: acp: Convert comma to semicolon ASoC: amd: acp: add legacy driver support for ACP7.1 based platforms ASoC: amd: acp: replace desc->rev check with acp pci revision id ASoC: SOF: amd: remove unused variable from sof_amd_acp_desc structure ASoC: amd: Add acpi machine id for acp7.0 version based platform ASoC: SOF: amd: add support for acp7.0 based platform ASoC: amd: acp: remove MODULE_ALIAS for legacy machine driver ASoC: amd: acp: remove MODULE_ALIAS for sof based generic machine driver ASoC: intel: sof_sdw: rename soundwire endpoint and dailink structures ASoC: intel: sof_sdw: rename soundwire parsing helper functions ASoC: sdw_util/intel: move soundwire endpoint and dai link structures ASoC: sdw_utils/intel: move soundwire endpoint parsing helper functions ASoC: amd: acp: refactor SoundWire machine driver code Wu Bo (1): ASoC: dwc: change to use devm_clk_get_enabled() helpers Yue Haibing (2): ASoC: fsl: lpc3xxx-i2s: Remove set but not used variable 'savedbitclkrate' ASoC: fsl: lpc3xxx: Make some symbols static Zhang Yi (2): ASoC: codecs: ES8326: suspend issue ASoC: codecs: ES8326: input issue after init Zhang Zekun (3): ASoC: audio-graph-card: Use for_each_child_of_node_scoped() to simplify code ASoC: audio-graph-card2: Use helper function of_get_child_count() ASoC: meson: Remove unused declartion in header file tangbin (3): ASoC: loongson: Remove useless variable definitions ASoC: loongson: remove unnecessary assignment in i2s_resume() ASoC: loongson: fix error release ying zuxin (1): ASoC: codecs: Use devm_clk_get_enabled() helpers .mailmap | 1 + .../testing/sysfs-bus-i2c-devices-turris-omnia-mcu | 6 +- Documentation/admin-guide/cifs/usage.rst | 2 +- Documentation/admin-guide/kernel-parameters.txt | 4 +- Documentation/arch/arm64/silicon-errata.rst | 18 + .../devicetree/bindings/ata/rockchip,dwc-ahci.yaml | 3 + .../bindings/display/panel/samsung,atna33xc20.yaml | 9 +- .../devicetree/bindings/mfd/mediatek,mt6357.yaml | 21 + .../bindings/sound/amlogic,axg-sound-card.yaml | 7 + .../bindings/sound/amlogic,gx-sound-card.yaml | 7 + .../devicetree/bindings/sound/cirrus,cs4271.yaml | 101 + Documentation/devicetree/bindings/sound/cs4271.txt | 57 - Documentation/devicetree/bindings/sound/da7213.txt | 45 - .../devicetree/bindings/sound/dlg,da7213.yaml | 103 + .../bindings/sound/fsl,imx-audio-es8328.yaml | 111 + .../devicetree/bindings/sound/fsl,saif.yaml | 83 + .../devicetree/bindings/sound/imx-audio-es8328.txt | 60 - .../bindings/sound/mediatek,mt8365-afe.yaml | 130 + .../bindings/sound/mediatek,mt8365-mt6357.yaml | 107 + .../bindings/sound/microchip,sama7g5-spdifrx.yaml | 5 +- .../devicetree/bindings/sound/mxs-saif.txt | 41 - .../devicetree/bindings/sound/pcm512x.txt | 53 - .../bindings/sound/qcom,apq8016-sbc-sndcard.yaml | 205 ++ .../bindings/sound/qcom,lpass-wsa-macro.yaml | 22 +- .../devicetree/bindings/sound/qcom,sm8250.yaml | 137 - .../devicetree/bindings/sound/realtek,rt5616.yaml | 12 + .../devicetree/bindings/sound/renesas,rsnd.yaml | 6 + .../devicetree/bindings/sound/renesas,rz-ssi.yaml | 4 + .../devicetree/bindings/sound/samsung,odroid.yaml | 5 - .../devicetree/bindings/sound/ti,pcm512x.yaml | 101 + .../bindings/sound/ti,tlv320dac3100.yaml | 127 + .../devicetree/bindings/sound/ti,tpa6130a2.yaml | 55 + .../devicetree/bindings/sound/tlv320aic31xx.txt | 77 - .../devicetree/bindings/sound/tpa6130a2.txt | 27 - .../devicetree/bindings/usb/microchip,usb2514.yaml | 1 + Documentation/driver-api/thermal/sysfs-api.rst | 65 +- Documentation/netlink/specs/ethtool.yaml | 2 +- Documentation/networking/ethtool-netlink.rst | 1 + .../process/embargoed-hardware-issues.rst | 153 +- .../userspace-api/media/v4l/pixfmt-yuv-luma.rst | 4 +- Documentation/virt/kvm/api.rst | 8 +- Documentation/wmi/devices/msi-wmi-platform.rst | 6 +- MAINTAINERS | 13 +- Makefile | 2 +- arch/alpha/include/asm/io.h | 4 +- arch/arm/Kconfig | 4 +- arch/arm/boot/compressed/Makefile | 1 + arch/arm/boot/compressed/vmlinux.lds.S | 2 +- arch/arm/boot/dts/arm/versatile-ab.dts | 2 +- arch/arm/include/asm/stacktrace.h | 7 + arch/arm/include/asm/vmlinux.lds.h | 2 +- arch/arm/kernel/entry-armv.S | 3 + arch/arm/kernel/entry-common.S | 3 + arch/arm/kernel/module.c | 5 - arch/arm/kernel/perf_callchain.c | 3 +- arch/arm/kernel/vmlinux-xip.lds.S | 4 +- arch/arm/kernel/vmlinux.lds.S | 6 +- arch/arm/mach-alpine/alpine_cpu_pm.c | 2 +- arch/arm/mach-pxa/gumstix.c | 11 +- arch/arm/mm/proc.c | 20 +- arch/arm64/Kconfig | 22 +- arch/arm64/boot/dts/ti/k3-am62-verdin-dahlia.dtsi | 22 - arch/arm64/boot/dts/ti/k3-am62-verdin.dtsi | 6 - .../boot/dts/ti/k3-am62p-j722s-common-mcu.dtsi | 2 + arch/arm64/boot/dts/ti/k3-am62p-main.dtsi | 3 +- arch/arm64/boot/dts/ti/k3-j722s-main.dtsi | 3 +- arch/arm64/boot/dts/ti/k3-j784s4-evm.dts | 25 +- arch/arm64/boot/dts/ti/k3-j784s4-main.dtsi | 4 +- arch/arm64/include/asm/cputype.h | 4 + arch/arm64/include/asm/jump_label.h | 1 + arch/arm64/kernel/Makefile.syscalls | 2 +- arch/arm64/kernel/cpu_errata.c | 11 +- arch/arm64/kernel/jump_label.c | 11 +- arch/loongarch/include/asm/hugetlb.h | 4 +- arch/loongarch/include/asm/kfence.h | 6 +- arch/loongarch/include/asm/kvm_host.h | 2 - arch/loongarch/include/asm/kvm_para.h | 4 +- arch/loongarch/include/asm/pgtable.h | 48 +- arch/loongarch/kernel/Makefile.syscalls | 3 +- arch/loongarch/kernel/efi.c | 6 + arch/loongarch/kvm/mmu.c | 8 +- arch/loongarch/mm/hugetlbpage.c | 6 +- arch/loongarch/mm/init.c | 10 +- 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fs/xfs/scrub/parent.c | 2 +- fs/xfs/scrub/trace.h | 10 +- fs/xfs/xfs_attr_list.c | 2 +- fs/xfs/xfs_trace.h | 10 +- fs/xfs/xfs_xattr.c | 19 +- include/asm-generic/vmlinux.lds.h | 11 +- include/drm/drm_buddy.h | 2 + include/linux/compiler.h | 9 + include/linux/cpuhotplug.h | 2 +- include/linux/cpumask.h | 2 +- include/linux/ethtool.h | 10 +- include/linux/i2c.h | 2 +- include/linux/kvm_host.h | 9 +- include/linux/minmax.h | 115 +- include/linux/profile.h | 1 - include/linux/ring_buffer.h | 1 - include/linux/soundwire/sdw_intel.h | 8 + include/linux/trace_events.h | 3 +- include/linux/virtio.h | 3 + include/linux/virtio_config.h | 4 - include/linux/virtio_net.h | 16 +- include/sound/cs35l56.h | 6 +- include/sound/soc-acpi-intel-match.h | 2 + include/sound/soc-acpi.h | 6 +- include/sound/soc-dai.h | 6 +- include/sound/soc-dpcm.h | 19 - include/sound/soc.h | 7 +- include/sound/soc_sdw_utils.h | 247 ++ include/sound/tas2563-tlv.h | 279 ++ include/sound/tas2781-tlv.h | 260 -- include/sound/tas2781.h | 71 +- include/sound/ump_convert.h | 1 + include/trace/events/btrfs.h | 8 + include/trace/events/mptcp.h | 2 +- include/uapi/asm-generic/unistd.h | 5 +- include/ufs/ufshcd.h | 1 + include/ufs/ufshci.h | 1 + init/Kconfig | 1 + io_uring/napi.c | 2 - io_uring/net.c | 7 +- io_uring/poll.c | 1 + kernel/dma/debug.c | 5 +- kernel/irq/irqdesc.c | 1 + kernel/jump_label.c | 4 +- kernel/kcov.c | 15 +- kernel/kprobes.c | 4 +- kernel/ksysfs.c | 7 + kernel/locking/lockdep.c | 6 + kernel/locking/qspinlock_paravirt.h | 2 +- kernel/module/main.c | 41 +- kernel/padata.c | 7 + kernel/profile.c | 242 +- kernel/sched/core.c | 68 +- kernel/sched/cputime.c | 6 + kernel/sched/stats.c | 10 - kernel/task_work.c | 6 + kernel/time/clocksource.c | 2 +- kernel/time/ntp.c | 9 +- kernel/time/tick-broadcast.c | 3 +- kernel/time/timekeeping.c | 2 +- kernel/trace/fgraph.c | 2 +- kernel/trace/preemptirq_delay_test.c | 2 - kernel/trace/ring_buffer.c | 12 - kernel/trace/trace.h | 23 + kernel/trace/trace_events.c | 41 +- kernel/trace/trace_events_hist.c | 4 +- kernel/trace/trace_events_inject.c | 2 +- kernel/trace/trace_events_trigger.c | 6 +- kernel/trace/tracing_map.c | 6 +- lib/btree.c | 1 - lib/decompress_unlzma.c | 2 + lib/vsprintf.c | 2 +- mm/list_lru.c | 28 +- mm/memcontrol.c | 22 +- mm/shmem.c | 14 +- mm/slub.c | 3 + mm/zsmalloc.c | 2 - net/bluetooth/hci_core.c | 7 - net/bluetooth/hci_event.c | 5 +- net/bluetooth/hci_sync.c | 35 + net/bluetooth/l2cap_core.c | 1 + net/bridge/br_multicast.c | 4 +- net/core/dev.c | 1 + net/core/link_watch.c | 4 +- net/core/rtnetlink.c | 2 +- net/ethtool/ioctl.c | 61 +- net/ethtool/rss.c | 8 +- net/ipv4/netfilter/iptable_nat.c | 18 +- net/ipv4/tcp_ao.c | 43 +- net/ipv4/tcp_input.c | 23 +- net/ipv4/tcp_offload.c | 3 + net/ipv4/udp_offload.c | 4 + net/ipv6/ndisc.c | 34 +- net/ipv6/netfilter/ip6table_nat.c | 14 +- net/iucv/af_iucv.c | 4 +- net/l2tp/l2tp_core.c | 15 +- net/mac80211/cfg.c | 7 +- net/mac80211/tx.c | 5 +- net/mac80211/util.c | 2 +- net/mptcp/mib.c | 2 + net/mptcp/mib.h | 2 + net/mptcp/options.c | 5 +- net/mptcp/pm.c | 12 + net/mptcp/pm_netlink.c | 93 +- net/mptcp/pm_userspace.c | 18 + net/mptcp/protocol.c | 18 +- net/mptcp/protocol.h | 4 + net/mptcp/subflow.c | 26 +- net/sched/act_ct.c | 4 +- net/sctp/input.c | 19 +- net/smc/af_smc.c | 7 +- net/smc/smc_stats.h | 2 +- net/sunrpc/svc.c | 2 +- net/wireless/scan.c | 11 +- net/wireless/sme.c | 1 + rust/kernel/firmware.rs | 2 +- scripts/syscall.tbl | 6 +- sound/core/seq/seq_ports.h | 14 +- sound/core/seq/seq_ump_convert.c | 132 +- sound/core/ump_convert.c | 60 +- sound/firewire/amdtp-stream.c | 38 +- sound/firewire/amdtp-stream.h | 1 + sound/hda/intel-dsp-config.c | 57 +- sound/hda/intel-sdw-acpi.c | 5 +- sound/pci/hda/cs35l41_hda.c | 13 - sound/pci/hda/cs35l56_hda.c | 38 +- sound/pci/hda/cs35l56_hda.h | 1 - sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_generic.c | 63 + sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/hda_intel.c | 10 +- sound/pci/hda/patch_conexant.c | 56 +- sound/pci/hda/patch_hdmi.c | 2 + sound/pci/hda/patch_realtek.c | 2 + sound/pci/hda/tas2781_hda_i2c.c | 2 +- sound/soc/Kconfig | 2 + sound/soc/Makefile | 1 + sound/soc/adi/axi-i2s.c | 6 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp-config.c | 12 + sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/acp/Kconfig | 22 + sound/soc/amd/acp/Makefile | 4 + sound/soc/amd/acp/acp-i2s.c | 188 +- sound/soc/amd/acp/acp-legacy-common.c | 40 +- sound/soc/amd/acp/acp-legacy-mach.c | 7 - sound/soc/amd/acp/acp-mach-common.c | 2 +- sound/soc/amd/acp/acp-mach.h | 1 + sound/soc/amd/acp/acp-pci.c | 4 + sound/soc/amd/acp/acp-pdm.c | 7 +- sound/soc/amd/acp/acp-platform.c | 117 +- sound/soc/amd/acp/acp-rembrandt.c | 2 +- sound/soc/amd/acp/acp-renoir.c | 2 +- sound/soc/amd/acp/acp-sdw-sof-mach.c | 509 +++ sound/soc/amd/acp/acp-sof-mach.c | 7 - sound/soc/amd/acp/acp63.c | 2 +- sound/soc/amd/acp/acp70.c | 87 +- sound/soc/amd/acp/amd-acp63-acpi-match.c | 90 + sound/soc/amd/acp/amd.h | 27 +- sound/soc/amd/acp/chip_offset_byte.h | 84 +- sound/soc/amd/acp/soc_amd_sdw_common.h | 44 + sound/soc/amd/mach-config.h | 2 + sound/soc/amd/ps/ps-pdm-dma.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 2 +- sound/soc/apple/mca.c | 2 +- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 44 +- sound/soc/atmel/mchp-pdmc.c | 99 +- sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/i2sc.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/bcm/bcm63xx-i2s-whistler.c | 2 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4613.c | 4 +- sound/soc/codecs/cs-amp-lib-test.c | 44 + sound/soc/codecs/cs35l56-shared.c | 25 +- sound/soc/codecs/cs42l42-sdw.c | 12 +- sound/soc/codecs/cs42l43.c | 2 +- sound/soc/codecs/cs43130.c | 113 +- sound/soc/codecs/cs47l15.c | 2 +- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/cs47l35.c | 2 +- sound/soc/codecs/cs47l85.c | 2 +- sound/soc/codecs/cs47l90.c | 2 +- sound/soc/codecs/cs47l92.c | 2 +- sound/soc/codecs/es8326.c | 6 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/codecs/lpass-tx-macro.c | 2 +- sound/soc/codecs/lpass-va-macro.c | 2 +- sound/soc/codecs/lpass-wsa-macro.c | 27 +- sound/soc/codecs/msm8916-wcd-digital.c | 2 +- sound/soc/codecs/mt6357.c | 1855 ++++++++++ sound/soc/codecs/mt6357.h | 660 ++++ sound/soc/codecs/peb2466.c | 14 +- sound/soc/codecs/rk817_codec.c | 2 +- sound/soc/codecs/rt1318.c | 3 +- sound/soc/codecs/rt1320-sdw.c | 2147 +++++++++++- sound/soc/codecs/rt1320-sdw.h | 3 + sound/soc/codecs/rt5682.c | 4 +- sound/soc/codecs/rt5682s.c | 4 +- sound/soc/codecs/sti-sas.c | 21 +- sound/soc/codecs/tas2552.c | 1 - sound/soc/codecs/tas2764.c | 1 - sound/soc/codecs/tas2770.c | 1 - sound/soc/codecs/tas2780.c | 1 - sound/soc/codecs/tas2781-comlib.c | 35 +- sound/soc/codecs/tas2781-fmwlib.c | 61 +- sound/soc/codecs/tas2781-i2c.c | 953 +++++- sound/soc/codecs/tas5086.c | 9 +- sound/soc/codecs/tlv320aic31xx.c | 102 +- sound/soc/codecs/tlv320aic32x4.c | 9 + sound/soc/codecs/wcd934x.c | 4 +- sound/soc/codecs/wcd937x.c | 2 +- sound/soc/codecs/wcd937x.h | 34 +- sound/soc/codecs/wcd938x.c | 20 +- sound/soc/codecs/wcd938x.h | 4 +- sound/soc/codecs/wcd939x.c | 2 +- sound/soc/codecs/wcd939x.h | 6 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- sound/soc/codecs/wsa881x.c | 44 +- sound/soc/codecs/wsa883x.c | 75 +- sound/soc/codecs/wsa884x.c | 240 +- sound/soc/dwc/dwc-i2s.c | 18 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_audmix.c | 14 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- sound/soc/fsl/fsl_mqs.c | 13 +- sound/soc/fsl/fsl_rpmsg.c | 11 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 13 +- sound/soc/fsl/fsl_ssi.c | 8 +- sound/soc/fsl/fsl_xcvr.c | 2 +- sound/soc/fsl/imx-audmux.c | 8 +- sound/soc/fsl/imx-card.c | 3 - sound/soc/fsl/imx-pcm-rpmsg.c | 16 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/lpc3xxx-i2s.c | 11 +- sound/soc/fsl/lpc3xxx-pcm.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 4 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- sound/soc/generic/audio-graph-card.c | 16 +- .../soc/generic/audio-graph-card2-custom-sample.c | 2 +- sound/soc/generic/audio-graph-card2.c | 15 +- sound/soc/generic/simple-card.c | 4 +- sound/soc/generic/test-component.c | 11 +- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/img/img-parallel-out.c | 2 +- sound/soc/img/img-spdif-in.c | 2 +- sound/soc/img/img-spdif-out.c | 2 +- sound/soc/img/pistachio-internal-dac.c | 2 +- sound/soc/intel/Kconfig | 120 - sound/soc/intel/Makefile | 1 - sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/avs/core.c | 8 +- sound/soc/intel/boards/Kconfig | 156 +- sound/soc/intel/boards/Makefile | 31 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 720 ---- sound/soc/intel/boards/bxt_rt298.c | 670 ---- sound/soc/intel/boards/bytcht_cx2072x.c | 4 + sound/soc/intel/boards/bytcht_da7213.c | 4 + sound/soc/intel/boards/bytcht_es8316.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 4 +- sound/soc/intel/boards/bytcr_rt5651.c | 4 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 4 + sound/soc/intel/boards/cht_bsw_rt5672.c | 4 + sound/soc/intel/boards/ehl_rt5660.c | 3 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 688 ---- sound/soc/intel/boards/kbl_da7219_max98927.c | 1175 ------- sound/soc/intel/boards/kbl_rt5660.c | 567 --- sound/soc/intel/boards/kbl_rt5663_max98927.c | 1073 ------ .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 869 ----- sound/soc/intel/boards/skl_hda_dsp_common.c | 168 - sound/soc/intel/boards/skl_hda_dsp_common.h | 67 - sound/soc/intel/boards/skl_hda_dsp_generic.c | 242 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 704 ---- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 751 ---- sound/soc/intel/boards/skl_rt286.c | 568 --- sound/soc/intel/boards/sof_board_helpers.c | 152 + sound/soc/intel/boards/sof_board_helpers.h | 3 + sound/soc/intel/boards/sof_es8336.c | 14 +- sound/soc/intel/boards/sof_pcm512x.c | 5 +- sound/soc/intel/boards/sof_rt5682.c | 7 + sound/soc/intel/boards/sof_sdw.c | 1384 +------- sound/soc/intel/boards/sof_sdw_common.h | 172 +- sound/soc/intel/boards/sof_sdw_hdmi.c | 16 +- sound/soc/intel/boards/sof_ssp_amp.c | 6 + sound/soc/intel/boards/sof_wm8804.c | 6 +- sound/soc/intel/catpt/device.c | 2 +- sound/soc/intel/common/Makefile | 1 + sound/soc/intel/common/soc-acpi-intel-adl-match.c | 105 + sound/soc/intel/common/soc-acpi-intel-arl-match.c | 263 ++ sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 6 - sound/soc/intel/common/soc-acpi-intel-ehl-match.c | 1 - sound/soc/intel/common/soc-acpi-intel-hda-match.c | 18 +- sound/soc/intel/common/soc-acpi-intel-icl-match.c | 6 - sound/soc/intel/common/soc-acpi-intel-kbl-match.c | 11 - sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 104 + sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 145 + sound/soc/intel/common/soc-acpi-intel-skl-match.c | 5 - sound/soc/intel/skylake/Makefile | 15 - sound/soc/intel/skylake/bxt-sst.c | 629 ---- sound/soc/intel/skylake/cnl-sst-dsp.c | 266 -- sound/soc/intel/skylake/cnl-sst-dsp.h | 103 - sound/soc/intel/skylake/cnl-sst.c | 508 --- sound/soc/intel/skylake/skl-debug.c | 248 -- sound/soc/intel/skylake/skl-i2s.h | 87 - sound/soc/intel/skylake/skl-messages.c | 1419 -------- sound/soc/intel/skylake/skl-nhlt.c | 269 -- sound/soc/intel/skylake/skl-pcm.c | 1507 -------- sound/soc/intel/skylake/skl-ssp-clk.c | 428 --- sound/soc/intel/skylake/skl-ssp-clk.h | 108 - sound/soc/intel/skylake/skl-sst-cldma.c | 373 -- sound/soc/intel/skylake/skl-sst-cldma.h | 243 -- sound/soc/intel/skylake/skl-sst-dsp.c | 462 --- sound/soc/intel/skylake/skl-sst-dsp.h | 256 -- sound/soc/intel/skylake/skl-sst-ipc.c | 1071 ------ sound/soc/intel/skylake/skl-sst-ipc.h | 169 - sound/soc/intel/skylake/skl-sst-utils.c | 425 --- sound/soc/intel/skylake/skl-sst.c | 599 ---- sound/soc/intel/skylake/skl-topology.c | 3605 -------------------- sound/soc/intel/skylake/skl-topology.h | 524 --- sound/soc/intel/skylake/skl.c | 1177 ------- sound/soc/intel/skylake/skl.h | 207 -- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/loongson/loongson_card.c | 125 +- sound/soc/loongson/loongson_dma.c | 25 +- sound/soc/loongson/loongson_i2s.c | 115 +- sound/soc/loongson/loongson_i2s.h | 24 +- sound/soc/loongson/loongson_i2s_pci.c | 51 +- sound/soc/mediatek/Kconfig | 20 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 16 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 9 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2 +- .../mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 4 +- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 2 +- sound/soc/mediatek/mt8365/Makefile | 15 + sound/soc/mediatek/mt8365/mt8365-afe-clk.c | 421 +++ sound/soc/mediatek/mt8365/mt8365-afe-clk.h | 32 + sound/soc/mediatek/mt8365/mt8365-afe-common.h | 448 +++ sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 2274 ++++++++++++ sound/soc/mediatek/mt8365/mt8365-dai-adda.c | 311 ++ sound/soc/mediatek/mt8365/mt8365-dai-dmic.c | 310 ++ sound/soc/mediatek/mt8365/mt8365-dai-i2s.c | 846 +++++ sound/soc/mediatek/mt8365/mt8365-dai-pcm.c | 293 ++ sound/soc/mediatek/mt8365/mt8365-mt6357.c | 343 ++ sound/soc/mediatek/mt8365/mt8365-reg.h | 993 ++++++ sound/soc/meson/aiu-fifo.h | 2 - sound/soc/meson/aiu.c | 2 +- sound/soc/meson/axg-card.c | 13 +- sound/soc/meson/gx-card.c | 3 +- sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 2 +- sound/soc/qcom/common.c | 1 - sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sm8250.c | 8 + sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/arndale.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 2 +- sound/soc/sdw_utils/Kconfig | 6 + sound/soc/sdw_utils/Makefile | 11 + .../soc_sdw_bridge_cs35l56.c} | 56 +- .../soc_sdw_cs42l42.c} | 13 +- .../soc_sdw_cs42l43.c} | 38 +- .../soc_sdw_cs_amp.c} | 18 +- .../sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} | 10 +- .../sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} | 56 +- .../soc_sdw_rt5682.c} | 12 +- .../sof_sdw_rt700.c => sdw_utils/soc_sdw_rt700.c} | 12 +- .../sof_sdw_rt711.c => sdw_utils/soc_sdw_rt711.c} | 38 +- .../soc_sdw_rt712_sdca.c} | 10 +- .../soc_sdw_rt722_sdca.c} | 10 +- .../soc_sdw_rt_amp.c} | 43 +- .../soc_sdw_rt_amp_coeff_tables.h} | 6 +- .../soc_sdw_rt_dmic.c} | 11 +- .../soc_sdw_rt_sdca_jack_common.c} | 40 +- sound/soc/sdw_utils/soc_sdw_utils.c | 1170 +++++++ sound/soc/sh/Kconfig | 1 + sound/soc/sh/fsi.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/rcar/adg.c | 4 +- sound/soc/sh/rcar/core.c | 19 +- sound/soc/sh/rcar/dma.c | 75 +- sound/soc/sh/rcar/rsnd.h | 10 - sound/soc/sh/rcar/ssi.c | 2 +- sound/soc/sh/rz-ssi.c | 259 +- sound/soc/sh/siu_dai.c | 2 +- sound/soc/soc-ac97.c | 4 +- sound/soc/soc-core.c | 21 +- sound/soc/soc-dai.c | 58 - sound/soc/soc-dapm.c | 5 +- sound/soc/soc-pcm.c | 362 +- sound/soc/soc-topology-test.c | 132 +- sound/soc/soc-topology.c | 9 +- sound/soc/sof/amd/Kconfig | 11 + sound/soc/sof/amd/Makefile | 4 +- sound/soc/sof/amd/acp-common.c | 3 + sound/soc/sof/amd/acp-dsp-offset.h | 24 +- sound/soc/sof/amd/acp-loader.c | 2 +- sound/soc/sof/amd/acp.c | 88 +- sound/soc/sof/amd/acp.h | 11 +- sound/soc/sof/amd/acp70.c | 142 + sound/soc/sof/amd/pci-acp63.c | 2 +- sound/soc/sof/amd/pci-acp70.c | 112 + sound/soc/sof/amd/pci-rmb.c | 1 - sound/soc/sof/amd/pci-rn.c | 1 - sound/soc/sof/amd/pci-vangogh.c | 1 - sound/soc/sof/imx/imx8.c | 2 +- sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/imx/imx8ulp.c | 2 +- sound/soc/sof/intel/Kconfig | 17 + sound/soc/sof/intel/Makefile | 2 + sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 2 +- sound/soc/sof/intel/hda-dsp.c | 1 + sound/soc/sof/intel/hda-stream.c | 4 +- sound/soc/sof/intel/hda.c | 282 +- sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/intel/lnl.c | 27 + sound/soc/sof/intel/mtl.c | 16 +- sound/soc/sof/intel/mtl.h | 2 + sound/soc/sof/intel/pci-ptl.c | 77 + sound/soc/sof/intel/shim.h | 1 + sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- sound/soc/sof/pcm.c | 2 +- sound/soc/sof/sof-audio.h | 8 +- sound/soc/sof/sof-priv.h | 16 +- sound/soc/sof/topology.c | 8 +- sound/soc/sprd/sprd-mcdt.c | 2 +- sound/soc/starfive/jh7110_pwmdac.c | 2 +- sound/soc/starfive/jh7110_tdm.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/stm/stm32_i2s.c | 6 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/stm/stm32_spdifrx.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra186_asrc.c | 2 +- sound/soc/tegra/tegra186_dspk.c | 2 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra210_admaif.c | 2 +- sound/soc/tegra/tegra210_adx.c | 2 +- sound/soc/tegra/tegra210_ahub.c | 2 +- sound/soc/tegra/tegra210_amx.c | 2 +- sound/soc/tegra/tegra210_dmic.c | 2 +- sound/soc/tegra/tegra210_i2s.c | 18 +- sound/soc/tegra/tegra210_mixer.c | 2 +- sound/soc/tegra/tegra210_mvc.c | 2 +- sound/soc/tegra/tegra210_ope.c | 2 +- sound/soc/tegra/tegra210_sfc.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 2 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/tegra/tegra_audio_graph_card.c | 2 +- sound/soc/tegra/tegra_pcm.c | 2 +- sound/soc/ti/ams-delta.c | 2 +- sound/soc/ti/davinci-i2s.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-pxs2.c | 2 +- sound/soc/uniphier/evea.c | 2 +- sound/soc/ux500/mop500.c | 2 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- sound/soc/xilinx/xlnx_spdif.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- sound/usb/line6/driver.c | 5 + sound/usb/quirks-table.h | 4 + sound/usb/stream.c | 4 +- tools/build/feature/Makefile | 53 +- tools/perf/Documentation/Build.txt | 28 + tools/perf/Makefile.config | 20 +- tools/perf/Makefile.perf | 27 +- .../pmu-events/arch/riscv/andes/ax45/firmware.json | 2 +- .../pmu-events/arch/riscv/riscv-sbi-firmware.json | 2 +- .../pmu-events/arch/riscv/sifive/u74/firmware.json | 2 +- .../arch/riscv/starfive/dubhe-80/firmware.json | 2 +- .../arch/riscv/thead/c900-legacy/firmware.json | 2 +- tools/perf/util/callchain.c | 2 +- tools/sound/dapm-graph | 44 +- tools/testing/selftests/bpf/Makefile | 2 +- .../selftests/bpf/prog_tests/uprobe_syscall.c | 4 +- tools/testing/selftests/dmabuf-heaps/dmabuf-heap.c | 4 +- tools/testing/selftests/drivers/net/hw/rss_ctx.py | 37 +- tools/testing/selftests/hid/hid_bpf.c | 26 + tools/testing/selftests/hid/progs/hid.c | 2 +- .../testing/selftests/hid/progs/hid_bpf_helpers.h | 2 +- tools/testing/selftests/kselftest/ksft.py | 2 +- tools/testing/selftests/kvm/riscv/get-reg-list.c | 8 +- tools/testing/selftests/mm/Makefile | 2 +- tools/testing/selftests/mm/mremap_test.c | 2 + tools/testing/selftests/net/mptcp/mptcp_connect.c | 8 +- tools/testing/selftests/net/mptcp/mptcp_join.sh | 158 +- tools/testing/selftests/seccomp/seccomp_bpf.c | 2 + virt/kvm/Kconfig | 4 +- virt/kvm/guest_memfd.c | 227 +- virt/kvm/kvm_main.c | 49 +- 984 files changed, 25292 insertions(+), 30662 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cirrus,cs4271.yaml delete mode 100644 Documentation/devicetree/bindings/sound/cs4271.txt delete mode 100644 Documentation/devicetree/bindings/sound/da7213.txt create mode 100644 Documentation/devicetree/bindings/sound/dlg,da7213.yaml create mode 100644 Documentation/devicetree/bindings/sound/fsl,imx-audio-es8328.yaml create mode 100644 Documentation/devicetree/bindings/sound/fsl,saif.yaml delete mode 100644 Documentation/devicetree/bindings/sound/imx-audio-es8328.txt create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8365-afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/mediatek,mt8365-mt6357.yaml delete mode 100644 Documentation/devicetree/bindings/sound/mxs-saif.txt delete mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 Documentation/devicetree/bindings/sound/qcom,apq8016-sbc-sndcard.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,pcm512x.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,tlv320dac3100.yaml create mode 100644 Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml delete mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt delete mode 100644 Documentation/devicetree/bindings/sound/tpa6130a2.txt delete mode 100644 arch/s390/kernel/alternative.h create mode 100644 include/sound/soc_sdw_utils.h create mode 100644 include/sound/tas2563-tlv.h create mode 100644 sound/soc/amd/acp/acp-sdw-sof-mach.c create mode 100644 sound/soc/amd/acp/amd-acp63-acpi-match.c create mode 100644 sound/soc/amd/acp/soc_amd_sdw_common.h create mode 100644 sound/soc/codecs/mt6357.c create mode 100644 sound/soc/codecs/mt6357.h delete mode 100644 sound/soc/intel/boards/bxt_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/bxt_rt298.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98357a.c delete mode 100644 sound/soc/intel/boards/kbl_da7219_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5660.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_max98927.c delete mode 100644 sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c delete mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c delete mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h delete mode 100644 sound/soc/intel/boards/skl_nau88l25_max98357a.c delete mode 100644 sound/soc/intel/boards/skl_nau88l25_ssm4567.c delete mode 100644 sound/soc/intel/boards/skl_rt286.c create mode 100644 sound/soc/intel/common/soc-acpi-intel-ptl-match.c delete mode 100644 sound/soc/intel/skylake/Makefile delete mode 100644 sound/soc/intel/skylake/bxt-sst.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/cnl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/cnl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-debug.c delete mode 100644 sound/soc/intel/skylake/skl-i2s.h delete mode 100644 sound/soc/intel/skylake/skl-messages.c delete mode 100644 sound/soc/intel/skylake/skl-nhlt.c delete mode 100644 sound/soc/intel/skylake/skl-pcm.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.c delete mode 100644 sound/soc/intel/skylake/skl-ssp-clk.h delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c delete mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c delete mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c delete mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h delete mode 100644 sound/soc/intel/skylake/skl-sst-utils.c delete mode 100644 sound/soc/intel/skylake/skl-sst.c delete mode 100644 sound/soc/intel/skylake/skl-topology.c delete mode 100644 sound/soc/intel/skylake/skl-topology.h delete mode 100644 sound/soc/intel/skylake/skl.c delete mode 100644 sound/soc/intel/skylake/skl.h create mode 100644 sound/soc/mediatek/mt8365/Makefile create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-clk.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-clk.h create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-common.h create mode 100644 sound/soc/mediatek/mt8365/mt8365-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-adda.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-dmic.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-mt6357.c create mode 100644 sound/soc/mediatek/mt8365/mt8365-reg.h create mode 100644 sound/soc/sdw_utils/Kconfig create mode 100644 sound/soc/sdw_utils/Makefile rename sound/soc/{intel/boards/bridge_cs35l56.c => sdw_utils/soc_sdw_bridge_cs35l56.c} (63%) rename sound/soc/{intel/boards/sof_sdw_cs42l42.c => sdw_utils/soc_sdw_cs42l42.c} (82%) rename sound/soc/{intel/boards/sof_sdw_cs42l43.c => sdw_utils/soc_sdw_cs42l43.c} (73%) rename sound/soc/{intel/boards/sof_sdw_cs_amp.c => sdw_utils/soc_sdw_cs_amp.c} (66%) rename sound/soc/{intel/boards/sof_sdw_dmic.c => sdw_utils/soc_sdw_dmic.c} (71%) rename sound/soc/{intel/boards/sof_sdw_maxim.c => sdw_utils/soc_sdw_maxim.c} (63%) rename sound/soc/{intel/boards/sof_sdw_rt5682.c => sdw_utils/soc_sdw_rt5682.c} (82%) rename sound/soc/{intel/boards/sof_sdw_rt700.c => sdw_utils/soc_sdw_rt700.c} (81%) rename sound/soc/{intel/boards/sof_sdw_rt711.c => sdw_utils/soc_sdw_rt711.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt712_sdca.c => sdw_utils/soc_sdw_rt712_sdca.c} (74%) rename sound/soc/{intel/boards/sof_sdw_rt722_sdca.c => sdw_utils/soc_sdw_rt722_sdca.c} (68%) rename sound/soc/{intel/boards/sof_sdw_rt_amp.c => sdw_utils/soc_sdw_rt_amp.c} (84%) rename sound/soc/{intel/boards/sof_sdw_amp_coeff_tables.h => sdw_utils/soc_sdw_rt_amp_coeff_tables.h} (97%) rename sound/soc/{intel/boards/sof_sdw_rt_dmic.c => sdw_utils/soc_sdw_rt_dmic.c} (70%) rename sound/soc/{intel/boards/sof_sdw_rt_sdca_jack_common.c => sdw_utils/soc_sdw_rt_sdca_jack_common.c} (78%) create mode 100644 sound/soc/sdw_utils/soc_sdw_utils.c create mode 100644 sound/soc/sof/amd/acp70.c create mode 100644 sound/soc/sof/amd/pci-acp70.c create mode 100644 sound/soc/sof/intel/pci-ptl.c

2 1

[GIT PULL] ASoC fixes for v6.11-rc7
by Mark Brown 14 Sep '24

14 Sep '24

The following changes since commit 77212f300bfd6fb3edaabd1daf863cabb521854a: ASoC: codecs: lpass-va-macro: set the default codec version for sm8250 (2024-08-30 13:54:24 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc7 for you to fetch changes up to 4f9a71435953f941969a4f017e2357db62d85a86: ASoC: meson: axg-card: fix 'use-after-free' (2024-09-11 16:16:34 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A few last minute fixes, plus an update for Pierre's contact details and status. It'd be good to get these into v6.11 (especially the MAINTAINERS update) but it wouldn't be the end of the world if they waited for the merge window, none of them are super remarkable and it's just a question of timing that they're last minute. ---------------------------------------------------------------- Arseniy Krasnov (1): ASoC: meson: axg-card: fix 'use-after-free' Bard Liao (2): ASoC: Intel: soc-acpi-intel-lnl-match: add missing empty item ASoC: Intel: soc-acpi-intel-mtl-match: add missing empty item Pierre-Louis Bossart (1): MAINTAINERS: update Pierre Bossart's email and role Shenghao Ding (1): ASoC: tas2781: fix to save the dsp bin file name into the correct array in case name_prefix is not NULL Su Hui (1): ASoC: codecs: avoid possible garbage value in peb2466_reg_read() .mailmap | 1 + MAINTAINERS | 6 +++--- sound/soc/codecs/peb2466.c | 3 ++- sound/soc/codecs/tas2781-i2c.c | 2 +- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 1 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 1 + sound/soc/meson/axg-card.c | 3 ++- 7 files changed, 11 insertions(+), 6 deletions(-)

2 1

[PATCH v3] ASoC: tas2781: Add Calibration Kcontrols for Chromebook
by Shenghao Ding 13 Sep '24

13 Sep '24

Add calibration related kcontrol for speaker impedance calibration and speaker leakage check for Chromebook. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Add missing grammatical period - Refer to the functions as func() in comments. - Create a macro for calibration register setting to reduce the line - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409030627.SFdj22Xs-lkp@intel.com/ v2: - Add channel no into the log for error debug - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - renamae tasdevice_chn_switch to put it in a single line - With inverted check this entire function becomes neater in tasdev_chn_switch. - Add registers for TAS2563 and TAS2781 calibration. - Add cali_data_backup for regsiter restore after calibration. - Add is_user_space_calidata to store the flag where the calibrated from, user space or bin file from driver parsing. - tasdevice_chn_switch for chip switch among multiple chips(tas2563 or tas2781) - Add loading the calibrated values from user space in tasdev_load_calibrated_data - Correct no dsp no work, it can still work in bypass mode. - Add calibrated register setting for tas2563&tas2781. - Define tas2781_cali_control and tas2563_cali_control for calibration --- include/sound/tas2781.h | 68 +++ sound/soc/codecs/tas2781-comlib.c | 26 + sound/soc/codecs/tas2781-fmwlib.c | 60 +- sound/soc/codecs/tas2781-i2c.c | 887 +++++++++++++++++++++++++++++- 4 files changed, 1030 insertions(+), 11 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index dbda552398b5..8cd6da0480b7 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -49,12 +49,59 @@ /*I2C Checksum */ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +/* XM_340 */ +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x63, 0x3c) +/* XM_341 */ +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x63, 0x38) + /* Volume control */ #define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) +#define TAS2563_IDLE TASDEVICE_REG(0x00, 0x00, 0x3e) +#define TAS2563_PRM_R0_REG TASDEVICE_REG(0x00, 0x0f, 0x34) + +#define TAS2563_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x02, 0x70) +#define TAS2563_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x02, 0x48) + +#define TAS2563_PRM_ENFF_REG TASDEVICE_REG(0x00, 0x0d, 0x54) +#define TAS2563_PRM_DISTCK_REG TASDEVICE_REG(0x00, 0x0d, 0x58) +#define TAS2563_PRM_TE_SCTHR_REG TASDEVICE_REG(0x00, 0x0f, 0x60) +#define TAS2563_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x0d, 0x74) +#define TAS2563_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x0d, 0x7c) +/* prm_Int_B0 */ +#define TAS2563_TE_TA1_REG TASDEVICE_REG(0x00, 0x10, 0x0c) +/* prm_Int_A1 */ +#define TAS2563_TE_TA1_AT_REG TASDEVICE_REG(0x00, 0x10, 0x10) +/* prm_TE_Beta */ +#define TAS2563_TE_TA2_REG TASDEVICE_REG(0x00, 0x0f, 0x64) +/* prm_TE_Beta1 */ +#define TAS2563_TE_AT_REG TASDEVICE_REG(0x00, 0x0f, 0x68) +/* prm_TE_1_Beta1 */ +#define TAS2563_TE_DT_REG TASDEVICE_REG(0x00, 0x0f, 0x70) + +#define TAS2781_PRM_INT_MASK_REG TASDEVICE_REG(0x00, 0x00, 0x3b) +#define TAS2781_PRM_CLK_CFG_REG TASDEVICE_REG(0x00, 0x00, 0x5c) +#define TAS2781_PRM_RSVD_REG TASDEVICE_REG(0x00, 0x01, 0x19) +#define TAS2781_PRM_TEST_57_REG TASDEVICE_REG(0x00, 0xfd, 0x39) +#define TAS2781_PRM_TEST_62_REG TASDEVICE_REG(0x00, 0xfd, 0x3e) +#define TAS2781_PRM_PVDD_UVLO_REG TASDEVICE_REG(0x00, 0x00, 0x71) +#define TAS2781_PRM_CHNL_0_REG TASDEVICE_REG(0x00, 0x00, 0x03) +#define TAS2781_PRM_NG_CFG0_REG TASDEVICE_REG(0x00, 0x00, 0x35) +#define TAS2781_PRM_IDLE_CH_DET_REG TASDEVICE_REG(0x00, 0x00, 0x66) +#define TAS2781_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x14, 0x38) +#define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) +#define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) + +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) +#define TAS2781_TEST_PAGE_UNLOCK 0x0D + +#define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) +#define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) +#define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) + #define TASDEVICE_CMD_SING_W 0x1 #define TASDEVICE_CMD_BURST 0x2 #define TASDEVICE_CMD_DELAY 0x3 @@ -70,7 +117,15 @@ enum device_catlog_id { OTHERS }; +struct bulk_reg_val { + int reg; + unsigned char val[4]; + unsigned char val_len; + bool is_locked; +}; + struct tasdevice { + struct bulk_reg_val *cali_data_backup; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -81,9 +136,19 @@ struct tasdevice { bool is_loaderr; }; +struct cali_reg { + unsigned int r0_reg; + unsigned int r0_low_reg; + unsigned int invr0_reg; + unsigned int pow_reg; + unsigned int tlimit_reg; +}; + struct calidata { unsigned char *data; unsigned long total_sz; + struct cali_reg cali_reg_array; + unsigned int cali_dat_sz_per_dev; }; struct tasdevice_priv { @@ -119,6 +184,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool is_user_space_calidata; unsigned int global_addr; int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, @@ -145,6 +211,8 @@ int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 664c371796d6..1e0b3aa95749 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -88,6 +88,32 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, return ret; } +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdev_chn_switch); + int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *val) { diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index f3a7605f0710..3de0132c345d 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2151,20 +2151,61 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *data = cali_data->data; struct tasdevice_calibration *cal; - struct tasdevice_fw *cal_fmw; + int k = i * (cali_data->cali_dat_sz_per_dev + 1); + int rc; - cal_fmw = priv->tasdevice[i].cali_data_fmw; + /* Load the calibrated data from cal bin file */ + if (!priv->is_user_space_calidata && cal_fmw) { + cal = cal_fmw->calibrations; - /* No calibrated data for current devices, playback will go ahead. */ - if (!cal_fmw) + if (cal) + load_calib_data(priv, &cal->dev_data); return; - - cal = cal_fmw->calibrations; - if (!cal) + } + if (!priv->is_user_space_calidata) + return; + /* load calibrated data from user space */ + if (data[k] != i) { + dev_err(priv->dev, "%s: no cal-data for dev %d from usr-spc\n", + __func__, i); return; + } + k++; - load_calib_data(priv, &cal->dev_data); + rc = tasdevice_dev_bulk_write(priv, i, p->r0_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_reg bulk_wr err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->r0_low_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_low_reg err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->invr0_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d invr0_reg err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->pow_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d pow_reg bulk_wr err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->tlimit_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d tlimit_reg err = %d\n", i, rc); + return; + } } int tasdevice_select_tuningprm_cfg(void *context, int prm_no, @@ -2259,9 +2300,10 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tas_priv->tasdevice[i].cur_conf = cfg_no; } } - } else + } else { dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", __func__, cfg_no); + } status |= cfg_info[rca_conf_no]->active_dev; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 3bc10d615fd3..d0ba7cbe03a8 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -33,6 +33,69 @@ #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> +#define X2563_CL_STT_VAL(xreg, xval) \ +{ .reg = xreg, \ + .val = { xval }, \ + .val_len = 1, } + +#define X2563_CL_STT_4BYTS(xreg, byte0, byte1, byte2, byte3) \ +{ .reg = xreg, \ + .val = { byte0, byte1, byte2, byte3 }, \ + .val_len = 4, } + +static const struct bulk_reg_val tas2563_cali_start_reg[] = { + X2563_CL_STT_VAL(TAS2563_IDLE, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_ENFF_REG, 0x40, 0x00, 0x00, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_DISTCK_REG, 0x40, 0x00, 0x00, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_TE_SCTHR_REG, 0x7f, 0xff, 0xff, 0xff), + X2563_CL_STT_4BYTS(TAS2563_PRM_PLT_FLAG_REG, 0x40, 0x00, 0x00, 0x00), + X2563_CL_STT_4BYTS(TAS2563_PRM_SINEGAIN_REG, 0x0a, 0x3d, 0x70, 0xa4), + X2563_CL_STT_4BYTS(TAS2563_TE_TA1_REG, 0x00, 0x36, 0x91, 0x5e), + X2563_CL_STT_4BYTS(TAS2563_TE_TA1_AT_REG, 0x00, 0x36, 0x91, 0x5e), + X2563_CL_STT_4BYTS(TAS2563_TE_TA2_REG, 0x00, 0x06, 0xd3, 0x72), + X2563_CL_STT_4BYTS(TAS2563_TE_AT_REG, 0x00, 0x36, 0x91, 0x5e), + X2563_CL_STT_4BYTS(TAS2563_TE_DT_REG, 0x00, 0x36, 0x91, 0x5e), +}; + +#define X2781_CL_STT_VAL(xreg, xval, xlocked) \ +{ .reg = xreg, \ + .val = { xval }, \ + .val_len = 1, \ + .is_locked = xlocked, } + +#define X2781_CL_STT_4BYTS_UNLOCKED(xreg, byte0, byte1, byte2, byte3) \ +{ .reg = xreg, \ + .val = { byte0, byte1, byte2, byte3 }, \ + .val_len = 4, \ + .is_locked = false, } + +#define X2781_CL_STT_LEN_UNLOCKED(xreg) \ +{ .reg = xreg, \ + .val_len = 4, \ + .is_locked = false, } + +static const struct bulk_reg_val tas2781_cali_start_reg[] = { + X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false), + X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false), + X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true), + X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false), + X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false), + X2781_CL_STT_VAL(TAS2781_PRM_NG_CFG0_REG, 0xb9, false), + X2781_CL_STT_VAL(TAS2781_PRM_IDLE_CH_DET_REG, 0x92, false), + /* + * This register is pilot tone threshold, different with the + * calibration tool version, it will be updated in + * tas2781_calib_start_put(), set to 1mA. + */ + X2781_CL_STT_4BYTS_UNLOCKED(0, 0x00, 0x00, 0x00, 0x56), + X2781_CL_STT_4BYTS_UNLOCKED(TAS2781_PRM_PLT_FLAG_REG, + 0x40, 0x00, 0x00, 0x00), + X2781_CL_STT_LEN_UNLOCKED(TAS2781_PRM_SINEGAIN_REG), + X2781_CL_STT_LEN_UNLOCKED(TAS2781_PRM_SINEGAIN2_REG), +}; + static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, @@ -141,6 +204,557 @@ static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned char *data = cali_data->data; + unsigned int i = 0; + unsigned int j, k; + int rc; + + guard(mutex)(&priv->codec_lock); + if (!priv->is_user_space_calidata) + return -1; + + if (!p->r0_reg) + return -1; + + dst[i++] = bytes_ext->max; + dst[i++] = 'r'; + + dst[i++] = TASDEVICE_BOOK_ID(p->r0_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->r0_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->r0_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->r0_low_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->r0_low_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->r0_low_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->invr0_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->invr0_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->invr0_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->pow_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->pow_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->pow_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->tlimit_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->tlimit_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->tlimit_reg); + + for (j = 0, k = 0; j < priv->ndev; j++) { + if (j == data[k]) { + dst[i++] = j; + k++; + } else { + dev_err(priv->dev, "chn %d device %u not match\n", + j, data[k]); + k += 21; + continue; + } + rc = tasdevice_dev_bulk_read(priv, j, p->r0_reg, &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_reg bulk_rd err = %d\n", + j, rc); + i += 20; + k += 20; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d r0_data is not same\n", j); + k += 4; + i += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->r0_low_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_low bulk_rd err = %d\n", + j, rc); + i += 16; + k += 16; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d r0_low is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->invr0_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d invr0 bulk_rd err = %d\n", + j, rc); + i += 12; + k += 12; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d invr0 is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->pow_reg, &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d pow_reg bulk_rd err = %d\n", + j, rc); + i += 8; + k += 8; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d pow_reg is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->tlimit_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d tlimit bulk_rd err = %d\n", + j, rc); + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d tlimit is not same\n", j); + i += 4; + k += 4; + } + return 0; +} + +static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, + unsigned char *dst) +{ + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int rc = -1; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + rc = tasdevice_dev_bulk_read(tas_priv, i, reg, &dst[1], + 4); + break; + } + } + + return rc; +} + +static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, + int *reg, unsigned char *dat) +{ + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int j; + + if (p == NULL) + return; + + /* Store the current setting from the chip */ + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_read(tas_priv, i, p[j].reg, + (int *)&p[j].val[0]); + } else { + switch (p[j].reg) { + case 0: { + if (!reg[0]) + continue; + p[j].reg = reg[0]; + } + break; + case TAS2781_PRM_PLT_FLAG_REG: + p[j].reg = reg[1]; + break; + case TAS2781_PRM_SINEGAIN_REG: + p[j].reg = reg[2]; + break; + case TAS2781_PRM_SINEGAIN2_REG: + p[j].reg = reg[3]; + break; + } + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } + + /* Update the setting for calibration */ + for (j = 0; j < sum - 2; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + tas2781_cali_start_reg[j].val[0]); + } else { + if (!p[j].reg) + continue; + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + (unsigned char *) + tas2781_cali_start_reg[j].val, 4); + } + } + + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, &dat[1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, &dat[5], 4); +} + +static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dat = ucontrol->value.bytes.data; + int i, reg[4]; + int j = 0; + + guard(mutex)(&priv->codec_lock); + if (priv->chip_id != TAS2781 || bytes_ext->max != dat[0] || + dat[1] != 'r') { + dev_err(priv->dev, "%s: package fmt or chipid incorrect\n", + __func__); + return 0; + } + j += 2; + /* refresh pilot tone and SineGain register */ + for (i = 0; i < ARRAY_SIZE(reg); i++) { + reg[i] = TASDEVICE_REG(dat[j], dat[j + 1], dat[j + 2]); + j += 3; + } + + for (i = 0; i < priv->ndev; i++) { + int k = i * 9 + j; + + if (dat[k] != i) { + dev_err(priv->dev, "%s:no cal-setting for dev %d\n", + __func__, i); + continue; + } + sngl_calib_start(priv, i, reg, dat + k); + } + return 1; +} + +static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + } else { + if (!p[j].reg) + continue; + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } + } +} + +static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct bulk_reg_val *q = (struct bulk_reg_val *)tas2563_cali_start_reg; + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int rc = 1; + int i, j; + + guard(mutex)(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2563) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_read(tas_priv, + i, p[j].reg, + (unsigned int *)&p[j].val[0]); + else + tasdevice_dev_bulk_read(tas_priv, + i, p[j].reg, p[j].val, 4); + } + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + q[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + q[j].val, 4); + } + } +out: + return rc; +} + +static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tasdev_calib_stop_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + + guard(mutex)(&priv->codec_lock); + if (priv->chip_id == TAS2563) + tas2563_calib_stop_put(priv); + else + tas2781_calib_stop_put(priv); + + return 1; +} + +static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *src = ucontrol->value.bytes.data; + unsigned char *dst = cali_data->data; + int rc = 1, i = 0; + int j; + + guard(mutex)(&priv->codec_lock); + if (src[0] != bytes_ext->max || src[1] != 'r') { + dev_err(priv->dev, "%s: pkg fmt invalid\n", __func__); + return 0; + } + for (j = 0; j < priv->ndev; j++) { + if (src[17 + j * 21] != j) { + dev_err(priv->dev, "%s: pkg fmt invalid\n", __func__); + return 0; + } + } + i += 2; + priv->is_user_space_calidata = true; + + p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + + memcpy(dst, &src[i], cali_data->total_sz); + return rc; +} + +static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct tasdevice *tasdev = tas_priv->tasdevice; + unsigned char *dst = ucontrol->value.bytes.data; + int i, val, rc = -1; + + dst[0] = bytes_ext->max; + guard(mutex)(&tas_priv->codec_lock); + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[1] = i; + rc = tasdevice_dev_read(tas_priv, i, + TAS2781_RUNTIME_LATCH_RE_REG, &val); + if (rc < 0) + dev_err(tas_priv->dev, "%s, get value error\n", + __func__); + else + dst[2] = val; + + break; + } + } + + return rc; +} + +static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG_TF; + else + reg = TAS2563_RUNTIME_RE_REG_TF; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG; + else + reg = TAS2563_RUNTIME_RE_REG; + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct calidata *cali_data = &tas_priv->cali_data; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + + if (tas_priv->chip_id == TAS2563) + reg = TAS2563_PRM_R0_REG; + else if (cali_data->cali_reg_array.r0_reg) + reg = cali_data->cali_reg_array.r0_reg; + else + return -1; + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A1_REG; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A2_REG; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_nop_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + static int tas2563_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -241,6 +855,16 @@ static const struct snd_kcontrol_new tasdevice_snd_controls[] = { tasdev_force_fwload_get, tasdev_force_fwload_put), }; +static const struct snd_kcontrol_new tasdevice_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tasdev_calib_stop_put), + SND_SOC_BYTES_EXT("Amp TF Data", 6, tasdev_tf_data_get, NULL), + SND_SOC_BYTES_EXT("Amp RE Data", 6, tasdev_re_data_get, NULL), + SND_SOC_BYTES_EXT("Amp R0 Data", 6, tasdev_r0_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA1 Data", 6, tasdev_XMA1_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -250,6 +874,10 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, dvc_tlv), }; +static const struct snd_kcontrol_new tas2781_cali_controls[] = { + SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), +}; + static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, @@ -257,6 +885,11 @@ static const struct snd_kcontrol_new tas2563_snd_controls[] = { tas2563_dvc_tlv), }; +static const struct snd_kcontrol_new tas2563_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Start", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tas2563_calib_start_put), +}; + static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -274,6 +907,31 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, return ret; } +static int tasdevice_info_active_num(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->ndev - 1; + + return 0; +} + +static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = TAS2563; + uinfo->value.integer.max = TAS2781; + + return 0; +} + static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -330,6 +988,17 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, return 0; } +static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->chip_id; + + return 0; +} + static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; @@ -421,11 +1090,47 @@ static int tasdevice_configuration_put( return ret; } +static int tasdevice_active_num_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + ucontrol->value.integer.value[0] = i; + return 0; + } + } + + return -1; +} + +static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int dev_id = ucontrol->value.integer.value[0]; + int max = tas_priv->ndev - 1, rc; + + dev_id = clamp(dev_id, 0, max); + + guard(mutex)(&tas_priv->codec_lock); + rc = tasdev_chn_switch(tas_priv, dev_id); + + return rc; +} + static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *prog_name, *conf_name; - int nr_controls = 2; + char *active_dev_num, *chip_id; + char *conf_name, *prog_name; + int nr_controls = 4; int mix_index = 0; int ret; @@ -466,6 +1171,30 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_configuration_put; mix_index++; + active_dev_num = devm_kstrdup(tas_priv->dev, "Activate Tasdevice Num", + GFP_KERNEL); + if (!active_dev_num) { + ret = -ENOMEM; + goto out; + } + dsp_ctrls[mix_index].name = active_dev_num; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_active_num; + dsp_ctrls[mix_index].get = tasdevice_active_num_get; + dsp_ctrls[mix_index].put = tasdevice_active_num_put; + mix_index++; + + chip_id = devm_kstrdup(tas_priv->dev, "Tasdevice Chip Id", GFP_KERNEL); + if (!chip_id) { + ret = -ENOMEM; + goto out; + } + dsp_ctrls[mix_index].name = chip_id; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_chip_id; + dsp_ctrls[mix_index].get = tasdevice_get_chip_id; + mix_index++; + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); @@ -473,6 +1202,149 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) return ret; } +static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) +{ + struct calidata *cali_data = &priv->cali_data; + struct tasdevice *tasdev = priv->tasdevice; + struct soc_bytes_ext *ext_cali_data; + struct snd_kcontrol_new *cali_ctrls; + unsigned int nctrls; + char *cali_name; + int rc, i; + + rc = snd_soc_add_component_controls(priv->codec, + tasdevice_cali_controls, ARRAY_SIZE(tasdevice_cali_controls)); + if (rc < 0) { + dev_err(priv->dev, "%s: Add cali controls err rc = %d", + __func__, rc); + return rc; + } + + if (priv->chip_id == TAS2781) { + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; + nctrls = ARRAY_SIZE(tas2781_cali_controls); + for (i = 0; i < priv->ndev; i++) { + tasdev[i].cali_data_backup = + kmemdup(tas2781_cali_start_reg, + sizeof(tas2781_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_backup) + return -ENOMEM; + } + } else { + cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; + nctrls = ARRAY_SIZE(tas2563_cali_controls); + for (i = 0; i < priv->ndev; i++) { + tasdev[i].cali_data_backup = + kmemdup(tas2563_cali_start_reg, + sizeof(tas2563_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_backup) + return -ENOMEM; + } + } + + rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, nctrls); + if (rc < 0) { + dev_err(priv->dev, "%s: Add chip cali ctrls err rc = %d", + __func__, rc); + return rc; + } + + /* index for cali_ctrls */ + i = 0; + if (priv->chip_id == TAS2781) + nctrls = 2; + else + nctrls = 1; + + /* + * Alloc kcontrol via devm_kzalloc(), which don't manually + * free the kcontrol。 + */ + cali_ctrls = devm_kcalloc(priv->dev, nctrls, + sizeof(cali_ctrls[0]), GFP_KERNEL); + if (!cali_ctrls) + return -ENOMEM; + + ext_cali_data = devm_kzalloc(priv->dev, sizeof(*ext_cali_data), + GFP_KERNEL); + if (!ext_cali_data) + return -ENOMEM; + + cali_name = devm_kstrdup(priv->dev, "Speaker Calibrated Data", + GFP_KERNEL); + if (!cali_name) + return -ENOMEM; + /* the number of calibrated data per tas2563/tas2781 */ + cali_data->cali_dat_sz_per_dev = 20; + /* + * Data structure for tas2563/tas2781 calibrated data: + * Pkg len (1 byte) + * Reg id (1 byte, constant 'r') + * book, page, register array for calibrated data (15 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info (1 byte) + * Calibrated data for Device #i (20 bytes) + * } + */ + ext_cali_data->max = priv->ndev * + (cali_data->cali_dat_sz_per_dev + 1) + 1 + 15 + 1; + priv->cali_data.total_sz = priv->ndev * + (cali_data->cali_dat_sz_per_dev + 1); + priv->cali_data.data = devm_kzalloc(priv->dev, + ext_cali_data->max, GFP_KERNEL); + cali_ctrls[i].name = cali_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].get = tasdev_cali_data_get; + cali_ctrls[i].put = tasdev_cali_data_put; + cali_ctrls[i].private_value = (unsigned long)ext_cali_data; + i++; + + cali_data->data = devm_kzalloc(priv->dev, cali_data->total_sz, + GFP_KERNEL); + if (!cali_data->data) + return -ENOMEM; + + if (priv->chip_id == TAS2781) { + struct soc_bytes_ext *ext_cali_start; + char *cali_start_name; + + ext_cali_start = devm_kzalloc(priv->dev, + sizeof(*ext_cali_start), GFP_KERNEL); + if (!ext_cali_start) + return -ENOMEM; + + cali_start_name = devm_kstrdup(priv->dev, + "Calibration Start", GFP_KERNEL); + if (!cali_start_name) + return -ENOMEM; + /* + * package structure for tas2781 ftc start: + * Pkg len (1 byte) + * Reg id (1 byte, constant 'r') + * book, page, register for pilot threshold, pilot tone + * and sine gain (12 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info (1 byte) + * Sine gain for Device #i (8 bytes) + * } + */ + ext_cali_start->max = 14 + priv->ndev * 9; + cali_ctrls[i].name = cali_start_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].put = tas2781_calib_start_put; + cali_ctrls[i].get = tasdev_nop_get; + cali_ctrls[i].private_value = (unsigned long)ext_cali_start; + i++; + } + + rc = snd_soc_add_component_controls(priv->codec, cali_ctrls, + nctrls < i ? nctrls : i); + + return rc; +} + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { @@ -519,6 +1391,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, goto out; } + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; /* If calibrated data occurs error, dsp will still works with default @@ -720,6 +1598,11 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) static void tasdevice_deinit(void *context) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) + kfree(tasdev[i].cali_data_backup); tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); -- 2.34.1

2 1

[PATCH 0/5] AMD SoundWire machine driver code refactor
by Vijendar Mukunda 13 Sep '24

13 Sep '24

This patch series moves common Soundwire endpoint parsing and dai creation logic to common placeholder from Intel generic SoundWire machine driver code to make it generic. AMD SoundWire machine driver code is refactored to use these functions for SoundWire endpoint parsing and dai creation logic. Link: https://github.com/thesofproject/linux/pull/5171 Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Vijendar Mukunda (5): ASoC: intel: sof_sdw: rename soundwire endpoint and dailink structures ASoC: intel: sof_sdw: rename soundwire parsing helper functions ASoC: sdw_util/intel: move soundwire endpoint and dai link structures ASoC: sdw_utils/intel: move soundwire endpoint parsing helper functions ASoC: amd: acp: refactor SoundWire machine driver code include/sound/soc_sdw_utils.h | 31 ++ sound/soc/amd/acp/acp-sdw-sof-mach.c | 540 ++++++++------------------- sound/soc/intel/boards/sof_sdw.c | 195 +--------- sound/soc/sdw_utils/soc_sdw_utils.c | 161 ++++++++ 4 files changed, 354 insertions(+), 573 deletions(-) -- 2.34.1

2 6

[PATCH v2] ASoC: atmel: mchp-pdmc: Retain Non-Runtime Controls
by Andrei Simion 13 Sep '24

13 Sep '24

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Avoid removing these controls, as doing so can cause issues if the stream is initiated from another control. Ensure these controls remain intact when the stream is started or finished. Instead of removing them, return an -EBUSY error code to indicate that the controller is busy, especially when the audio filter and the SINC filter are in use. [andrei.simion(a)microchip.com: Reword the commit title and the commit message. Replace spinlock and busy variable with atomic_t busy_stream.] Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> --- v1 -> v2: - drop spinlock and u8 busy:1; - use atomic_t variable busy_stream - use atomic_read, atomic_set --- sound/soc/atmel/mchp-pdmc.c | 57 ++++++++++--------------------------- 1 file changed, 15 insertions(+), 42 deletions(-) diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index d97d153ee375..939cd44ebc8a 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -124,6 +124,7 @@ struct mchp_pdmc { int mic_no; int sinc_order; bool audio_filter_en; + atomic_t busy_stream; }; static const char *const mchp_pdmc_sinc_filter_order_text[] = { @@ -167,6 +168,10 @@ static int mchp_pdmc_sinc_order_put(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; + + if (atomic_read(&dd->busy_stream)) + return -EBUSY; + if (val == dd->sinc_order) return 0; @@ -193,6 +198,9 @@ static int mchp_pdmc_af_put(struct snd_kcontrol *kcontrol, struct mchp_pdmc *dd = snd_soc_component_get_drvdata(component); bool af = uvalue->value.integer.value[0] ? true : false; + if (atomic_read(&dd->busy_stream)) + return -EBUSY; + if (dd->audio_filter_en == af) return 0; @@ -379,52 +387,10 @@ static const struct snd_kcontrol_new mchp_pdmc_snd_controls[] = { }, }; -static int mchp_pdmc_close(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - return snd_soc_add_component_controls(component, mchp_pdmc_snd_controls, - ARRAY_SIZE(mchp_pdmc_snd_controls)); -} - -static int mchp_pdmc_open(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - int i; - - /* remove controls that can't be changed at runtime */ - for (i = 0; i < ARRAY_SIZE(mchp_pdmc_snd_controls); i++) { - const struct snd_kcontrol_new *control = &mchp_pdmc_snd_controls[i]; - struct snd_ctl_elem_id id; - int err; - - if (component->name_prefix) - snprintf(id.name, sizeof(id.name), "%s %s", component->name_prefix, - control->name); - else - strscpy(id.name, control->name, sizeof(id.name)); - - id.numid = 0; - id.iface = control->iface; - id.device = control->device; - id.subdevice = control->subdevice; - id.index = control->index; - err = snd_ctl_remove_id(component->card->snd_card, &id); - if (err < 0) - dev_err(component->dev, "%d: Failed to remove %s\n", err, - control->name); - } - - return 0; -} - static const struct snd_soc_component_driver mchp_pdmc_dai_component = { .name = "mchp-pdmc", .controls = mchp_pdmc_snd_controls, .num_controls = ARRAY_SIZE(mchp_pdmc_snd_controls), - .open = &mchp_pdmc_open, - .close = &mchp_pdmc_close, - .legacy_dai_naming = 1, - .trigger_start = SND_SOC_TRIGGER_ORDER_LDC, }; static const unsigned int mchp_pdmc_1mic[] = {1}; @@ -587,6 +553,11 @@ static int mchp_pdmc_hw_params(struct snd_pcm_substream *substream, cfgr_val |= MCHP_PDMC_CFGR_BSSEL(i); } + /* + * from these point forward, we consider the controller busy, so the + * audio filter and SINC order can't be changed + */ + atomic_set(&dd->busy_stream, 1); for (osr_start = dd->audio_filter_en ? 64 : 8; osr_start <= 256 && best_diff_rate; osr_start *= 2) { long round_rate; @@ -1143,6 +1114,8 @@ static void mchp_pdmc_remove(struct platform_device *pdev) { struct mchp_pdmc *dd = platform_get_drvdata(pdev); + atomic_set(&dd->busy_stream, 0); + if (!pm_runtime_status_suspended(dd->dev)) mchp_pdmc_runtime_suspend(dd->dev); base-commit: 5acd9952f95fb4b7da6d09a3be39195a80845eb6 -- 2.34.1

2 1

change getopt_long return variable from char to int
by GitHub pull_request - opened 12 Sep '24

12 Sep '24

alsa-project/alsa-utils pull request #275 was opened from sadguitarius: fixes #274 Request URL : https://github.com/alsa-project/alsa-utils/pull/275 Patch URL : https://github.com/alsa-project/alsa-utils/pull/275.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aseqsend getopt_long variable should be int instead of char
by GitHub issues - opened 12 Sep '24

12 Sep '24

alsa-project/alsa-utils issue #274 was opened from sadguitarius: Hi, just wanted to point out that aseqsend is failing because the value returned by getopt_long can never be -1 since it's a char. Issue URL : https://github.com/alsa-project/alsa-utils/issues/274 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ALSA: usb-audio: Add logitech Audio profile quirk
by Joshua Pius 12 Sep '24

12 Sep '24

Specify shortnames for the following Logitech Devices: Rally bar, Rally bar mini, Tap, MeetUp and Huddle. Signed-off-by: Joshua Pius <joshuapius(a)chromium.org> --- sound/usb/card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 778de9244f1e..9c411b82a218 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -384,6 +384,12 @@ static const struct usb_audio_device_name usb_audio_names[] = { /* Creative/Toshiba Multimedia Center SB-0500 */ DEVICE_NAME(0x041e, 0x3048, "Toshiba", "SB-0500"), + /* Logitech Audio Devices */ + DEVICE_NAME(0x046d, 0x0867, "Logitech, Inc.", "Logi-MeetUp"), + DEVICE_NAME(0x046d, 0x0874, "Logitech, Inc.", "Logi-Tap-Audio"), + DEVICE_NAME(0x046d, 0x087c, "Logitech, Inc.", "Logi-Huddle"), + DEVICE_NAME(0x046d, 0x0898, "Logitech, Inc.", "Logi-RB-Audio"), + DEVICE_NAME(0x046d, 0x08d2, "Logitech, Inc.", "Logi-RBM-Audio"), DEVICE_NAME(0x046d, 0x0990, "Logitech, Inc.", "QuickCam Pro 9000"), DEVICE_NAME(0x05e1, 0x0408, "Syntek", "STK1160"), -- 2.46.0.598.g6f2099f65c-goog

2 1

[PATCH v2 0/3] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 12 Sep '24

12 Sep '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (3): ASoC: sma1307: Add driver for Iron Device SMA1307 ASoC: dt-bindings: irondevice,sma1307: Add initial DT binding doc: ABI: testing: sma1307: Add support for SMA1307 .../ABI/testing/sysfs-bus-i2c-devices-sma1307 | 17 + .../bindings/sound/irondevice,sma1307.yaml | 54 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2191 +++++++++++++++++ sound/soc/codecs/sma1307.h | 454 ++++ 6 files changed, 2726 insertions(+) create mode 100644 Documentation/ABI/testing/sysfs-bus-i2c-devices-sma1307 create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

6 22

[PATCH 0/3] Improvements for mchp-pdmc
by Andrei Simion 12 Sep '24

12 Sep '24

This patch set is intended to enhance the functionality and maintainability of the mchp-pdmc driver: - Enhances performance by refining maxburst logic. - Introduces a name for better identification and management. - Ensures controls remain intact when streams start/finish, returning -EBUSY if the controller is busy. Codrin Ciubotariu (3): ASoC: atmel: mchp-pdmc: Improve maxburst calculation for better performance ASoC: atmel: mchp-pdmc: Add snd_soc_dai_driver name ASoC: atmel: mchp-pdmc: Retain Non-Runtime Controls sound/soc/atmel/mchp-pdmc.c | 118 +++++++++++++++++++----------------- 1 file changed, 63 insertions(+), 55 deletions(-) base-commit: 6708132e80a2ced620bde9b9c36e426183544a23 -- 2.34.1

3 6

ucm2: soundwire: rt722: add ConflictingDevice in each device
by GitHub pull_request - opened 12 Sep '24

12 Sep '24

alsa-project/alsa-ucm-conf pull request #441 was opened from shumingfan: This patch allows only one input and one output to be opened simultaneously. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/441 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/441.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ALSA: usb-audio: Add logitech Audio profile quirk
by Joshua Pius 12 Sep '24

12 Sep '24

Specify shortnames for the following Logitech Devices: Rally bar, Rally bar mini, Tap, MeetUp and Huddle. Signed-off-by: Joshua Pius <joshuapius(a)google.com> --- sound/usb/card.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 778de9244f1e..9c411b82a218 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -384,6 +384,12 @@ static const struct usb_audio_device_name usb_audio_names[] = { /* Creative/Toshiba Multimedia Center SB-0500 */ DEVICE_NAME(0x041e, 0x3048, "Toshiba", "SB-0500"), + /* Logitech Audio Devices */ + DEVICE_NAME(0x046d, 0x0867, "Logitech, Inc.", "Logi-MeetUp"), + DEVICE_NAME(0x046d, 0x0874, "Logitech, Inc.", "Logi-Tap-Audio"), + DEVICE_NAME(0x046d, 0x087c, "Logitech, Inc.", "Logi-Huddle"), + DEVICE_NAME(0x046d, 0x0898, "Logitech, Inc.", "Logi-RB-Audio"), + DEVICE_NAME(0x046d, 0x08d2, "Logitech, Inc.", "Logi-RBM-Audio"), DEVICE_NAME(0x046d, 0x0990, "Logitech, Inc.", "QuickCam Pro 9000"), DEVICE_NAME(0x05e1, 0x0408, "Syntek", "STK1160"), -- 2.46.0.598.g6f2099f65c-goog

3 5

Mismatched volume controls of Sound Blaster X3
by GitHub issues - opened 12 Sep '24

12 Sep '24

alsa-project/alsa-utils issue #273 was opened from buboleck: Hello, Recently I got Creative Sound Blaster X3 (SB1810) as it seemed it works fine with linux. I'm using Gentoo with Plasma desktop (ALSA + Pipewire) and the onboard sound is OK but the SB X3 seems is misbehaving, I found that the volume controls for RearLeft is swapped with Center and RearRight is swapped with the LFE. (Also tried live Ubuntu 24.04 same issue) On the other hand maybe related, I was not able to get stereo upmix working in any way, ALSA upmix does not want to work as also the Pipiwire upmix. Gentoo kernel 6.10.9 alsa-lib-1.2.12 alsa-tools-1.2.11 alsa-utils-1.2.12 alsa-plugins-1.2.12 pyalsa-1.2.12 Issue URL : https://github.com/alsa-project/alsa-utils/issues/273 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Mic mute LED only follows internal rather than selected
by GitHub issues - opened 12 Sep '24

12 Sep '24

alsa-project/alsa-ucm-conf issue #440 was opened from JL2210: I have a laptop keyboard with a key to mute the microphone. Using this key mutes whichever mic is selected as default. However, the light on the key only changes with the internal microphone rather than the active mic. Pressing the key when a headset is plugged in doesn't change the LED, but pressing it without a headset does. [alsa-info.txt](https://github.com/user-attachments/files/16971038/alsa-info… Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/440 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1] ASoC: meson: axg-card: fix 'use-after-free'
by Arseniy Krasnov 11 Sep '24

11 Sep '24

Buffer 'card->dai_link' is reallocated in 'meson_card_reallocate_links()', so move 'pad' pointer initialization after this function when memory is already reallocated. Kasan bug report: ================================================================== BUG: KASAN: slab-use-after-free in axg_card_add_link+0x76c/0x9bc Read of size 8 at addr ffff000000e8b260 by task modprobe/356 CPU: 0 PID: 356 Comm: modprobe Tainted: G O 6.9.12-sdkernel #1 Call trace: dump_backtrace+0x94/0xec show_stack+0x18/0x24 dump_stack_lvl+0x78/0x90 print_report+0xfc/0x5c0 kasan_report+0xb8/0xfc __asan_load8+0x9c/0xb8 axg_card_add_link+0x76c/0x9bc [snd_soc_meson_axg_sound_card] meson_card_probe+0x344/0x3b8 [snd_soc_meson_card_utils] platform_probe+0x8c/0xf4 really_probe+0x110/0x39c __driver_probe_device+0xb8/0x18c driver_probe_device+0x108/0x1d8 __driver_attach+0xd0/0x25c bus_for_each_dev+0xe0/0x154 driver_attach+0x34/0x44 bus_add_driver+0x134/0x294 driver_register+0xa8/0x1e8 __platform_driver_register+0x44/0x54 axg_card_pdrv_init+0x20/0x1000 [snd_soc_meson_axg_sound_card] do_one_initcall+0xdc/0x25c do_init_module+0x10c/0x334 load_module+0x24c4/0x26cc init_module_from_file+0xd4/0x128 __arm64_sys_finit_module+0x1f4/0x41c invoke_syscall+0x60/0x188 el0_svc_common.constprop.0+0x78/0x13c do_el0_svc+0x30/0x40 el0_svc+0x38/0x78 el0t_64_sync_handler+0x100/0x12c el0t_64_sync+0x190/0x194 Fixes: 7864a79f37b5 ("ASoC: meson: add axg sound card support") Cc: <Stable(a)vger.kernel.org> Signed-off-by: Arseniy Krasnov <avkrasnov(a)salutedevices.com> --- sound/soc/meson/axg-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 8c5605c1e34e8..eb0302f207407 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -104,7 +104,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, int *index) { struct meson_card *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_dai_link *pad = &card->dai_link[*index]; + struct snd_soc_dai_link *pad; struct snd_soc_dai_link *lb; struct snd_soc_dai_link_component *dlc; int ret; @@ -114,6 +114,7 @@ static int axg_card_add_tdm_loopback(struct snd_soc_card *card, if (ret) return ret; + pad = &card->dai_link[*index]; lb = &card->dai_link[*index + 1]; lb->name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-lb", pad->name); -- 2.30.1

3 2

[PATCH v3] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property
by Kuninori Morimoto 11 Sep '24

11 Sep '24

At least if rsnd is using DPCM connection on Audio-Graph-Card2, fw_devlink might doesn't have enough information to break the cycle (Same problem might occur with Multi-CPU/Codec or Codec2Codec). In such case, rsnd driver will not be probed. Add post-init-providers support to break the link cycle. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v2 -> v3 - remove "|" from description - removed common post-init-providers description - 100 char to 80 char for 1 line Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9def..3bc93c59535e9 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,12 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: At least if rsnd is using DPCM connection on Audio-Graph-Card2, + fw_devlink might doesn't have enough information to break the cycle. rsnd + driver will not be probed in such case. Same problem might occur with + Multi-CPU/Codec or Codec2Codec. + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base -- 2.43.0

3 2

[PATCH] ASoC: dt-bindings: microchip,sama7g5-spdifrx: Add common DAI reference
by Andrei Simion 11 Sep '24

11 Sep '24

Update the spdifrx yaml file to reference the dai-common.yaml schema, enabling the use of the 'sound-name-prefix' property Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> --- .../devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml index 2f43c684ab88..7fbab5871be4 100644 --- a/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml +++ b/Documentation/devicetree/bindings/sound/microchip,sama7g5-spdifrx.yaml @@ -13,6 +13,9 @@ description: The Microchip Sony/Philips Digital Interface Receiver is a serial port compliant with the IEC-60958 standard. +allOf: + - $ref: dai-common.yaml# + properties: "#sound-dai-cells": const: 0 @@ -53,7 +56,7 @@ required: - dmas - dma-names -additionalProperties: false +unevaluatedProperties: false examples: - | base-commit: 6708132e80a2ced620bde9b9c36e426183544a23 -- 2.34.1

3 2

[PATCH] ASoC: codecs: avoid possible garbage value in peb2466_reg_read()
by Su Hui 11 Sep '24

11 Sep '24

Clang static checker (scan-build) warning: sound/soc/codecs/peb2466.c:232:8: Assigned value is garbage or undefined [core.uninitialized.Assign] 232 | *val = tmp; | ^ ~~~ When peb2466_read_byte() fails, 'tmp' will have a garbage value. Add a judgemnet to avoid this problem. Fixes: 227f609c7c0e ("ASoC: codecs: Add support for the Infineon PEB2466 codec") Signed-off-by: Su Hui <suhui(a)nfschina.com> --- sound/soc/codecs/peb2466.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/peb2466.c b/sound/soc/codecs/peb2466.c index 76ee7e3f4d9b..67ea70cef0c7 100644 --- a/sound/soc/codecs/peb2466.c +++ b/sound/soc/codecs/peb2466.c @@ -229,7 +229,8 @@ static int peb2466_reg_read(void *context, unsigned int reg, unsigned int *val) case PEB2466_CMD_XOP: case PEB2466_CMD_SOP: ret = peb2466_read_byte(peb2466, reg, &tmp); - *val = tmp; + if (!ret) + *val = tmp; break; default: dev_err(&peb2466->spi->dev, "Not a XOP or SOP command\n"); -- 2.30.2

2 1

[PATCH 0/1] Add support for primary mi2s on SM8250
by Jens Reidel 11 Sep '24

11 Sep '24

This patch adds support for the primary mi2s interface on devices using SM8250 audio drivers. Tested on SM7150 (xiaomi-davinci). SM7150 sound is close to SM8250 and we intend to use it as a fallback in the future. To: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-sound(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: linux(a)mainlining.org Jens Reidel (1): ASoC: qcom: sm8250: enable primary mi2s sound/soc/qcom/sm8250.c | 8 ++++++++ 1 file changed, 8 insertions(+) -- 2.46.0

4 5

[PATCH 00/13] ALSA: update sample rate definitions
by Jerome Brunet 11 Sep '24

11 Sep '24

This patchset adds rate definitions for 12kHz, 24kHz and 128kHz. It is follow-up on the series/discussion [0] about adding 128kHz for spdif/eARC support. The outcome was to add 12kHz and 24kHz as well and clean up the drivers that no longer require custom rules to allow these rates. Identifying these drivers is not straight forward, I tried my best but I may have missed some. Those will continue to work anyway so it is not critical. Some drivers using these rates have not been changed on purpose. The reason for this may be: * The driver used other uncommon rates that still don't have a definition so a custom rule is still required. * The constraint structure is used in some other way by the driver and removing it would not help the readability or maintainability. This is the case the freescale asrc drivers for example. There is one change per driver so, if there is a problem later on, it will easier to properly add Fixes tag. The series has been tested with * ARM64 defconfig - spdif 128kHz at runtime. * x86_64 allmodconfig - compile test only Last, the change adding IEC958 definitions has been dropped for this patchset and will be resent separately [0]: https://lore.kernel.org/all/20240628122429.2018059-1-jbrunet@baylibre.com/ --- Jerome Brunet (13): ALSA: pcm: add more sample rate definitions ALSA: cmipci: drop SNDRV_PCM_RATE_KNOT ALSA: emu10k1: drop SNDRV_PCM_RATE_KNOT ALSA: hdsp: drop SNDRV_PCM_RATE_KNOT ALSA: hdspm: drop SNDRV_PCM_RATE_KNOT ASoC: cs35l36: drop SNDRV_PCM_RATE_KNOT ASoC: cs35l41: drop SNDRV_PCM_RATE_KNOT ASoC: cs53l30: drop SNDRV_PCM_RATE_KNOT ASoC: Intel: avs: drop SNDRV_PCM_RATE_KNOT ASoC: qcom: q6asm-dai: drop SNDRV_PCM_RATE_KNOT ASoC: sunxi: sun4i-codec: drop SNDRV_PCM_RATE_KNOT ASoC: cs35l34: drop useless rate contraint ASoC: spdif: extend supported rates to 768kHz include/sound/pcm.h | 31 +++++++++++++++++-------------- sound/core/pcm_native.c | 6 +++--- sound/pci/cmipci.c | 32 +++++++++----------------------- sound/pci/emu10k1/emupcm.c | 31 +++++-------------------------- sound/pci/rme9652/hdsp.c | 18 ++++++------------ sound/pci/rme9652/hdspm.c | 16 +--------------- sound/soc/codecs/cs35l34.c | 21 --------------------- sound/soc/codecs/cs35l36.c | 34 ++++++++++++---------------------- sound/soc/codecs/cs35l41.c | 34 +++++++++++----------------------- sound/soc/codecs/cs53l30.c | 24 +++--------------------- sound/soc/codecs/spdif_receiver.c | 3 ++- sound/soc/codecs/spdif_transmitter.c | 3 ++- sound/soc/intel/avs/pcm.c | 22 ++++++---------------- sound/soc/qcom/qdsp6/q6asm-dai.c | 31 ++++++++++--------------------- sound/soc/sunxi/sun4i-codec.c | 28 +++++++++------------------- 15 files changed, 96 insertions(+), 238 deletions(-) --- base-commit: 785f4052380684dbcc156203c537c799e2f4be09 change-id: 20240905-alsa-12-24-128-8edab4c08dd4 Best regards, -- Jerome

11 35

[PATCH 1/2] ASoC: amd: acp: remove MODULE_ALIAS for legacy machine driver
by Vijendar Mukunda 11 Sep '24

11 Sep '24

As module device table added for AMD legacy machine driver, MODULE_ALIAS is not required. Remove MODULE_ALIAS for AMD legacy machine driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp/acp-legacy-mach.c | 7 ------- 1 file changed, 7 deletions(-) diff --git a/sound/soc/amd/acp/acp-legacy-mach.c b/sound/soc/amd/acp/acp-legacy-mach.c index 0d529e32e552..d104f7e8fdcd 100644 --- a/sound/soc/amd/acp/acp-legacy-mach.c +++ b/sound/soc/amd/acp/acp-legacy-mach.c @@ -242,11 +242,4 @@ module_platform_driver(acp_asoc_audio); MODULE_IMPORT_NS(SND_SOC_AMD_MACH); MODULE_DESCRIPTION("ACP chrome audio support"); -MODULE_ALIAS("platform:acp3xalc56821019"); -MODULE_ALIAS("platform:acp3xalc5682sm98360"); -MODULE_ALIAS("platform:acp3xalc5682s1019"); -MODULE_ALIAS("platform:acp3x-es83xx"); -MODULE_ALIAS("platform:rmb-nau8825-max"); -MODULE_ALIAS("platform:rmb-rt5682s-rt1019"); -MODULE_ALIAS("platform:acp-pdm-mach"); MODULE_LICENSE("GPL v2"); -- 2.34.1

2 2

[PATCH v1 0/4] Add static channel mapping between soundwire master and slave
by Mohammad Rafi Shaik 11 Sep '24

11 Sep '24

Add static channel map support between soundwire master and slave. This patch series will resolve channel mask mismatch between master and slave. Scenario: wcd937x AMIC2 usecase Master Slave (wcd937x) +--------------+ +--------------+ | +--------+ | | +--------+ | AMIC1 ----->| | PORT1 | | | | TX1 | |<-----------AMIC1 AMIC2 ----->| | | | | | | | | +--------+ | | +--------+ | | | | | AMIC3 ----->| +--------+ | | +--------+ | | | PORT2 | | | | TX2 | |<-----------AMIC2 | | | | | | | |<-----------AMIC3 | +--------+ | | +--------+ | | | | | | +--------+ | | +--------+ | DMIC0...DMIC3------>| | PORT3 | | | | TX3 | |<-----------DMIC0...DMIC3 | | | | | | | |<-----------MBHC | +--------+ | | +--------+ | | | | | | +--------+ | | +--------+ | DMIC4...DMIC37----->| | PORT4 | | | | TX4 | |<-----------DMIC4...DMIC7 | | | | | | | | | +--------+ | | +--------+ | | | | | +------------- + +--------------+ For AMIC2 usecase, The Slave need to configure TX2 Port with channel mask 1 and for Master required PORT1 with channel mask 2, In existing design master and slave configured with same channel mask, it will fail AMIC2 usecase. The New design will help to configure channel mapping between master and slave from device tree. Mohammad Rafi Shaik (4): ASoC: dt-bindings: wcd938x-sdw: Add static channel mapping support soundwire: stream: Add set_master_channel_map() to set static channel mapping soundwire: qcom: Add static channel mapping support in soundwire master ASoC: codecs: wcd937x: Add static channel mapping support in wcd937x-sdw .../bindings/sound/qcom,wcd937x-sdw.yaml | 28 ++++++++++ drivers/soundwire/qcom.c | 18 +++++++ drivers/soundwire/stream.c | 16 ++++++ include/linux/soundwire/sdw.h | 5 ++ sound/soc/codecs/wcd937x-sdw.c | 52 ++++++++++++++++--- sound/soc/codecs/wcd937x.c | 12 ++++- sound/soc/codecs/wcd937x.h | 6 ++- 7 files changed, 126 insertions(+), 11 deletions(-) base-commit: 9aaeb87ce1e966169a57f53a02ba05b30880ffb8 -- 2.25.1

6 10

[PATCH RESEND] soundwire: stream: Revert "soundwire: stream: fix programming slave ports for non-continous port maps"
by Krzysztof Kozlowski 11 Sep '24

11 Sep '24

This reverts commit ab8d66d132bc8f1992d3eb6cab8d32dda6733c84 because it breaks codecs using non-continuous masks in source and sink ports. The commit missed the point that port numbers are not used as indices for iterating over prop.sink_ports or prop.source_ports. Soundwire core and existing codecs expect that the array passed as prop.sink_ports and prop.source_ports is continuous. The port mask still might be non-continuous, but that's unrelated. Reported-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Closes: https://lore.kernel.org/all/b6c75eee-761d-44c8-8413-2a5b34ee2f98@linux.inte… Fixes: ab8d66d132bc ("soundwire: stream: fix programming slave ports for non-continous port maps") Acked-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Reviewed-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Resending with Ack/Rb tags and missing Cc-stable. --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index f275143d7b18..7aa4900dcf31 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - unsigned long mask; + u8 num_ports; int i; if (direction == SDW_DATA_DIR_TX) { - mask = slave->prop.source_ports; + num_ports = hweight32(slave->prop.source_ports); dpn_prop = slave->prop.src_dpn_prop; } else { - mask = slave->prop.sink_ports; + num_ports = hweight32(slave->prop.sink_ports); dpn_prop = slave->prop.sink_dpn_prop; } - for_each_set_bit(i, &mask, 32) { + for (i = 0; i < num_ports; i++) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

3 4

[PATCH] ASoC: tlv320aic31xx: Fix typos
by Andrew Kreimer 11 Sep '24

11 Sep '24

Fix typos in comments. Reported-by: Matthew Wilcox <willy(a)infradead.org> Signed-off-by: Andrew Kreimer <algonell(a)gmail.com> --- sound/soc/codecs/tlv320aic31xx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 7e624c4b77b6..187d68e8688c 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -895,7 +895,7 @@ static int aic31xx_setup_pll(struct snd_soc_component *component, dev_err(component->dev, "%s: Sample rate (%u) and format not supported\n", __func__, params_rate(params)); - /* See bellow for details how fix this. */ + /* See below for details on how to fix this. */ return -EINVAL; } if (bclk_score != 0) { -- 2.46.0

2 3

[Patch v5 00/12] Add audio support for LPC32XX CPUs
by Piotr Wojtaszczyk 11 Sep '24

11 Sep '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 (12): 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: Use simple-mfd for clock control block ARM: dts: lpc32xx: Add missing dma properties ARM: dts: lpc32xx: Add missing 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: Fix potential deadlock warning from del_timer_sync() call in isr .../devicetree/bindings/dma/arm-pl08x.yaml | 7 + .../bindings/dma/nxp,lpc3220-dmamux.yaml | 49 +++ .../bindings/sound/nxp,lpc3220-i2s.yaml | 73 ++++ MAINTAINERS | 20 + 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 | 26 +- drivers/dma/Kconfig | 9 + drivers/dma/Makefile | 1 + drivers/dma/lpc32xx-dmamux.c | 195 +++++++++ drivers/i2c/busses/i2c-pnx.c | 48 +-- drivers/mtd/nand/raw/lpc32xx_mlc.c | 26 +- drivers/mtd/nand/raw/lpc32xx_slc.c | 26 +- sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/lpc3xxx-i2s.c | 375 ++++++++++++++++++ sound/soc/fsl/lpc3xxx-i2s.h | 79 ++++ sound/soc/fsl/lpc3xxx-pcm.c | 72 ++++ 19 files changed, 993 insertions(+), 130 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

7 19

[PATCH AUTOSEL 5.10 1/4] ASoC: intel: fix module autoloading
by Sasha Levin 10 Sep '24

10 Sep '24

From: Liao Chen <liaochen4(a)huawei.com> [ Upstream commit ae61a3391088d29aa8605c9f2db84295ab993a49 ] Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from of_device_id table. Signed-off-by: Liao Chen <liaochen4(a)huawei.com> Link: https://patch.msgid.link/20240826084924.368387-2-liaochen4@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/keembay/kmb_platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index 291a686568c2..c7b754034d24 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -634,6 +634,7 @@ static const struct of_device_id kmb_plat_of_match[] = { { .compatible = "intel,keembay-tdm", .data = &intel_kmb_tdm_dai}, {} }; +MODULE_DEVICE_TABLE(of, kmb_plat_of_match); static int kmb_plat_dai_probe(struct platform_device *pdev) { -- 2.43.0

1 0

[PATCH AUTOSEL 5.15 1/5] ASoC: intel: fix module autoloading
by Sasha Levin 10 Sep '24

10 Sep '24

From: Liao Chen <liaochen4(a)huawei.com> [ Upstream commit ae61a3391088d29aa8605c9f2db84295ab993a49 ] Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from of_device_id table. Signed-off-by: Liao Chen <liaochen4(a)huawei.com> Link: https://patch.msgid.link/20240826084924.368387-2-liaochen4@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/keembay/kmb_platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index a6fb74ba1c42..86a4c32686e7 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -815,6 +815,7 @@ static const struct of_device_id kmb_plat_of_match[] = { { .compatible = "intel,keembay-tdm", .data = &intel_kmb_tdm_dai}, {} }; +MODULE_DEVICE_TABLE(of, kmb_plat_of_match); static int kmb_plat_dai_probe(struct platform_device *pdev) { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 3/8] ASoC: intel: fix module autoloading
by Sasha Levin 10 Sep '24

10 Sep '24

From: Liao Chen <liaochen4(a)huawei.com> [ Upstream commit ae61a3391088d29aa8605c9f2db84295ab993a49 ] Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from of_device_id table. Signed-off-by: Liao Chen <liaochen4(a)huawei.com> Link: https://patch.msgid.link/20240826084924.368387-2-liaochen4@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/keembay/kmb_platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index b4893365d01d..d5c48bed7a25 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -817,6 +817,7 @@ static const struct of_device_id kmb_plat_of_match[] = { { .compatible = "intel,keembay-tdm", .data = &intel_kmb_tdm_dai}, {} }; +MODULE_DEVICE_TABLE(of, kmb_plat_of_match); static int kmb_plat_dai_probe(struct platform_device *pdev) { -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 2/8] ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict
by Sasha Levin 10 Sep '24

10 Sep '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 839a4ec06f75cec8fec2cc5fc14e921d0c3f7369 ] There are 2G and 4G RAM versions of the Lenovo Yoga Tab 3 X90F and it turns out that the 2G version has a DMI product name of "CHERRYVIEW D1 PLATFORM" where as the 4G version has "CHERRYVIEW C0 PLATFORM". The sys-vendor + product-version check are unique enough that the product-name check is not necessary. Drop the product-name check so that the existing DMI match for the 4G RAM version also matches the 2G RAM version. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240823074305.16873-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 5e2ec60e2954..e4c3492a0c28 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -84,7 +84,6 @@ static const struct dmi_system_id lenovo_yoga_tab3_x90[] = { /* Lenovo Yoga Tab 3 Pro YT3-X90, codec missing from DSDT */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), - DMI_MATCH(DMI_PRODUCT_NAME, "CHERRYVIEW D1 PLATFORM"), DMI_MATCH(DMI_PRODUCT_VERSION, "Blade3-10A-001"), }, }, -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 03/12] ASoC: intel: fix module autoloading
by Sasha Levin 10 Sep '24

10 Sep '24

From: Liao Chen <liaochen4(a)huawei.com> [ Upstream commit ae61a3391088d29aa8605c9f2db84295ab993a49 ] Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from of_device_id table. Signed-off-by: Liao Chen <liaochen4(a)huawei.com> Link: https://patch.msgid.link/20240826084924.368387-2-liaochen4@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/keembay/kmb_platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index 6b06b7b5ede8..ffe558ef4922 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -815,6 +815,7 @@ static const struct of_device_id kmb_plat_of_match[] = { { .compatible = "intel,keembay-tdm", .data = &intel_kmb_tdm_dai}, {} }; +MODULE_DEVICE_TABLE(of, kmb_plat_of_match); static int kmb_plat_dai_probe(struct platform_device *pdev) { -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 02/12] ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict
by Sasha Levin 10 Sep '24

10 Sep '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 839a4ec06f75cec8fec2cc5fc14e921d0c3f7369 ] There are 2G and 4G RAM versions of the Lenovo Yoga Tab 3 X90F and it turns out that the 2G version has a DMI product name of "CHERRYVIEW D1 PLATFORM" where as the 4G version has "CHERRYVIEW C0 PLATFORM". The sys-vendor + product-version check are unique enough that the product-name check is not necessary. Drop the product-name check so that the existing DMI match for the 4G RAM version also matches the 2G RAM version. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240823074305.16873-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 5e2ec60e2954..e4c3492a0c28 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -84,7 +84,6 @@ static const struct dmi_system_id lenovo_yoga_tab3_x90[] = { /* Lenovo Yoga Tab 3 Pro YT3-X90, codec missing from DSDT */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), - DMI_MATCH(DMI_PRODUCT_NAME, "CHERRYVIEW D1 PLATFORM"), DMI_MATCH(DMI_PRODUCT_VERSION, "Blade3-10A-001"), }, }, -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 04/18] ASoC: intel: fix module autoloading
by Sasha Levin 10 Sep '24

10 Sep '24

From: Liao Chen <liaochen4(a)huawei.com> [ Upstream commit ae61a3391088d29aa8605c9f2db84295ab993a49 ] Add MODULE_DEVICE_TABLE(), so modules could be properly autoloaded based on the alias from of_device_id table. Signed-off-by: Liao Chen <liaochen4(a)huawei.com> Link: https://patch.msgid.link/20240826084924.368387-2-liaochen4@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/keembay/kmb_platform.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c index 37ea2e1d2e92..aa5de167e790 100644 --- a/sound/soc/intel/keembay/kmb_platform.c +++ b/sound/soc/intel/keembay/kmb_platform.c @@ -814,6 +814,7 @@ static const struct of_device_id kmb_plat_of_match[] = { { .compatible = "intel,keembay-tdm", .data = &intel_kmb_tdm_dai}, {} }; +MODULE_DEVICE_TABLE(of, kmb_plat_of_match); static int kmb_plat_dai_probe(struct platform_device *pdev) { -- 2.43.0

1 0

[PATCH AUTOSEL 6.10 03/18] ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict
by Sasha Levin 10 Sep '24

10 Sep '24

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit 839a4ec06f75cec8fec2cc5fc14e921d0c3f7369 ] There are 2G and 4G RAM versions of the Lenovo Yoga Tab 3 X90F and it turns out that the 2G version has a DMI product name of "CHERRYVIEW D1 PLATFORM" where as the 4G version has "CHERRYVIEW C0 PLATFORM". The sys-vendor + product-version check are unique enough that the product-name check is not necessary. Drop the product-name check so that the existing DMI match for the 4G RAM version also matches the 2G RAM version. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://patch.msgid.link/20240823074305.16873-1-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 5e2ec60e2954..e4c3492a0c28 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -84,7 +84,6 @@ static const struct dmi_system_id lenovo_yoga_tab3_x90[] = { /* Lenovo Yoga Tab 3 Pro YT3-X90, codec missing from DSDT */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), - DMI_MATCH(DMI_PRODUCT_NAME, "CHERRYVIEW D1 PLATFORM"), DMI_MATCH(DMI_PRODUCT_VERSION, "Blade3-10A-001"), }, }, -- 2.43.0

1 0

[PATCH 0/2] Adjust Stream Name and DT Bindings Updates
by Andrei Simion 10 Sep '24

10 Sep '24

This patch set proposes the following changes to improve the flexibility and configurability of the mchp-i2s-mcc driver by allowing the interface name to be set through the device tree and by introducing a new property to better manage multiple interfaces. Codrin Ciubotariu (2): ASoC: atmel: mchp-i2s-mcc: Remove interface name from stream_name ASoC: dt-bindings: microchip,sama7g5-i2smcc: Add 'sound-name-prefix' property .../bindings/sound/microchip,sama7g5-i2smcc.yaml | 7 +++++++ sound/soc/atmel/mchp-i2s-mcc.c | 4 ++-- 2 files changed, 9 insertions(+), 2 deletions(-) base-commit: 9aaeb87ce1e966169a57f53a02ba05b30880ffb8 -- 2.34.1

4 8

[PATCH] ASoC: Switch back to struct platform_driver::remove()
by Uwe Kleine-König 10 Sep '24

10 Sep '24

After commit 0edb555a65d1 ("platform: Make platform_driver::remove() return void") .remove() is (again) the right callback to implement for platform drivers. Convert all drivers below sound/soc to use .remove(), with the eventual goal to drop struct platform_driver::remove_new(). As .remove() and .remove_new() have the same prototypes, conversion is done by just changing the structure member name in the driver initializer. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)baylibre.com> --- Hello, this is a single patch for all of sound/soc. If you prefer, I can split it. If so, just tell how (per subdir? per driver? somehow else?) Also I trimmed the list of recipents compared to what scripts/get_maintainer.sh emits. I dropped all the maintainers for the individual drivers and only kept the list. Otherwise the set of recipents is too huge. Best regards Uwe sound/soc/adi/axi-i2s.c | 2 +- sound/soc/adi/axi-spdif.c | 2 +- sound/soc/amd/acp-pcm-dma.c | 2 +- sound/soc/amd/acp/acp-rembrandt.c | 2 +- sound/soc/amd/acp/acp-renoir.c | 2 +- sound/soc/amd/acp/acp-sdw-sof-mach.c | 2 +- sound/soc/amd/acp/acp63.c | 2 +- sound/soc/amd/acp/acp70.c | 2 +- sound/soc/amd/ps/ps-pdm-dma.c | 2 +- sound/soc/amd/ps/ps-sdw-dma.c | 2 +- sound/soc/amd/raven/acp3x-pcm-dma.c | 2 +- sound/soc/amd/renoir/acp3x-pdm-dma.c | 2 +- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 2 +- sound/soc/amd/yc/acp6x-pdm-dma.c | 2 +- sound/soc/apple/mca.c | 2 +- sound/soc/atmel/atmel-i2s.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mchp-i2s-mcc.c | 2 +- sound/soc/atmel/mchp-pdmc.c | 2 +- sound/soc/atmel/mchp-spdifrx.c | 2 +- sound/soc/atmel/mchp-spdiftx.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/atmel/tse850-pcm5142.c | 2 +- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/i2sc.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/au1x/psc-i2s.c | 2 +- sound/soc/bcm/bcm63xx-i2s-whistler.c | 2 +- sound/soc/bcm/cygnus-ssp.c | 2 +- sound/soc/cirrus/edb93xx.c | 2 +- sound/soc/cirrus/ep93xx-i2s.c | 2 +- sound/soc/codecs/cs42l43.c | 2 +- sound/soc/codecs/cs47l15.c | 2 +- sound/soc/codecs/cs47l24.c | 2 +- sound/soc/codecs/cs47l35.c | 2 +- sound/soc/codecs/cs47l85.c | 2 +- sound/soc/codecs/cs47l90.c | 2 +- sound/soc/codecs/cs47l92.c | 2 +- sound/soc/codecs/inno_rk3036.c | 2 +- sound/soc/codecs/lpass-rx-macro.c | 2 +- sound/soc/codecs/lpass-tx-macro.c | 2 +- sound/soc/codecs/lpass-va-macro.c | 2 +- sound/soc/codecs/lpass-wsa-macro.c | 2 +- sound/soc/codecs/msm8916-wcd-digital.c | 2 +- sound/soc/codecs/rk817_codec.c | 2 +- sound/soc/codecs/wcd937x.c | 2 +- sound/soc/codecs/wcd938x.c | 2 +- sound/soc/codecs/wcd939x.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- sound/soc/codecs/wm8998.c | 2 +- sound/soc/dwc/dwc-i2s.c | 2 +- sound/soc/fsl/fsl_asrc.c | 2 +- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_audmix.c | 2 +- sound/soc/fsl/fsl_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_micfil.c | 2 +- sound/soc/fsl/fsl_mqs.c | 2 +- sound/soc/fsl/fsl_rpmsg.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 2 +- sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-pcm-rpmsg.c | 2 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_psc_ac97.c | 2 +- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/p1022_ds.c | 2 +- sound/soc/fsl/p1022_rdk.c | 2 +- sound/soc/fsl/pcm030-audio-fabric.c | 2 +- sound/soc/generic/audio-graph-card.c | 2 +- sound/soc/generic/audio-graph-card2-custom-sample.c | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/test-component.c | 2 +- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/img/img-parallel-out.c | 2 +- sound/soc/img/img-spdif-in.c | 2 +- sound/soc/img/img-spdif-out.c | 2 +- sound/soc/img/pistachio-internal-dac.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 2 +- sound/soc/intel/atom/sst/sst_acpi.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/sof_es8336.c | 2 +- sound/soc/intel/boards/sof_pcm512x.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 2 +- sound/soc/intel/boards/sof_wm8804.c | 2 +- sound/soc/intel/catpt/device.c | 2 +- sound/soc/kirkwood/kirkwood-i2s.c | 2 +- sound/soc/mediatek/common/mtk-btcvsd.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 2 +- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 2 +- sound/soc/mediatek/mt8365/mt8365-afe-pcm.c | 2 +- sound/soc/meson/aiu.c | 2 +- sound/soc/meson/axg-card.c | 2 +- sound/soc/meson/gx-card.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/pxa2xx-ac97.c | 2 +- sound/soc/qcom/lpass-apq8016.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass-sc7180.c | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/rockchip/rockchip_i2s.c | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 2 +- sound/soc/rockchip/rockchip_pdm.c | 2 +- sound/soc/rockchip/rockchip_rt5645.c | 2 +- sound/soc/rockchip/rockchip_spdif.c | 2 +- sound/soc/samsung/arndale.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 2 +- sound/soc/samsung/snow.c | 2 +- sound/soc/samsung/spdif.c | 2 +- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/hac.c | 2 +- sound/soc/sh/rcar/core.c | 2 +- sound/soc/sh/rz-ssi.c | 2 +- sound/soc/sh/siu_dai.c | 2 +- sound/soc/sof/imx/imx8.c | 2 +- sound/soc/sof/imx/imx8m.c | 2 +- sound/soc/sof/imx/imx8ulp.c | 2 +- sound/soc/sof/intel/bdw.c | 2 +- sound/soc/sof/intel/byt.c | 2 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 2 +- sound/soc/sprd/sprd-mcdt.c | 2 +- sound/soc/starfive/jh7110_pwmdac.c | 2 +- sound/soc/starfive/jh7110_tdm.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 2 +- sound/soc/stm/stm32_i2s.c | 2 +- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/stm/stm32_spdifrx.c | 2 +- sound/soc/sunxi/sun4i-codec.c | 2 +- sound/soc/sunxi/sun4i-i2s.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/sunxi/sun50i-dmic.c | 2 +- sound/soc/sunxi/sun8i-codec.c | 2 +- sound/soc/tegra/tegra186_asrc.c | 2 +- sound/soc/tegra/tegra186_dspk.c | 2 +- sound/soc/tegra/tegra20_ac97.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra210_admaif.c | 2 +- sound/soc/tegra/tegra210_adx.c | 2 +- sound/soc/tegra/tegra210_ahub.c | 2 +- sound/soc/tegra/tegra210_amx.c | 2 +- sound/soc/tegra/tegra210_dmic.c | 2 +- sound/soc/tegra/tegra210_i2s.c | 2 +- sound/soc/tegra/tegra210_mixer.c | 2 +- sound/soc/tegra/tegra210_mvc.c | 2 +- sound/soc/tegra/tegra210_ope.c | 2 +- sound/soc/tegra/tegra210_sfc.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 2 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/tegra/tegra_audio_graph_card.c | 2 +- sound/soc/ti/ams-delta.c | 2 +- sound/soc/ti/davinci-i2s.c | 2 +- sound/soc/ti/davinci-mcasp.c | 2 +- sound/soc/ti/omap-mcbsp.c | 2 +- sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-pxs2.c | 2 +- sound/soc/uniphier/evea.c | 2 +- sound/soc/ux500/mop500.c | 2 +- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/xilinx/xlnx_formatter_pcm.c | 2 +- sound/soc/xilinx/xlnx_spdif.c | 2 +- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 184 files changed, 184 insertions(+), 184 deletions(-) diff --git a/sound/soc/adi/axi-i2s.c b/sound/soc/adi/axi-i2s.c index 4107eddc9ca8..41f89384f8fd 100644 --- a/sound/soc/adi/axi-i2s.c +++ b/sound/soc/adi/axi-i2s.c @@ -293,7 +293,7 @@ static struct platform_driver axi_i2s_driver = { .of_match_table = axi_i2s_of_match, }, .probe = axi_i2s_probe, - .remove_new = axi_i2s_dev_remove, + .remove = axi_i2s_dev_remove, }; module_platform_driver(axi_i2s_driver); diff --git a/sound/soc/adi/axi-spdif.c b/sound/soc/adi/axi-spdif.c index 10545bd99704..5581134201a3 100644 --- a/sound/soc/adi/axi-spdif.c +++ b/sound/soc/adi/axi-spdif.c @@ -258,7 +258,7 @@ static struct platform_driver axi_spdif_driver = { .of_match_table = axi_spdif_of_match, }, .probe = axi_spdif_probe, - .remove_new = axi_spdif_dev_remove, + .remove = axi_spdif_dev_remove, }; module_platform_driver(axi_spdif_driver); diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index b857e2676fe8..897dde630022 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1426,7 +1426,7 @@ static const struct dev_pm_ops acp_pm_ops = { static struct platform_driver acp_dma_driver = { .probe = acp_audio_probe, - .remove_new = acp_audio_remove, + .remove = acp_audio_remove, .driver = { .name = DRV_NAME, .pm = &acp_pm_ops, diff --git a/sound/soc/amd/acp/acp-rembrandt.c b/sound/soc/amd/acp/acp-rembrandt.c index e19981c7d65a..396434a45eea 100644 --- a/sound/soc/amd/acp/acp-rembrandt.c +++ b/sound/soc/amd/acp/acp-rembrandt.c @@ -295,7 +295,7 @@ static const struct dev_pm_ops rmb_dma_pm_ops = { static struct platform_driver rembrandt_driver = { .probe = rembrandt_audio_probe, - .remove_new = rembrandt_audio_remove, + .remove = rembrandt_audio_remove, .driver = { .name = "acp_asoc_rembrandt", .pm = &rmb_dma_pm_ops, diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index db835ed7c208..5e3f730aa6bf 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -244,7 +244,7 @@ static const struct dev_pm_ops rn_dma_pm_ops = { static struct platform_driver renoir_driver = { .probe = renoir_audio_probe, - .remove_new = renoir_audio_remove, + .remove = renoir_audio_remove, .driver = { .name = "acp_asoc_renoir", .pm = &rn_dma_pm_ops, diff --git a/sound/soc/amd/acp/acp-sdw-sof-mach.c b/sound/soc/amd/acp/acp-sdw-sof-mach.c index 08f368b3bbc8..b1cd173f607d 100644 --- a/sound/soc/amd/acp/acp-sdw-sof-mach.c +++ b/sound/soc/amd/acp/acp-sdw-sof-mach.c @@ -729,7 +729,7 @@ static struct platform_driver sof_sdw_driver = { .pm = &snd_soc_pm_ops, }, .probe = mc_probe, - .remove_new = mc_remove, + .remove = mc_remove, .id_table = mc_id_table, }; diff --git a/sound/soc/amd/acp/acp63.c b/sound/soc/amd/acp/acp63.c index f340920b3289..f325c374f228 100644 --- a/sound/soc/amd/acp/acp63.c +++ b/sound/soc/amd/acp/acp63.c @@ -304,7 +304,7 @@ static const struct dev_pm_ops acp63_dma_pm_ops = { static struct platform_driver acp63_driver = { .probe = acp63_audio_probe, - .remove_new = acp63_audio_remove, + .remove = acp63_audio_remove, .driver = { .name = "acp_asoc_acp63", .pm = &acp63_dma_pm_ops, diff --git a/sound/soc/amd/acp/acp70.c b/sound/soc/amd/acp/acp70.c index 1b0b59a22924..68d2590e1a4e 100644 --- a/sound/soc/amd/acp/acp70.c +++ b/sound/soc/amd/acp/acp70.c @@ -278,7 +278,7 @@ static const struct dev_pm_ops acp70_dma_pm_ops = { static struct platform_driver acp70_driver = { .probe = acp_acp70_audio_probe, - .remove_new = acp_acp70_audio_remove, + .remove = acp_acp70_audio_remove, .driver = { .name = "acp_asoc_acp70", .pm = &acp70_dma_pm_ops, diff --git a/sound/soc/amd/ps/ps-pdm-dma.c b/sound/soc/amd/ps/ps-pdm-dma.c index 7bbacbab1095..318fc260f293 100644 --- a/sound/soc/amd/ps/ps-pdm-dma.c +++ b/sound/soc/amd/ps/ps-pdm-dma.c @@ -448,7 +448,7 @@ static const struct dev_pm_ops acp63_pdm_pm_ops = { static struct platform_driver acp63_pdm_dma_driver = { .probe = acp63_pdm_audio_probe, - .remove_new = acp63_pdm_audio_remove, + .remove = acp63_pdm_audio_remove, .driver = { .name = "acp_ps_pdm_dma", .pm = &acp63_pdm_pm_ops, diff --git a/sound/soc/amd/ps/ps-sdw-dma.c b/sound/soc/amd/ps/ps-sdw-dma.c index 2f630753278d..3b4b9c6b3171 100644 --- a/sound/soc/amd/ps/ps-sdw-dma.c +++ b/sound/soc/amd/ps/ps-sdw-dma.c @@ -551,7 +551,7 @@ static const struct dev_pm_ops acp63_pm_ops = { static struct platform_driver acp63_sdw_dma_driver = { .probe = acp63_sdw_platform_probe, - .remove_new = acp63_sdw_platform_remove, + .remove = acp63_sdw_platform_remove, .driver = { .name = "amd_ps_sdw_dma", .pm = &acp63_pm_ops, diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 3a50558f6751..bb9ed52d744d 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -509,7 +509,7 @@ static const struct dev_pm_ops acp3x_pm_ops = { static struct platform_driver acp3x_dma_driver = { .probe = acp3x_audio_probe, - .remove_new = acp3x_audio_remove, + .remove = acp3x_audio_remove, .driver = { .name = "acp3x_rv_i2s_dma", .pm = &acp3x_pm_ops, diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index c3b47e9bd239..95ac8c680037 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -489,7 +489,7 @@ static const struct dev_pm_ops acp_pdm_pm_ops = { static struct platform_driver acp_pdm_dma_driver = { .probe = acp_pdm_audio_probe, - .remove_new = acp_pdm_audio_remove, + .remove = acp_pdm_audio_remove, .driver = { .name = "acp_rn_pdm_dma", .pm = &acp_pdm_pm_ops, diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index 491b16e52a72..d5965f2b09bc 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -499,7 +499,7 @@ static const struct dev_pm_ops acp5x_pm_ops = { static struct platform_driver acp5x_dma_driver = { .probe = acp5x_audio_probe, - .remove_new = acp5x_audio_remove, + .remove = acp5x_audio_remove, .driver = { .name = "acp5x_i2s_dma", .pm = &acp5x_pm_ops, diff --git a/sound/soc/amd/yc/acp6x-pdm-dma.c b/sound/soc/amd/yc/acp6x-pdm-dma.c index 72c4591e451b..3eb3e82efb10 100644 --- a/sound/soc/amd/yc/acp6x-pdm-dma.c +++ b/sound/soc/amd/yc/acp6x-pdm-dma.c @@ -440,7 +440,7 @@ static const struct dev_pm_ops acp6x_pdm_pm_ops = { static struct platform_driver acp6x_pdm_dma_driver = { .probe = acp6x_pdm_audio_probe, - .remove_new = acp6x_pdm_audio_remove, + .remove = acp6x_pdm_audio_remove, .driver = { .name = "acp_yc_pdm_dma", .pm = &acp6x_pdm_pm_ops, diff --git a/sound/soc/apple/mca.c b/sound/soc/apple/mca.c index 3780aca71076..c9e7d40c47cc 100644 --- a/sound/soc/apple/mca.c +++ b/sound/soc/apple/mca.c @@ -1179,7 +1179,7 @@ static struct platform_driver apple_mca_driver = { .of_match_table = apple_mca_of_match, }, .probe = apple_mca_probe, - .remove_new = apple_mca_remove, + .remove = apple_mca_remove, }; module_platform_driver(apple_mca_driver); diff --git a/sound/soc/atmel/atmel-i2s.c b/sound/soc/atmel/atmel-i2s.c index 6c20c643f321..762199faf872 100644 --- a/sound/soc/atmel/atmel-i2s.c +++ b/sound/soc/atmel/atmel-i2s.c @@ -733,7 +733,7 @@ static struct platform_driver atmel_i2s_driver = { .of_match_table = atmel_i2s_dt_ids, }, .probe = atmel_i2s_probe, - .remove_new = atmel_i2s_remove, + .remove = atmel_i2s_remove, }; module_platform_driver(atmel_i2s_driver); diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index b7f16ea0cdfc..0f4021c6c588 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -187,7 +187,7 @@ static struct platform_driver atmel_asoc_wm8904_driver = { .pm = &snd_soc_pm_ops, }, .probe = atmel_asoc_wm8904_probe, - .remove_new = atmel_asoc_wm8904_remove, + .remove = atmel_asoc_wm8904_remove, }; module_platform_driver(atmel_asoc_wm8904_driver); diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 35a56b266b8d..fbcb0070c188 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -1129,7 +1129,7 @@ static struct platform_driver mchp_i2s_mcc_driver = { .of_match_table = mchp_i2s_mcc_dt_ids, }, .probe = mchp_i2s_mcc_probe, - .remove_new = mchp_i2s_mcc_remove, + .remove = mchp_i2s_mcc_remove, }; module_platform_driver(mchp_i2s_mcc_driver); diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index dcc4e14b3dde..260074018da9 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -1153,7 +1153,7 @@ static struct platform_driver mchp_pdmc_driver = { .pm = pm_ptr(&mchp_pdmc_pm_ops), }, .probe = mchp_pdmc_probe, - .remove_new = mchp_pdmc_remove, + .remove = mchp_pdmc_remove, }; module_platform_driver(mchp_pdmc_driver); diff --git a/sound/soc/atmel/mchp-spdifrx.c b/sound/soc/atmel/mchp-spdifrx.c index 33ce5e54482b..b2507a1491b7 100644 --- a/sound/soc/atmel/mchp-spdifrx.c +++ b/sound/soc/atmel/mchp-spdifrx.c @@ -1194,7 +1194,7 @@ static void mchp_spdifrx_remove(struct platform_device *pdev) static struct platform_driver mchp_spdifrx_driver = { .probe = mchp_spdifrx_probe, - .remove_new = mchp_spdifrx_remove, + .remove = mchp_spdifrx_remove, .driver = { .name = "mchp_spdifrx", .of_match_table = mchp_spdifrx_dt_ids, diff --git a/sound/soc/atmel/mchp-spdiftx.c b/sound/soc/atmel/mchp-spdiftx.c index a201a96fa690..4c60ea652896 100644 --- a/sound/soc/atmel/mchp-spdiftx.c +++ b/sound/soc/atmel/mchp-spdiftx.c @@ -888,7 +888,7 @@ static void mchp_spdiftx_remove(struct platform_device *pdev) static struct platform_driver mchp_spdiftx_driver = { .probe = mchp_spdiftx_probe, - .remove_new = mchp_spdiftx_remove, + .remove = mchp_spdiftx_remove, .driver = { .name = "mchp_spdiftx", .of_match_table = mchp_spdiftx_dt_ids, diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index d3ec9826d505..335e216ea7b4 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -207,7 +207,7 @@ static struct platform_driver at91sam9g20ek_audio_driver = { .of_match_table = of_match_ptr(at91sam9g20ek_wm8731_dt_ids), }, .probe = at91sam9g20ek_audio_probe, - .remove_new = at91sam9g20ek_audio_remove, + .remove = at91sam9g20ek_audio_remove, }; module_platform_driver(at91sam9g20ek_audio_driver); diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index d1c1f370a9cd..1b5ef4e9d2b8 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -196,7 +196,7 @@ static struct platform_driver sam9x5_wm8731_driver = { .of_match_table = of_match_ptr(sam9x5_wm8731_of_match), }, .probe = sam9x5_wm8731_driver_probe, - .remove_new = sam9x5_wm8731_driver_remove, + .remove = sam9x5_wm8731_driver_remove, }; module_platform_driver(sam9x5_wm8731_driver); diff --git a/sound/soc/atmel/tse850-pcm5142.c b/sound/soc/atmel/tse850-pcm5142.c index 5d208e0b4b90..0a9efd5f2861 100644 --- a/sound/soc/atmel/tse850-pcm5142.c +++ b/sound/soc/atmel/tse850-pcm5142.c @@ -431,7 +431,7 @@ static struct platform_driver tse850_driver = { .of_match_table = tse850_dt_ids, }, .probe = tse850_probe, - .remove_new = tse850_remove, + .remove = tse850_remove, }; module_platform_driver(tse850_driver); diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index b0e1a1253e10..f8ab936250dc 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -336,7 +336,7 @@ static struct platform_driver au1xac97c_driver = { .pm = AU1XPSCAC97_PMOPS, }, .probe = au1xac97c_drvprobe, - .remove_new = au1xac97c_drvremove, + .remove = au1xac97c_drvremove, }; module_platform_driver(au1xac97c_driver); diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 064406080d72..7d296f29dade 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -313,7 +313,7 @@ static struct platform_driver au1xi2s_driver = { .pm = AU1XI2SC_PMOPS, }, .probe = au1xi2s_drvprobe, - .remove_new = au1xi2s_drvremove, + .remove = au1xi2s_drvremove, }; module_platform_driver(au1xi2s_driver); diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 1727eeb12b64..8a59a50978b9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -486,7 +486,7 @@ static struct platform_driver au1xpsc_ac97_driver = { .pm = AU1XPSCAC97_PMOPS, }, .probe = au1xpsc_ac97_drvprobe, - .remove_new = au1xpsc_ac97_drvremove, + .remove = au1xpsc_ac97_drvremove, }; module_platform_driver(au1xpsc_ac97_driver); diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 52734dec8247..bee013555e7a 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -404,7 +404,7 @@ static struct platform_driver au1xpsc_i2s_driver = { .pm = AU1XPSCI2S_PMOPS, }, .probe = au1xpsc_i2s_drvprobe, - .remove_new = au1xpsc_i2s_drvremove, + .remove = au1xpsc_i2s_drvremove, }; module_platform_driver(au1xpsc_i2s_driver); diff --git a/sound/soc/bcm/bcm63xx-i2s-whistler.c b/sound/soc/bcm/bcm63xx-i2s-whistler.c index c64609718738..c47ed1e6ea2b 100644 --- a/sound/soc/bcm/bcm63xx-i2s-whistler.c +++ b/sound/soc/bcm/bcm63xx-i2s-whistler.c @@ -293,7 +293,7 @@ static struct platform_driver bcm63xx_i2s_driver = { .of_match_table = of_match_ptr(snd_soc_bcm_audio_match), }, .probe = bcm63xx_i2s_dev_probe, - .remove_new = bcm63xx_i2s_dev_remove, + .remove = bcm63xx_i2s_dev_remove, }; module_platform_driver(bcm63xx_i2s_driver); diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index 90088516fed0..e0ce0232eb1e 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1390,7 +1390,7 @@ MODULE_DEVICE_TABLE(of, cygnus_ssp_of_match); static struct platform_driver cygnus_ssp_driver = { .probe = cygnus_ssp_probe, - .remove_new = cygnus_ssp_remove, + .remove = cygnus_ssp_remove, .driver = { .name = "cygnus-ssp", .of_match_table = cygnus_ssp_of_match, diff --git a/sound/soc/cirrus/edb93xx.c b/sound/soc/cirrus/edb93xx.c index 8bb67d7d2b4b..8dac754ddb0d 100644 --- a/sound/soc/cirrus/edb93xx.c +++ b/sound/soc/cirrus/edb93xx.c @@ -105,7 +105,7 @@ static struct platform_driver edb93xx_driver = { .name = "edb93xx-audio", }, .probe = edb93xx_probe, - .remove_new = edb93xx_remove, + .remove = edb93xx_remove, }; module_platform_driver(edb93xx_driver); diff --git a/sound/soc/cirrus/ep93xx-i2s.c b/sound/soc/cirrus/ep93xx-i2s.c index 522de4b80293..d45862ceb0c9 100644 --- a/sound/soc/cirrus/ep93xx-i2s.c +++ b/sound/soc/cirrus/ep93xx-i2s.c @@ -523,7 +523,7 @@ MODULE_DEVICE_TABLE(of, ep93xx_i2s_of_ids); static struct platform_driver ep93xx_i2s_driver = { .probe = ep93xx_i2s_probe, - .remove_new = ep93xx_i2s_remove, + .remove = ep93xx_i2s_remove, .driver = { .name = "ep93xx-i2s", .of_match_table = ep93xx_i2s_of_ids, diff --git a/sound/soc/codecs/cs42l43.c b/sound/soc/codecs/cs42l43.c index 5183b4586424..d0098b4558b5 100644 --- a/sound/soc/codecs/cs42l43.c +++ b/sound/soc/codecs/cs42l43.c @@ -2461,7 +2461,7 @@ static struct platform_driver cs42l43_codec_driver = { }, .probe = cs42l43_codec_probe, - .remove_new = cs42l43_codec_remove, + .remove = cs42l43_codec_remove, .id_table = cs42l43_codec_id_table, }; module_platform_driver(cs42l43_codec_driver); diff --git a/sound/soc/codecs/cs47l15.c b/sound/soc/codecs/cs47l15.c index ab6e7cd99733..29a2bcfb3048 100644 --- a/sound/soc/codecs/cs47l15.c +++ b/sound/soc/codecs/cs47l15.c @@ -1493,7 +1493,7 @@ static struct platform_driver cs47l15_codec_driver = { .name = "cs47l15-codec", }, .probe = &cs47l15_probe, - .remove_new = cs47l15_remove, + .remove = cs47l15_remove, }; module_platform_driver(cs47l15_codec_driver); diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index ec405ef66a8e..e2a839fae4fc 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1344,7 +1344,7 @@ static struct platform_driver cs47l24_codec_driver = { .name = "cs47l24-codec", }, .probe = cs47l24_probe, - .remove_new = cs47l24_remove, + .remove = cs47l24_remove, }; module_platform_driver(cs47l24_codec_driver); diff --git a/sound/soc/codecs/cs47l35.c b/sound/soc/codecs/cs47l35.c index 0d7ee7ea6257..85555c7a2e4b 100644 --- a/sound/soc/codecs/cs47l35.c +++ b/sound/soc/codecs/cs47l35.c @@ -1769,7 +1769,7 @@ static struct platform_driver cs47l35_codec_driver = { .name = "cs47l35-codec", }, .probe = &cs47l35_probe, - .remove_new = cs47l35_remove, + .remove = cs47l35_remove, }; module_platform_driver(cs47l35_codec_driver); diff --git a/sound/soc/codecs/cs47l85.c b/sound/soc/codecs/cs47l85.c index 2dfb867e6edd..d34f4e8c26d3 100644 --- a/sound/soc/codecs/cs47l85.c +++ b/sound/soc/codecs/cs47l85.c @@ -2720,7 +2720,7 @@ static struct platform_driver cs47l85_codec_driver = { .name = "cs47l85-codec", }, .probe = &cs47l85_probe, - .remove_new = cs47l85_remove, + .remove = cs47l85_remove, }; module_platform_driver(cs47l85_codec_driver); diff --git a/sound/soc/codecs/cs47l90.c b/sound/soc/codecs/cs47l90.c index 2549cb1fc121..a9e703981f37 100644 --- a/sound/soc/codecs/cs47l90.c +++ b/sound/soc/codecs/cs47l90.c @@ -2644,7 +2644,7 @@ static struct platform_driver cs47l90_codec_driver = { .name = "cs47l90-codec", }, .probe = &cs47l90_probe, - .remove_new = cs47l90_remove, + .remove = cs47l90_remove, }; module_platform_driver(cs47l90_codec_driver); diff --git a/sound/soc/codecs/cs47l92.c b/sound/soc/codecs/cs47l92.c index 0c05ae0b09fb..2c355c61acd8 100644 --- a/sound/soc/codecs/cs47l92.c +++ b/sound/soc/codecs/cs47l92.c @@ -2092,7 +2092,7 @@ static struct platform_driver cs47l92_codec_driver = { .name = "cs47l92-codec", }, .probe = &cs47l92_probe, - .remove_new = cs47l92_remove, + .remove = cs47l92_remove, }; module_platform_driver(cs47l92_codec_driver); diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c index 11320423c69c..fdd19f8e8864 100644 --- a/sound/soc/codecs/inno_rk3036.c +++ b/sound/soc/codecs/inno_rk3036.c @@ -476,7 +476,7 @@ static struct platform_driver rk3036_codec_platform_driver = { .of_match_table = of_match_ptr(rk3036_codec_of_match), }, .probe = rk3036_codec_platform_probe, - .remove_new = rk3036_codec_platform_remove, + .remove = rk3036_codec_platform_remove, }; module_platform_driver(rk3036_codec_platform_driver); diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index ce42749660c8..71e0d3bffd3f 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -4024,7 +4024,7 @@ static struct platform_driver rx_macro_driver = { .pm = &rx_macro_pm_ops, }, .probe = rx_macro_probe, - .remove_new = rx_macro_remove, + .remove = rx_macro_remove, }; module_platform_driver(rx_macro_driver); diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 209c12ec16dd..a134584acf90 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -2534,7 +2534,7 @@ static struct platform_driver tx_macro_driver = { .pm = &tx_macro_pm_ops, }, .probe = tx_macro_probe, - .remove_new = tx_macro_remove, + .remove = tx_macro_remove, }; module_platform_driver(tx_macro_driver); diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index e95d1f29ef18..c781da476240 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1738,7 +1738,7 @@ static struct platform_driver va_macro_driver = { .pm = &va_macro_pm_ops, }, .probe = va_macro_probe, - .remove_new = va_macro_remove, + .remove = va_macro_remove, }; module_platform_driver(va_macro_driver); diff --git a/sound/soc/codecs/lpass-wsa-macro.c b/sound/soc/codecs/lpass-wsa-macro.c index 76900274acf3..c989d82d1d3c 100644 --- a/sound/soc/codecs/lpass-wsa-macro.c +++ b/sound/soc/codecs/lpass-wsa-macro.c @@ -2980,7 +2980,7 @@ static struct platform_driver wsa_macro_driver = { .pm = &wsa_macro_pm_ops, }, .probe = wsa_macro_probe, - .remove_new = wsa_macro_remove, + .remove = wsa_macro_remove, }; module_platform_driver(wsa_macro_driver); diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 978c4d056e81..ebb6f2e84818 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -1241,7 +1241,7 @@ static struct platform_driver msm8916_wcd_digital_driver = { .of_match_table = msm8916_wcd_digital_match_table, }, .probe = msm8916_wcd_digital_probe, - .remove_new = msm8916_wcd_digital_remove, + .remove = msm8916_wcd_digital_remove, }; module_platform_driver(msm8916_wcd_digital_driver); diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 5fea600bc3a4..3c5b66357661 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -529,7 +529,7 @@ static struct platform_driver rk817_codec_driver = { .name = "rk817-codec", }, .probe = rk817_platform_probe, - .remove_new = rk817_platform_remove, + .remove = rk817_platform_remove, }; module_platform_driver(rk817_codec_driver); diff --git a/sound/soc/codecs/wcd937x.c b/sound/soc/codecs/wcd937x.c index af296b77a723..45f32d281908 100644 --- a/sound/soc/codecs/wcd937x.c +++ b/sound/soc/codecs/wcd937x.c @@ -2957,7 +2957,7 @@ MODULE_DEVICE_TABLE(of, wcd937x_of_match); static struct platform_driver wcd937x_codec_driver = { .probe = wcd937x_probe, - .remove_new = wcd937x_remove, + .remove = wcd937x_remove, .driver = { .name = "wcd937x_codec", .of_match_table = of_match_ptr(wcd937x_of_match), diff --git a/sound/soc/codecs/wcd938x.c b/sound/soc/codecs/wcd938x.c index aa92f1bfc921..f2a4f3262bdb 100644 --- a/sound/soc/codecs/wcd938x.c +++ b/sound/soc/codecs/wcd938x.c @@ -3596,7 +3596,7 @@ MODULE_DEVICE_TABLE(of, wcd938x_dt_match); static struct platform_driver wcd938x_codec_driver = { .probe = wcd938x_probe, - .remove_new = wcd938x_remove, + .remove = wcd938x_remove, .driver = { .name = "wcd938x_codec", .of_match_table = of_match_ptr(wcd938x_dt_match), diff --git a/sound/soc/codecs/wcd939x.c b/sound/soc/codecs/wcd939x.c index 68fc591670dc..4a417a92514d 100644 --- a/sound/soc/codecs/wcd939x.c +++ b/sound/soc/codecs/wcd939x.c @@ -3683,7 +3683,7 @@ MODULE_DEVICE_TABLE(of, wcd939x_dt_match); static struct platform_driver wcd939x_codec_driver = { .probe = wcd939x_probe, - .remove_new = wcd939x_remove, + .remove = wcd939x_remove, .driver = { .name = "wcd939x_codec", .of_match_table = of_match_ptr(wcd939x_dt_match), diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 4ecf07c7448c..651f1319204d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2174,7 +2174,7 @@ static struct platform_driver wm5102_codec_driver = { .name = "wm5102-codec", }, .probe = wm5102_probe, - .remove_new = wm5102_remove, + .remove = wm5102_remove, }; module_platform_driver(wm5102_codec_driver); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 0f299cd07b2e..502196253d42 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2534,7 +2534,7 @@ static struct platform_driver wm5110_codec_driver = { .name = "wm5110-codec", }, .probe = wm5110_probe, - .remove_new = wm5110_remove, + .remove = wm5110_remove, }; module_platform_driver(wm5110_codec_driver); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a99908582a50..a4abe6e53bfc 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4699,7 +4699,7 @@ static struct platform_driver wm8994_codec_driver = { .pm = &wm8994_pm_ops, }, .probe = wm8994_probe, - .remove_new = wm8994_remove, + .remove = wm8994_remove, }; module_platform_driver(wm8994_codec_driver); diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 87442840f0af..5389c363b14e 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1210,7 +1210,7 @@ static struct platform_driver wm8997_codec_driver = { .name = "wm8997-codec", }, .probe = wm8997_probe, - .remove_new = wm8997_remove, + .remove = wm8997_remove, }; module_platform_driver(wm8997_codec_driver); diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 3c2c4d12c08e..b72b8a64be8f 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1426,7 +1426,7 @@ static struct platform_driver wm8998_codec_driver = { .name = "wm8998-codec", }, .probe = wm8998_probe, - .remove_new = wm8998_remove, + .remove = wm8998_remove, }; module_platform_driver(wm8998_codec_driver); diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index e6a5eebe5bc9..57b789d7fbed 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -1089,7 +1089,7 @@ static const struct dev_pm_ops dwc_pm_ops = { static struct platform_driver dw_i2s_driver = { .probe = dw_i2s_probe, - .remove_new = dw_i2s_remove, + .remove = dw_i2s_remove, .driver = { .name = "designware-i2s", .of_match_table = of_match_ptr(dw_i2s_of_match), diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index b793263291dc..bd5c46d763c0 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -1392,7 +1392,7 @@ MODULE_DEVICE_TABLE(of, fsl_asrc_ids); static struct platform_driver fsl_asrc_driver = { .probe = fsl_asrc_probe, - .remove_new = fsl_asrc_remove, + .remove = fsl_asrc_remove, .driver = { .name = "fsl-asrc", .of_match_table = fsl_asrc_ids, diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index a6cbaa6364c7..021d73e409aa 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -296,7 +296,7 @@ static const struct dev_pm_ops fsl_aud2htx_pm_ops = { static struct platform_driver fsl_aud2htx_driver = { .probe = fsl_aud2htx_probe, - .remove_new = fsl_aud2htx_remove, + .remove = fsl_aud2htx_remove, .driver = { .name = "fsl-aud2htx", .pm = pm_ptr(&fsl_aud2htx_pm_ops), diff --git a/sound/soc/fsl/fsl_audmix.c b/sound/soc/fsl/fsl_audmix.c index f3a24758aedb..3cd9a66b70a1 100644 --- a/sound/soc/fsl/fsl_audmix.c +++ b/sound/soc/fsl/fsl_audmix.c @@ -548,7 +548,7 @@ static const struct dev_pm_ops fsl_audmix_pm = { static struct platform_driver fsl_audmix_driver = { .probe = fsl_audmix_probe, - .remove_new = fsl_audmix_remove, + .remove = fsl_audmix_remove, .driver = { .name = "fsl-audmix", .of_match_table = fsl_audmix_ids, diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index c4bc9395dff7..aca066b5a43c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -911,7 +911,7 @@ static struct platform_driver fsl_soc_dma_driver = { .of_match_table = fsl_soc_dma_ids, }, .probe = fsl_soc_dma_probe, - .remove_new = fsl_soc_dma_remove, + .remove = fsl_soc_dma_remove, }; module_platform_driver(fsl_soc_dma_driver); diff --git a/sound/soc/fsl/fsl_easrc.c b/sound/soc/fsl/fsl_easrc.c index 962f30912091..82359edd6a8b 100644 --- a/sound/soc/fsl/fsl_easrc.c +++ b/sound/soc/fsl/fsl_easrc.c @@ -2093,7 +2093,7 @@ static const struct dev_pm_ops fsl_easrc_pm_ops = { static struct platform_driver fsl_easrc_driver = { .probe = fsl_easrc_probe, - .remove_new = fsl_easrc_remove, + .remove = fsl_easrc_remove, .driver = { .name = "fsl-easrc", .pm = pm_ptr(&fsl_easrc_pm_ops), diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index d0d8a01da9bd..a65f5b9935a2 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1198,7 +1198,7 @@ static const struct dev_pm_ops fsl_esai_pm_ops = { static struct platform_driver fsl_esai_driver = { .probe = fsl_esai_probe, - .remove_new = fsl_esai_remove, + .remove = fsl_esai_remove, .driver = { .name = "fsl-esai-dai", .pm = &fsl_esai_pm_ops, diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c index 22b240a70ad4..193be098fa5e 100644 --- a/sound/soc/fsl/fsl_micfil.c +++ b/sound/soc/fsl/fsl_micfil.c @@ -1317,7 +1317,7 @@ static const struct dev_pm_ops fsl_micfil_pm_ops = { static struct platform_driver fsl_micfil_driver = { .probe = fsl_micfil_probe, - .remove_new = fsl_micfil_remove, + .remove = fsl_micfil_remove, .driver = { .name = "fsl-micfil-dai", .pm = &fsl_micfil_pm_ops, diff --git a/sound/soc/fsl/fsl_mqs.c b/sound/soc/fsl/fsl_mqs.c index df160834c81b..145f9ca15e43 100644 --- a/sound/soc/fsl/fsl_mqs.c +++ b/sound/soc/fsl/fsl_mqs.c @@ -381,7 +381,7 @@ MODULE_DEVICE_TABLE(of, fsl_mqs_dt_ids); static struct platform_driver fsl_mqs_driver = { .probe = fsl_mqs_probe, - .remove_new = fsl_mqs_remove, + .remove = fsl_mqs_remove, .driver = { .name = "fsl-mqs", .of_match_table = fsl_mqs_dt_ids, diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index be46fbfd487a..0a551be3053b 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -328,7 +328,7 @@ static const struct dev_pm_ops fsl_rpmsg_pm_ops = { static struct platform_driver fsl_rpmsg_driver = { .probe = fsl_rpmsg_probe, - .remove_new = fsl_rpmsg_remove, + .remove = fsl_rpmsg_remove, .driver = { .name = "fsl_rpmsg", .pm = pm_ptr(&fsl_rpmsg_pm_ops), diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d03b0172b8ad..ab58a4461073 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1817,7 +1817,7 @@ static const struct dev_pm_ops fsl_sai_pm_ops = { static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, - .remove_new = fsl_sai_remove, + .remove = fsl_sai_remove, .driver = { .name = "fsl-sai", .pm = &fsl_sai_pm_ops, diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index eace399cb064..b6ff04f7138a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1763,7 +1763,7 @@ static struct platform_driver fsl_spdif_driver = { .pm = pm_ptr(&fsl_spdif_pm), }, .probe = fsl_spdif_probe, - .remove_new = fsl_spdif_remove, + .remove = fsl_spdif_remove, }; module_platform_driver(fsl_spdif_driver); diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c4c1d9c44056..320108bebf30 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1734,7 +1734,7 @@ static struct platform_driver fsl_ssi_driver = { .pm = pm_sleep_ptr(&fsl_ssi_pm), }, .probe = fsl_ssi_probe, - .remove_new = fsl_ssi_remove, + .remove = fsl_ssi_remove, }; module_platform_driver(fsl_ssi_driver); diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c index c98e3db74fe7..beede7344efd 100644 --- a/sound/soc/fsl/fsl_xcvr.c +++ b/sound/soc/fsl/fsl_xcvr.c @@ -1540,7 +1540,7 @@ static struct platform_driver fsl_xcvr_driver = { .pm = pm_ptr(&fsl_xcvr_pm_ops), .of_match_table = fsl_xcvr_dt_ids, }, - .remove_new = fsl_xcvr_remove, + .remove = fsl_xcvr_remove, }; module_platform_driver(fsl_xcvr_driver); diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index f97ae14dc452..43e14f2eca8d 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -354,7 +354,7 @@ static const struct dev_pm_ops imx_audmux_pm = { static struct platform_driver imx_audmux_driver = { .probe = imx_audmux_probe, - .remove_new = imx_audmux_remove, + .remove = imx_audmux_remove, .driver = { .name = DRIVER_NAME, .pm = pm_sleep_ptr(&imx_audmux_pm), diff --git a/sound/soc/fsl/imx-pcm-rpmsg.c b/sound/soc/fsl/imx-pcm-rpmsg.c index 22156f99dcee..1daf0be3d100 100644 --- a/sound/soc/fsl/imx-pcm-rpmsg.c +++ b/sound/soc/fsl/imx-pcm-rpmsg.c @@ -822,7 +822,7 @@ MODULE_DEVICE_TABLE(platform, imx_rpmsg_pcm_id_table); static struct platform_driver imx_pcm_rpmsg_driver = { .probe = imx_rpmsg_pcm_probe, - .remove_new = imx_rpmsg_pcm_remove, + .remove = imx_rpmsg_pcm_remove, .id_table = imx_rpmsg_pcm_id_table, .driver = { .name = IMX_PCM_DRV_NAME, diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 3c1e69092d2f..8bcf54ef709e 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -214,7 +214,7 @@ static struct platform_driver imx_sgtl5000_driver = { .of_match_table = imx_sgtl5000_dt_ids, }, .probe = imx_sgtl5000_probe, - .remove_new = imx_sgtl5000_remove, + .remove = imx_sgtl5000_remove, }; module_platform_driver(imx_sgtl5000_driver); diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 0423cf43c7a0..8554fb690772 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -327,7 +327,7 @@ MODULE_DEVICE_TABLE(of, psc_ac97_match); static struct platform_driver psc_ac97_driver = { .probe = psc_ac97_of_probe, - .remove_new = psc_ac97_of_remove, + .remove = psc_ac97_of_remove, .driver = { .name = "mpc5200-psc-ac97", .of_match_table = psc_ac97_match, diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c index 931b430fa983..9ad44eeed6ad 100644 --- a/sound/soc/fsl/mpc5200_psc_i2s.c +++ b/sound/soc/fsl/mpc5200_psc_i2s.c @@ -225,7 +225,7 @@ MODULE_DEVICE_TABLE(of, psc_i2s_match); static struct platform_driver psc_i2s_driver = { .probe = psc_i2s_of_probe, - .remove_new = psc_i2s_of_remove, + .remove = psc_i2s_of_remove, .driver = { .name = "mpc5200-psc-i2s", .of_match_table = psc_i2s_match, diff --git a/sound/soc/fsl/p1022_ds.c b/sound/soc/fsl/p1022_ds.c index 6f5eecf6d88c..66db05970d82 100644 --- a/sound/soc/fsl/p1022_ds.c +++ b/sound/soc/fsl/p1022_ds.c @@ -408,7 +408,7 @@ static void p1022_ds_remove(struct platform_device *pdev) static struct platform_driver p1022_ds_driver = { .probe = p1022_ds_probe, - .remove_new = p1022_ds_remove, + .remove = p1022_ds_remove, .driver = { /* * The name must match 'compatible' property in the device tree, diff --git a/sound/soc/fsl/p1022_rdk.c b/sound/soc/fsl/p1022_rdk.c index a42149311c8b..d4568346714f 100644 --- a/sound/soc/fsl/p1022_rdk.c +++ b/sound/soc/fsl/p1022_rdk.c @@ -370,7 +370,7 @@ static void p1022_rdk_remove(struct platform_device *pdev) static struct platform_driver p1022_rdk_driver = { .probe = p1022_rdk_probe, - .remove_new = p1022_rdk_remove, + .remove = p1022_rdk_remove, .driver = { /* * The name must match 'compatible' property in the device tree, diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 2bab0fc1de59..5542c4ee6d12 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -124,7 +124,7 @@ MODULE_DEVICE_TABLE(of, pcm030_audio_match); static struct platform_driver pcm030_fabric_driver = { .probe = pcm030_fabric_probe, - .remove_new = pcm030_fabric_remove, + .remove = pcm030_fabric_remove, .driver = { .name = DRV_NAME, .of_match_table = pcm030_audio_match, diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 03add80e75f3..4b384475b972 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -664,7 +664,7 @@ static struct platform_driver graph_card = { .of_match_table = graph_of_match, }, .probe = graph_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(graph_card); diff --git a/sound/soc/generic/audio-graph-card2-custom-sample.c b/sound/soc/generic/audio-graph-card2-custom-sample.c index 8e5a51098490..7151d426bee9 100644 --- a/sound/soc/generic/audio-graph-card2-custom-sample.c +++ b/sound/soc/generic/audio-graph-card2-custom-sample.c @@ -177,7 +177,7 @@ static struct platform_driver custom_card = { .of_match_table = custom_of_match, }, .probe = custom_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(custom_card); diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index bf4c1ab4ec58..4ad3d1b0714f 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -1438,7 +1438,7 @@ static struct platform_driver graph_card = { .of_match_table = graph_of_match, }, .probe = graph_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(graph_card); diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 42c60b92cca5..76a1d05e2ebe 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -838,7 +838,7 @@ static struct platform_driver simple_card = { .of_match_table = simple_of_match, }, .probe = simple_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(simple_card); diff --git a/sound/soc/generic/test-component.c b/sound/soc/generic/test-component.c index df2487b700cc..407288055741 100644 --- a/sound/soc/generic/test-component.c +++ b/sound/soc/generic/test-component.c @@ -637,7 +637,7 @@ static struct platform_driver test_driver = { .of_match_table = test_of_match, }, .probe = test_driver_probe, - .remove_new = test_driver_remove, + .remove = test_driver_remove, }; module_platform_driver(test_driver); diff --git a/sound/soc/img/img-i2s-in.c b/sound/soc/img/img-i2s-in.c index b69a364d619e..6a988976fb0d 100644 --- a/sound/soc/img/img-i2s-in.c +++ b/sound/soc/img/img-i2s-in.c @@ -607,7 +607,7 @@ static struct platform_driver img_i2s_in_driver = { .pm = &img_i2s_in_pm_ops }, .probe = img_i2s_in_probe, - .remove_new = img_i2s_in_dev_remove + .remove = img_i2s_in_dev_remove }; module_platform_driver(img_i2s_in_driver); diff --git a/sound/soc/img/img-i2s-out.c b/sound/soc/img/img-i2s-out.c index 6f9831c6d6e0..1211e6184d97 100644 --- a/sound/soc/img/img-i2s-out.c +++ b/sound/soc/img/img-i2s-out.c @@ -607,7 +607,7 @@ static struct platform_driver img_i2s_out_driver = { .pm = &img_i2s_out_pm_ops }, .probe = img_i2s_out_probe, - .remove_new = img_i2s_out_dev_remove + .remove = img_i2s_out_dev_remove }; module_platform_driver(img_i2s_out_driver); diff --git a/sound/soc/img/img-parallel-out.c b/sound/soc/img/img-parallel-out.c index 815e68a7048c..4ec63119d67c 100644 --- a/sound/soc/img/img-parallel-out.c +++ b/sound/soc/img/img-parallel-out.c @@ -311,7 +311,7 @@ static struct platform_driver img_prl_out_driver = { .pm = &img_prl_out_pm_ops }, .probe = img_prl_out_probe, - .remove_new = img_prl_out_dev_remove + .remove = img_prl_out_dev_remove }; module_platform_driver(img_prl_out_driver); diff --git a/sound/soc/img/img-spdif-in.c b/sound/soc/img/img-spdif-in.c index 9646e9d3f0bc..3c513f5b8c54 100644 --- a/sound/soc/img/img-spdif-in.c +++ b/sound/soc/img/img-spdif-in.c @@ -878,7 +878,7 @@ static struct platform_driver img_spdif_in_driver = { .pm = &img_spdif_in_pm_ops }, .probe = img_spdif_in_probe, - .remove_new = img_spdif_in_dev_remove + .remove = img_spdif_in_dev_remove }; module_platform_driver(img_spdif_in_driver); diff --git a/sound/soc/img/img-spdif-out.c b/sound/soc/img/img-spdif-out.c index dfa72afa946e..402695b5fc41 100644 --- a/sound/soc/img/img-spdif-out.c +++ b/sound/soc/img/img-spdif-out.c @@ -468,7 +468,7 @@ static struct platform_driver img_spdif_out_driver = { .pm = &img_spdif_out_pm_ops }, .probe = img_spdif_out_probe, - .remove_new = img_spdif_out_dev_remove + .remove = img_spdif_out_dev_remove }; module_platform_driver(img_spdif_out_driver); diff --git a/sound/soc/img/pistachio-internal-dac.c b/sound/soc/img/pistachio-internal-dac.c index da6251680e41..fdeceb271e7f 100644 --- a/sound/soc/img/pistachio-internal-dac.c +++ b/sound/soc/img/pistachio-internal-dac.c @@ -271,7 +271,7 @@ static struct platform_driver pistachio_internal_dac_plat_driver = { .pm = &pistachio_internal_dac_pm_ops }, .probe = pistachio_internal_dac_probe, - .remove_new = pistachio_internal_dac_remove + .remove = pistachio_internal_dac_remove }; module_platform_driver(pistachio_internal_dac_plat_driver); diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 8652b4a20020..373d68b4cf88 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -812,7 +812,7 @@ static struct platform_driver sst_platform_driver = { .pm = &sst_platform_pm, }, .probe = sst_platform_probe, - .remove_new = sst_platform_remove, + .remove = sst_platform_remove, }; module_platform_driver(sst_platform_driver); diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index 29d44c989e5f..9956dc63db74 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -358,7 +358,7 @@ static struct platform_driver sst_acpi_driver = { .pm = &intel_sst_pm, }, .probe = sst_acpi_probe, - .remove_new = sst_acpi_remove, + .remove = sst_acpi_remove, }; module_platform_driver(sst_acpi_driver); diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 3539c9ff0fd2..d3327bc237b5 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -709,7 +709,7 @@ static struct platform_driver snd_byt_cht_es8316_mc_driver = { .name = "bytcht_es8316", }, .probe = snd_byt_cht_es8316_mc_probe, - .remove_new = snd_byt_cht_es8316_mc_remove, + .remove = snd_byt_cht_es8316_mc_remove, }; module_platform_driver(snd_byt_cht_es8316_mc_driver); diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 4479825c08b5..2ed49acb4e36 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1918,7 +1918,7 @@ static struct platform_driver snd_byt_rt5640_mc_driver = { .name = "bytcr_rt5640", }, .probe = snd_byt_rt5640_mc_probe, - .remove_new = snd_byt_rt5640_mc_remove, + .remove = snd_byt_rt5640_mc_remove, }; module_platform_driver(snd_byt_rt5640_mc_driver); diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 1f54da98aacf..8e4b821efe92 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -1142,7 +1142,7 @@ static struct platform_driver snd_byt_rt5651_mc_driver = { .name = "bytcr_rt5651", }, .probe = snd_byt_rt5651_mc_probe, - .remove_new = snd_byt_rt5651_mc_remove, + .remove = snd_byt_rt5651_mc_remove, }; module_platform_driver(snd_byt_rt5651_mc_driver); diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index e5a7cc606aa9..0b10d89cb189 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -663,7 +663,7 @@ static struct platform_driver snd_byt_wm5102_mc_driver = { .name = "bytcr_wm5102", }, .probe = snd_byt_wm5102_mc_probe, - .remove_new = snd_byt_wm5102_mc_remove, + .remove = snd_byt_wm5102_mc_remove, }; module_platform_driver(snd_byt_wm5102_mc_driver); diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index f43bc20d6aae..d7c114858833 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -637,7 +637,7 @@ static struct platform_driver snd_cht_mc_driver = { .name = "cht-bsw-max98090", }, .probe = snd_cht_mc_probe, - .remove_new = snd_cht_mc_remove, + .remove = snd_cht_mc_remove, }; module_platform_driver(snd_cht_mc_driver) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 578271b4230a..fc998fe4b196 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -838,7 +838,7 @@ static struct platform_driver sof_es8336_driver = { .pm = &snd_soc_pm_ops, }, .probe = sof_es8336_probe, - .remove_new = sof_es8336_remove, + .remove = sof_es8336_remove, .id_table = board_ids, }; module_platform_driver(sof_es8336_driver); diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index b01cb2329542..5de883b9563e 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -430,7 +430,7 @@ static void sof_pcm512x_remove(struct platform_device *pdev) static struct platform_driver sof_audio = { .probe = sof_audio_probe, - .remove_new = sof_pcm512x_remove, + .remove = sof_pcm512x_remove, .driver = { .name = "sof_pcm512x", .pm = &snd_soc_pm_ops, diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 855109d22131..f10ed95770ea 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -1368,7 +1368,7 @@ static struct platform_driver sof_sdw_driver = { .pm = &snd_soc_pm_ops, }, .probe = mc_probe, - .remove_new = mc_remove, + .remove = mc_remove, .id_table = mc_id_table, }; diff --git a/sound/soc/intel/boards/sof_wm8804.c b/sound/soc/intel/boards/sof_wm8804.c index 0a5ce34d7f7b..facc6c32cbfe 100644 --- a/sound/soc/intel/boards/sof_wm8804.c +++ b/sound/soc/intel/boards/sof_wm8804.c @@ -294,7 +294,7 @@ static struct platform_driver sof_wm8804_driver = { .pm = &snd_soc_pm_ops, }, .probe = sof_wm8804_probe, - .remove_new = sof_wm8804_remove, + .remove = sof_wm8804_remove, }; module_platform_driver(sof_wm8804_driver); diff --git a/sound/soc/intel/catpt/device.c b/sound/soc/intel/catpt/device.c index 2e1fa79a04d4..2aa637124bec 100644 --- a/sound/soc/intel/catpt/device.c +++ b/sound/soc/intel/catpt/device.c @@ -374,7 +374,7 @@ MODULE_DEVICE_TABLE(acpi, catpt_ids); static struct platform_driver catpt_acpi_driver = { .probe = catpt_acpi_probe, - .remove_new = catpt_acpi_remove, + .remove = catpt_acpi_remove, .driver = { .name = "intel_catpt", .acpi_match_table = catpt_ids, diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d1eb90310afa..99bd066c7309 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -759,7 +759,7 @@ MODULE_DEVICE_TABLE(of, mvebu_audio_of_match); static struct platform_driver kirkwood_i2s_driver = { .probe = kirkwood_i2s_dev_probe, - .remove_new = kirkwood_i2s_dev_remove, + .remove = kirkwood_i2s_dev_remove, .driver = { .name = DRV_NAME, .of_match_table = of_match_ptr(mvebu_audio_of_match), diff --git a/sound/soc/mediatek/common/mtk-btcvsd.c b/sound/soc/mediatek/common/mtk-btcvsd.c index c12d170fa1de..d07f288f9752 100644 --- a/sound/soc/mediatek/common/mtk-btcvsd.c +++ b/sound/soc/mediatek/common/mtk-btcvsd.c @@ -1400,7 +1400,7 @@ static struct platform_driver mtk_btcvsd_snd_driver = { .of_match_table = mtk_btcvsd_snd_dt_match, }, .probe = mtk_btcvsd_snd_probe, - .remove_new = mtk_btcvsd_snd_remove, + .remove = mtk_btcvsd_snd_remove, }; module_platform_driver(mtk_btcvsd_snd_driver); diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index 6a17deb874df..5f11bc5438bd 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1473,7 +1473,7 @@ static struct platform_driver mt2701_afe_pcm_driver = { .pm = &mt2701_afe_pm_ops, }, .probe = mt2701_afe_pcm_dev_probe, - .remove_new = mt2701_afe_pcm_dev_remove, + .remove = mt2701_afe_pcm_dev_remove, }; module_platform_driver(mt2701_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c index c1dee119e93a..9159b42adf6a 100644 --- a/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c +++ b/sound/soc/mediatek/mt6797/mt6797-afe-pcm.c @@ -890,7 +890,7 @@ static struct platform_driver mt6797_afe_pcm_driver = { .pm = &mt6797_afe_pm_ops, }, .probe = mt6797_afe_pcm_dev_probe, - .remove_new = mt6797_afe_pcm_dev_remove, + .remove = mt6797_afe_pcm_dev_remove, }; module_platform_driver(mt6797_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c index 572ded279b53..d400bea0ef9c 100644 --- a/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c +++ b/sound/soc/mediatek/mt7986/mt7986-afe-pcm.c @@ -601,7 +601,7 @@ static struct platform_driver mt7986_afe_pcm_driver = { .pm = &mt7986_afe_pm_ops, }, .probe = mt7986_afe_pcm_dev_probe, - .remove_new = mt7986_afe_pcm_dev_remove, + .remove = mt7986_afe_pcm_dev_remove, }; module_platform_driver(mt7986_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c index b6291b7c811e..03250273ea9c 100644 --- a/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c +++ b/sound/soc/mediatek/mt8173/mt8173-afe-pcm.c @@ -1223,7 +1223,7 @@ static struct platform_driver mt8173_afe_pcm_driver = { .pm = &mt8173_afe_pm_ops, }, .probe = mt8173_afe_pcm_dev_probe, - .remove_new = mt8173_afe_pcm_dev_remove, + .remove = mt8173_afe_pcm_dev_remove, }; module_platform_driver(mt8173_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 25348fdf75fa..3f377ba4ad53 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1268,7 +1268,7 @@ static struct platform_driver mt8183_afe_pcm_driver = { .pm = &mt8183_afe_pm_ops, }, .probe = mt8183_afe_pcm_dev_probe, - .remove_new = mt8183_afe_pcm_dev_remove, + .remove = mt8183_afe_pcm_dev_remove, }; module_platform_driver(mt8183_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c index 424c5c68f78a..9b502f4cd6ea 100644 --- a/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c +++ b/sound/soc/mediatek/mt8192/mt8192-afe-pcm.c @@ -2325,7 +2325,7 @@ static struct platform_driver mt8192_afe_pcm_driver = { .pm = &mt8192_afe_pm_ops, }, .probe = mt8192_afe_pcm_dev_probe, - .remove_new = mt8192_afe_pcm_dev_remove, + .remove = mt8192_afe_pcm_dev_remove, }; module_platform_driver(mt8192_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index 38891d1bd18a..8016bfb35015 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -3199,7 +3199,7 @@ static struct platform_driver mt8195_afe_pcm_driver = { .pm = &mt8195_afe_pm_ops, }, .probe = mt8195_afe_pcm_dev_probe, - .remove_new = mt8195_afe_pcm_dev_remove, + .remove = mt8195_afe_pcm_dev_remove, }; module_platform_driver(mt8195_afe_pcm_driver); diff --git a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c index df6dd8c5bbe5..49efcdea975e 100644 --- a/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c +++ b/sound/soc/mediatek/mt8365/mt8365-afe-pcm.c @@ -2264,7 +2264,7 @@ static struct platform_driver mt8365_afe_pcm_driver = { .pm = &mt8365_afe_pm_ops, }, .probe = mt8365_afe_pcm_dev_probe, - .remove_new = mt8365_afe_pcm_dev_remove, + .remove = mt8365_afe_pcm_dev_remove, }; module_platform_driver(mt8365_afe_pcm_driver); diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 5d1419ed7a62..f2890111c1d2 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -345,7 +345,7 @@ MODULE_DEVICE_TABLE(of, aiu_of_match); static struct platform_driver aiu_pdrv = { .probe = aiu_probe, - .remove_new = aiu_remove, + .remove = aiu_remove, .driver = { .name = "meson-aiu", .of_match_table = aiu_of_match, diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 646ab87afac2..59deb332bd35 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -360,7 +360,7 @@ MODULE_DEVICE_TABLE(of, axg_card_of_match); static struct platform_driver axg_card_pdrv = { .probe = meson_card_probe, - .remove_new = meson_card_remove, + .remove = meson_card_remove, .driver = { .name = "axg-sound-card", .of_match_table = axg_card_of_match, diff --git a/sound/soc/meson/gx-card.c b/sound/soc/meson/gx-card.c index 7edca3e49c8f..455f6bfc9f8f 100644 --- a/sound/soc/meson/gx-card.c +++ b/sound/soc/meson/gx-card.c @@ -129,7 +129,7 @@ MODULE_DEVICE_TABLE(of, gx_card_of_match); static struct platform_driver gx_card_pdrv = { .probe = meson_card_probe, - .remove_new = meson_card_remove, + .remove = meson_card_remove, .driver = { .name = "gx-sound-card", .of_match_table = gx_card_of_match, diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 310e3ac77424..a41a13ae38a5 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -185,7 +185,7 @@ static struct platform_driver mxs_sgtl5000_audio_driver = { .of_match_table = mxs_sgtl5000_dt_ids, }, .probe = mxs_sgtl5000_probe, - .remove_new = mxs_sgtl5000_remove, + .remove = mxs_sgtl5000_remove, }; module_platform_driver(mxs_sgtl5000_audio_driver); diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index abfaf3cdf5bb..73f36c9dd35c 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -574,7 +574,7 @@ static struct platform_driver asoc_mmp_sspa_driver = { .of_match_table = of_match_ptr(mmp_sspa_of_match), }, .probe = asoc_mmp_sspa_probe, - .remove_new = asoc_mmp_sspa_remove, + .remove = asoc_mmp_sspa_remove, }; module_platform_driver(asoc_mmp_sspa_driver); diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 80e0ea0ec9fb..78f50032afc5 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -286,7 +286,7 @@ static DEFINE_SIMPLE_DEV_PM_OPS(pxa2xx_ac97_pm_ops, static struct platform_driver pxa2xx_ac97_driver = { .probe = pxa2xx_ac97_dev_probe, - .remove_new = pxa2xx_ac97_dev_remove, + .remove = pxa2xx_ac97_dev_remove, .driver = { .name = "pxa2xx-ac97", .pm = &pxa2xx_ac97_pm_ops, diff --git a/sound/soc/qcom/lpass-apq8016.c b/sound/soc/qcom/lpass-apq8016.c index 9005c85f8c54..b8f23414eb77 100644 --- a/sound/soc/qcom/lpass-apq8016.c +++ b/sound/soc/qcom/lpass-apq8016.c @@ -300,7 +300,7 @@ static struct platform_driver apq8016_lpass_cpu_platform_driver = { .of_match_table = of_match_ptr(apq8016_lpass_cpu_device_id), }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, }; module_platform_driver(apq8016_lpass_cpu_platform_driver); diff --git a/sound/soc/qcom/lpass-ipq806x.c b/sound/soc/qcom/lpass-ipq806x.c index 5c874139f39d..e57d29ea4ce7 100644 --- a/sound/soc/qcom/lpass-ipq806x.c +++ b/sound/soc/qcom/lpass-ipq806x.c @@ -172,7 +172,7 @@ static struct platform_driver ipq806x_lpass_cpu_platform_driver = { .of_match_table = of_match_ptr(ipq806x_lpass_cpu_device_id), }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, }; module_platform_driver(ipq806x_lpass_cpu_platform_driver); diff --git a/sound/soc/qcom/lpass-sc7180.c b/sound/soc/qcom/lpass-sc7180.c index e6bcdf6ed796..fbead6af3d95 100644 --- a/sound/soc/qcom/lpass-sc7180.c +++ b/sound/soc/qcom/lpass-sc7180.c @@ -315,7 +315,7 @@ static struct platform_driver sc7180_lpass_cpu_platform_driver = { .pm = &sc7180_lpass_pm_ops, }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, .shutdown = asoc_qcom_lpass_cpu_platform_shutdown, }; diff --git a/sound/soc/qcom/lpass-sc7280.c b/sound/soc/qcom/lpass-sc7280.c index 47c622327a8d..7cd3e291382a 100644 --- a/sound/soc/qcom/lpass-sc7280.c +++ b/sound/soc/qcom/lpass-sc7280.c @@ -445,7 +445,7 @@ static struct platform_driver sc7280_lpass_cpu_platform_driver = { .pm = &sc7280_lpass_pm_ops, }, .probe = asoc_qcom_lpass_cpu_platform_probe, - .remove_new = asoc_qcom_lpass_cpu_platform_remove, + .remove = asoc_qcom_lpass_cpu_platform_remove, .shutdown = asoc_qcom_lpass_cpu_platform_shutdown, }; diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c index 81fde0681f95..90228699ba7d 100644 --- a/sound/soc/qcom/qdsp6/q6routing.c +++ b/sound/soc/qcom/qdsp6/q6routing.c @@ -1161,7 +1161,7 @@ static struct platform_driver q6pcm_routing_platform_driver = { .of_match_table = of_match_ptr(q6pcm_routing_device_id), }, .probe = q6pcm_routing_probe, - .remove_new = q6pcm_routing_remove, + .remove = q6pcm_routing_remove, }; module_platform_driver(q6pcm_routing_platform_driver); diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index b378f870b3ad..4315da4a47c1 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -866,7 +866,7 @@ static const struct dev_pm_ops rockchip_i2s_pm_ops = { static struct platform_driver rockchip_i2s_driver = { .probe = rockchip_i2s_probe, - .remove_new = rockchip_i2s_remove, + .remove = rockchip_i2s_remove, .driver = { .name = DRV_NAME, .of_match_table = of_match_ptr(rockchip_i2s_match), diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index ee517d7b5b7b..d1f28699652f 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -1423,7 +1423,7 @@ static const struct dev_pm_ops rockchip_i2s_tdm_pm_ops = { static struct platform_driver rockchip_i2s_tdm_driver = { .probe = rockchip_i2s_tdm_probe, - .remove_new = rockchip_i2s_tdm_remove, + .remove = rockchip_i2s_tdm_remove, .driver = { .name = DRV_NAME, .of_match_table = rockchip_i2s_tdm_match, diff --git a/sound/soc/rockchip/rockchip_pdm.c b/sound/soc/rockchip/rockchip_pdm.c index d16a4a67a6a2..cae91108f7a8 100644 --- a/sound/soc/rockchip/rockchip_pdm.c +++ b/sound/soc/rockchip/rockchip_pdm.c @@ -703,7 +703,7 @@ static const struct dev_pm_ops rockchip_pdm_pm_ops = { static struct platform_driver rockchip_pdm_driver = { .probe = rockchip_pdm_probe, - .remove_new = rockchip_pdm_remove, + .remove = rockchip_pdm_remove, .driver = { .name = "rockchip-pdm", .of_match_table = of_match_ptr(rockchip_pdm_match), diff --git a/sound/soc/rockchip/rockchip_rt5645.c b/sound/soc/rockchip/rockchip_rt5645.c index 449f62820045..b085d80ea2e4 100644 --- a/sound/soc/rockchip/rockchip_rt5645.c +++ b/sound/soc/rockchip/rockchip_rt5645.c @@ -233,7 +233,7 @@ MODULE_DEVICE_TABLE(of, rockchip_rt5645_of_match); static struct platform_driver snd_rk_mc_driver = { .probe = snd_rk_mc_probe, - .remove_new = snd_rk_mc_remove, + .remove = snd_rk_mc_remove, .driver = { .name = DRV_NAME, .pm = &snd_soc_pm_ops, diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index eb9d5dee196e..d87c0e4f6f91 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -380,7 +380,7 @@ static const struct dev_pm_ops rk_spdif_pm_ops = { static struct platform_driver rk_spdif_driver = { .probe = rk_spdif_probe, - .remove_new = rk_spdif_remove, + .remove = rk_spdif_remove, .driver = { .name = "rockchip-spdif", .of_match_table = of_match_ptr(rk_spdif_match), diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index f02873b6ce7f..9619f550608c 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -207,7 +207,7 @@ static struct platform_driver arndale_audio_driver = { .of_match_table = arndale_audio_of_match, }, .probe = arndale_audio_probe, - .remove_new = arndale_audio_remove, + .remove = arndale_audio_remove, }; module_platform_driver(arndale_audio_driver); diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 1bcabb114e29..8f6deb06e234 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1741,7 +1741,7 @@ static const struct dev_pm_ops samsung_i2s_pm = { static struct platform_driver samsung_i2s_driver = { .probe = samsung_i2s_probe, - .remove_new = samsung_i2s_remove, + .remove = samsung_i2s_remove, .id_table = samsung_i2s_driver_ids, .driver = { .name = "samsung-i2s", diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 110ae14dd7ea..ed865cc07e2e 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -341,7 +341,7 @@ static struct platform_driver odroid_audio_driver = { .pm = &snd_soc_pm_ops, }, .probe = odroid_audio_probe, - .remove_new = odroid_audio_remove, + .remove = odroid_audio_remove, }; module_platform_driver(odroid_audio_driver); diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 573b2dee7f07..a03ba9374c2e 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -590,7 +590,7 @@ static void s3c_pcm_dev_remove(struct platform_device *pdev) static struct platform_driver s3c_pcm_driver = { .probe = s3c_pcm_dev_probe, - .remove_new = s3c_pcm_dev_remove, + .remove = s3c_pcm_dev_remove, .driver = { .name = "samsung-pcm", }, diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index aad0f9b4d4fc..4bbe7bcdb845 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -245,7 +245,7 @@ static struct platform_driver snow_driver = { .of_match_table = snow_of_match, }, .probe = snow_probe, - .remove_new = snow_remove, + .remove = snow_remove, }; module_platform_driver(snow_driver); diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index f44e3180e8d3..235d0063d1b3 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -476,7 +476,7 @@ static void spdif_remove(struct platform_device *pdev) static struct platform_driver samsung_spdif_driver = { .probe = spdif_probe, - .remove_new = spdif_remove, + .remove = spdif_remove, .driver = { .name = "samsung-spdif", }, diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 087e379aa3bc..221ce91f1950 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -2107,7 +2107,7 @@ static struct platform_driver fsi_driver = { .of_match_table = fsi_of_match, }, .probe = fsi_probe, - .remove_new = fsi_remove, + .remove = fsi_remove, .id_table = fsi_id_table, }; diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index cc200f45826c..db618c09d1e0 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -334,7 +334,7 @@ static struct platform_driver hac_pcm_driver = { }, .probe = hac_soc_platform_probe, - .remove_new = hac_soc_platform_remove, + .remove = hac_soc_platform_remove, }; module_platform_driver(hac_pcm_driver); diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 15cb5e7008f9..9784718a2b6f 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -2087,7 +2087,7 @@ static struct platform_driver rsnd_driver = { .of_match_table = rsnd_of_match, }, .probe = rsnd_probe, - .remove_new = rsnd_remove, + .remove = rsnd_remove, }; module_platform_driver(rsnd_driver); diff --git a/sound/soc/sh/rz-ssi.c b/sound/soc/sh/rz-ssi.c index d0bf0487bf1b..040ce0431fd2 100644 --- a/sound/soc/sh/rz-ssi.c +++ b/sound/soc/sh/rz-ssi.c @@ -1204,7 +1204,7 @@ static struct platform_driver rz_ssi_driver = { .of_match_table = rz_ssi_of_match, }, .probe = rz_ssi_probe, - .remove_new = rz_ssi_remove, + .remove = rz_ssi_remove, }; module_platform_driver(rz_ssi_driver); diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index d0b5c543fd2f..7e771a164a80 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -788,7 +788,7 @@ static struct platform_driver siu_driver = { .name = "siu-pcm-audio", }, .probe = siu_probe, - .remove_new = siu_remove, + .remove = siu_remove, }; module_platform_driver(siu_driver); diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c index 9f24e3c283dd..f09ee0dea2cc 100644 --- a/sound/soc/sof/imx/imx8.c +++ b/sound/soc/sof/imx/imx8.c @@ -658,7 +658,7 @@ MODULE_DEVICE_TABLE(of, sof_of_imx8_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_imx8_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .driver = { .name = "sof-audio-of-imx8", .pm = &sof_of_pm, diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index cdd1e79ef9f6..01d3ad3314f3 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -505,7 +505,7 @@ MODULE_DEVICE_TABLE(of, sof_of_imx8m_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_imx8m_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .driver = { .name = "sof-audio-of-imx8m", .pm = &sof_of_pm, diff --git a/sound/soc/sof/imx/imx8ulp.c b/sound/soc/sof/imx/imx8ulp.c index 2585b1beef23..e5eee1c9f6da 100644 --- a/sound/soc/sof/imx/imx8ulp.c +++ b/sound/soc/sof/imx/imx8ulp.c @@ -507,7 +507,7 @@ MODULE_DEVICE_TABLE(of, sof_of_imx8ulp_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_imx8ulp_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .driver = { .name = "sof-audio-of-imx8ulp", .pm = &sof_of_pm, diff --git a/sound/soc/sof/intel/bdw.c b/sound/soc/sof/intel/bdw.c index 7f18080e4e19..322ff118f0f6 100644 --- a/sound/soc/sof/intel/bdw.c +++ b/sound/soc/sof/intel/bdw.c @@ -684,7 +684,7 @@ static int sof_broadwell_probe(struct platform_device *pdev) /* acpi_driver definition */ static struct platform_driver snd_sof_acpi_intel_bdw_driver = { .probe = sof_broadwell_probe, - .remove_new = sof_acpi_remove, + .remove = sof_acpi_remove, .driver = { .name = "sof-audio-acpi-intel-bdw", .pm = &sof_acpi_pm, diff --git a/sound/soc/sof/intel/byt.c b/sound/soc/sof/intel/byt.c index 7a57e162fb1c..f531710cf94e 100644 --- a/sound/soc/sof/intel/byt.c +++ b/sound/soc/sof/intel/byt.c @@ -465,7 +465,7 @@ static int sof_baytrail_probe(struct platform_device *pdev) /* acpi_driver definition */ static struct platform_driver snd_sof_acpi_intel_byt_driver = { .probe = sof_baytrail_probe, - .remove_new = sof_acpi_remove, + .remove = sof_acpi_remove, .driver = { .name = "sof-audio-acpi-intel-byt", .pm = &sof_acpi_pm, diff --git a/sound/soc/sof/mediatek/mt8186/mt8186.c b/sound/soc/sof/mediatek/mt8186/mt8186.c index 74522400207e..9466f7d2e535 100644 --- a/sound/soc/sof/mediatek/mt8186/mt8186.c +++ b/sound/soc/sof/mediatek/mt8186/mt8186.c @@ -656,7 +656,7 @@ MODULE_DEVICE_TABLE(of, sof_of_mt8186_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_mt8186_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .shutdown = sof_of_shutdown, .driver = { .name = "sof-audio-of-mt8186", diff --git a/sound/soc/sof/mediatek/mt8195/mt8195.c b/sound/soc/sof/mediatek/mt8195/mt8195.c index 82d221f53a46..5b4423ed8023 100644 --- a/sound/soc/sof/mediatek/mt8195/mt8195.c +++ b/sound/soc/sof/mediatek/mt8195/mt8195.c @@ -612,7 +612,7 @@ MODULE_DEVICE_TABLE(of, sof_of_mt8195_ids); /* DT driver definition */ static struct platform_driver snd_sof_of_mt8195_driver = { .probe = sof_of_probe, - .remove_new = sof_of_remove, + .remove = sof_of_remove, .shutdown = sof_of_shutdown, .driver = { .name = "sof-audio-of-mt8195", diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index 688419c6b092..814a1cde1d35 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -993,7 +993,7 @@ MODULE_DEVICE_TABLE(of, sprd_mcdt_of_match); static struct platform_driver sprd_mcdt_driver = { .probe = sprd_mcdt_probe, - .remove_new = sprd_mcdt_remove, + .remove = sprd_mcdt_remove, .driver = { .name = "sprd-mcdt", .of_match_table = sprd_mcdt_of_match, diff --git a/sound/soc/starfive/jh7110_pwmdac.c b/sound/soc/starfive/jh7110_pwmdac.c index 4a4dd431b82b..a603dd17931c 100644 --- a/sound/soc/starfive/jh7110_pwmdac.c +++ b/sound/soc/starfive/jh7110_pwmdac.c @@ -516,7 +516,7 @@ static struct platform_driver jh7110_pwmdac_driver = { .pm = pm_ptr(&jh7110_pwmdac_pm_ops), }, .probe = jh7110_pwmdac_probe, - .remove_new = jh7110_pwmdac_remove, + .remove = jh7110_pwmdac_remove, }; module_platform_driver(jh7110_pwmdac_driver); diff --git a/sound/soc/starfive/jh7110_tdm.c b/sound/soc/starfive/jh7110_tdm.c index 1e0ff6720747..d38090e68df5 100644 --- a/sound/soc/starfive/jh7110_tdm.c +++ b/sound/soc/starfive/jh7110_tdm.c @@ -660,7 +660,7 @@ static struct platform_driver jh7110_tdm_driver = { .pm = pm_ptr(&jh7110_tdm_pm_ops), }, .probe = jh7110_tdm_probe, - .remove_new = jh7110_tdm_dev_remove, + .remove = jh7110_tdm_dev_remove, }; module_platform_driver(jh7110_tdm_driver); diff --git a/sound/soc/stm/stm32_adfsdm.c b/sound/soc/stm/stm32_adfsdm.c index fb5dd9a68bea..9351727dce1a 100644 --- a/sound/soc/stm/stm32_adfsdm.c +++ b/sound/soc/stm/stm32_adfsdm.c @@ -398,7 +398,7 @@ static struct platform_driver stm32_adfsdm_driver = { .of_match_table = stm32_adfsdm_of_match, }, .probe = stm32_adfsdm_probe, - .remove_new = stm32_adfsdm_remove, + .remove = stm32_adfsdm_remove, }; module_platform_driver(stm32_adfsdm_driver); diff --git a/sound/soc/stm/stm32_i2s.c b/sound/soc/stm/stm32_i2s.c index a96aa308681a..faa00103ee7f 100644 --- a/sound/soc/stm/stm32_i2s.c +++ b/sound/soc/stm/stm32_i2s.c @@ -1216,7 +1216,7 @@ static struct platform_driver stm32_i2s_driver = { .pm = &stm32_i2s_pm_ops, }, .probe = stm32_i2s_probe, - .remove_new = stm32_i2s_remove, + .remove = stm32_i2s_remove, }; module_platform_driver(stm32_i2s_driver); diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index ad2492efb1cd..7bc4a96b7503 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1619,7 +1619,7 @@ static struct platform_driver stm32_sai_sub_driver = { .pm = &stm32_sai_sub_pm_ops, }, .probe = stm32_sai_sub_probe, - .remove_new = stm32_sai_sub_remove, + .remove = stm32_sai_sub_remove, }; module_platform_driver(stm32_sai_sub_driver); diff --git a/sound/soc/stm/stm32_spdifrx.c b/sound/soc/stm/stm32_spdifrx.c index 9eed3c57e3f1..d1b32ba1e1a2 100644 --- a/sound/soc/stm/stm32_spdifrx.c +++ b/sound/soc/stm/stm32_spdifrx.c @@ -1072,7 +1072,7 @@ static struct platform_driver stm32_spdifrx_driver = { .pm = &stm32_spdifrx_pm_ops, }, .probe = stm32_spdifrx_probe, - .remove_new = stm32_spdifrx_remove, + .remove = stm32_spdifrx_remove, }; module_platform_driver(stm32_spdifrx_driver); diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 25af47b63bdd..330bc0c09f56 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1828,7 +1828,7 @@ static struct platform_driver sun4i_codec_driver = { .of_match_table = sun4i_codec_of_match, }, .probe = sun4i_codec_probe, - .remove_new = sun4i_codec_remove, + .remove = sun4i_codec_remove, }; module_platform_driver(sun4i_codec_driver); diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index 3af0b2aab291..40de99a34bc3 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -1684,7 +1684,7 @@ static const struct dev_pm_ops sun4i_i2s_pm_ops = { static struct platform_driver sun4i_i2s_driver = { .probe = sun4i_i2s_probe, - .remove_new = sun4i_i2s_remove, + .remove = sun4i_i2s_remove, .driver = { .name = "sun4i-i2s", .of_match_table = sun4i_i2s_match, diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index f41c30955857..0aa416423246 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -726,7 +726,7 @@ static struct platform_driver sun4i_spdif_driver = { .pm = &sun4i_spdif_pm, }, .probe = sun4i_spdif_probe, - .remove_new = sun4i_spdif_remove, + .remove = sun4i_spdif_remove, }; module_platform_driver(sun4i_spdif_driver); diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c index 884394ddaf86..3e751b5694fe 100644 --- a/sound/soc/sunxi/sun50i-dmic.c +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -426,7 +426,7 @@ static struct platform_driver sun50i_dmic_driver = { .pm = &sun50i_dmic_pm, }, .probe = sun50i_dmic_probe, - .remove_new = sun50i_dmic_remove, + .remove = sun50i_dmic_remove, }; module_platform_driver(sun50i_dmic_driver); diff --git a/sound/soc/sunxi/sun8i-codec.c b/sound/soc/sunxi/sun8i-codec.c index b5dafb749c3f..8c645e04d571 100644 --- a/sound/soc/sunxi/sun8i-codec.c +++ b/sound/soc/sunxi/sun8i-codec.c @@ -1713,7 +1713,7 @@ static struct platform_driver sun8i_codec_driver = { .pm = &sun8i_codec_pm_ops, }, .probe = sun8i_codec_probe, - .remove_new = sun8i_codec_remove, + .remove = sun8i_codec_remove, }; module_platform_driver(sun8i_codec_driver); diff --git a/sound/soc/tegra/tegra186_asrc.c b/sound/soc/tegra/tegra186_asrc.c index 22af5135d77a..d914dba56013 100644 --- a/sound/soc/tegra/tegra186_asrc.c +++ b/sound/soc/tegra/tegra186_asrc.c @@ -1034,7 +1034,7 @@ static struct platform_driver tegra186_asrc_driver = { .pm = &tegra186_asrc_pm_ops, }, .probe = tegra186_asrc_platform_probe, - .remove_new = tegra186_asrc_platform_remove, + .remove = tegra186_asrc_platform_remove, }; module_platform_driver(tegra186_asrc_driver) diff --git a/sound/soc/tegra/tegra186_dspk.c b/sound/soc/tegra/tegra186_dspk.c index 21cd41fec7a9..508128b7783e 100644 --- a/sound/soc/tegra/tegra186_dspk.c +++ b/sound/soc/tegra/tegra186_dspk.c @@ -542,7 +542,7 @@ static struct platform_driver tegra186_dspk_driver = { .pm = &tegra186_dspk_pm_ops, }, .probe = tegra186_dspk_platform_probe, - .remove_new = tegra186_dspk_platform_remove, + .remove = tegra186_dspk_platform_remove, }; module_platform_driver(tegra186_dspk_driver); diff --git a/sound/soc/tegra/tegra20_ac97.c b/sound/soc/tegra/tegra20_ac97.c index 8011afe93c96..08c58e8f3c22 100644 --- a/sound/soc/tegra/tegra20_ac97.c +++ b/sound/soc/tegra/tegra20_ac97.c @@ -448,7 +448,7 @@ static struct platform_driver tegra20_ac97_driver = { .of_match_table = tegra20_ac97_of_match, }, .probe = tegra20_ac97_platform_probe, - .remove_new = tegra20_ac97_platform_remove, + .remove = tegra20_ac97_platform_remove, }; module_platform_driver(tegra20_ac97_driver); diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index f11618e8f13e..3b9823d1a87a 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -500,7 +500,7 @@ static struct platform_driver tegra20_i2s_driver = { .pm = &tegra20_i2s_pm_ops, }, .probe = tegra20_i2s_platform_probe, - .remove_new = tegra20_i2s_platform_remove, + .remove = tegra20_i2s_platform_remove, }; module_platform_driver(tegra20_i2s_driver); diff --git a/sound/soc/tegra/tegra210_admaif.c b/sound/soc/tegra/tegra210_admaif.c index 9f9334e48049..a866aeb2719d 100644 --- a/sound/soc/tegra/tegra210_admaif.c +++ b/sound/soc/tegra/tegra210_admaif.c @@ -856,7 +856,7 @@ static const struct dev_pm_ops tegra_admaif_pm_ops = { static struct platform_driver tegra_admaif_driver = { .probe = tegra_admaif_probe, - .remove_new = tegra_admaif_remove, + .remove = tegra_admaif_remove, .driver = { .name = "tegra210-admaif", .of_match_table = tegra_admaif_of_match, diff --git a/sound/soc/tegra/tegra210_adx.c b/sound/soc/tegra/tegra210_adx.c index d2530443a221..109f763fe211 100644 --- a/sound/soc/tegra/tegra210_adx.c +++ b/sound/soc/tegra/tegra210_adx.c @@ -532,7 +532,7 @@ static struct platform_driver tegra210_adx_driver = { .pm = &tegra210_adx_pm_ops, }, .probe = tegra210_adx_platform_probe, - .remove_new = tegra210_adx_platform_remove, + .remove = tegra210_adx_platform_remove, }; module_platform_driver(tegra210_adx_driver); diff --git a/sound/soc/tegra/tegra210_ahub.c b/sound/soc/tegra/tegra210_ahub.c index ab3c6b2544d2..1920b996e9aa 100644 --- a/sound/soc/tegra/tegra210_ahub.c +++ b/sound/soc/tegra/tegra210_ahub.c @@ -1416,7 +1416,7 @@ static const struct dev_pm_ops tegra_ahub_pm_ops = { static struct platform_driver tegra_ahub_driver = { .probe = tegra_ahub_probe, - .remove_new = tegra_ahub_remove, + .remove = tegra_ahub_remove, .driver = { .name = "tegra210-ahub", .of_match_table = tegra_ahub_of_match, diff --git a/sound/soc/tegra/tegra210_amx.c b/sound/soc/tegra/tegra210_amx.c index 91e405909e0f..38a2d6ec033b 100644 --- a/sound/soc/tegra/tegra210_amx.c +++ b/sound/soc/tegra/tegra210_amx.c @@ -589,7 +589,7 @@ static struct platform_driver tegra210_amx_driver = { .pm = &tegra210_amx_pm_ops, }, .probe = tegra210_amx_platform_probe, - .remove_new = tegra210_amx_platform_remove, + .remove = tegra210_amx_platform_remove, }; module_platform_driver(tegra210_amx_driver); diff --git a/sound/soc/tegra/tegra210_dmic.c b/sound/soc/tegra/tegra210_dmic.c index e53c0278ae9a..d9b577f146dc 100644 --- a/sound/soc/tegra/tegra210_dmic.c +++ b/sound/soc/tegra/tegra210_dmic.c @@ -559,7 +559,7 @@ static struct platform_driver tegra210_dmic_driver = { .pm = &tegra210_dmic_pm_ops, }, .probe = tegra210_dmic_probe, - .remove_new = tegra210_dmic_remove, + .remove = tegra210_dmic_remove, }; module_platform_driver(tegra210_dmic_driver) diff --git a/sound/soc/tegra/tegra210_i2s.c b/sound/soc/tegra/tegra210_i2s.c index e93ceb7afb4c..a3908b15dfdc 100644 --- a/sound/soc/tegra/tegra210_i2s.c +++ b/sound/soc/tegra/tegra210_i2s.c @@ -1019,7 +1019,7 @@ static struct platform_driver tegra210_i2s_driver = { .pm = &tegra210_i2s_pm_ops, }, .probe = tegra210_i2s_probe, - .remove_new = tegra210_i2s_remove, + .remove = tegra210_i2s_remove, }; module_platform_driver(tegra210_i2s_driver) diff --git a/sound/soc/tegra/tegra210_mixer.c b/sound/soc/tegra/tegra210_mixer.c index 024614f6ec0b..e07e2f1d2f70 100644 --- a/sound/soc/tegra/tegra210_mixer.c +++ b/sound/soc/tegra/tegra210_mixer.c @@ -674,7 +674,7 @@ static struct platform_driver tegra210_mixer_driver = { .pm = &tegra210_mixer_pm_ops, }, .probe = tegra210_mixer_platform_probe, - .remove_new = tegra210_mixer_platform_remove, + .remove = tegra210_mixer_platform_remove, }; module_platform_driver(tegra210_mixer_driver); diff --git a/sound/soc/tegra/tegra210_mvc.c b/sound/soc/tegra/tegra210_mvc.c index b89f5edafa03..4ead52564ab6 100644 --- a/sound/soc/tegra/tegra210_mvc.c +++ b/sound/soc/tegra/tegra210_mvc.c @@ -766,7 +766,7 @@ static struct platform_driver tegra210_mvc_driver = { .pm = &tegra210_mvc_pm_ops, }, .probe = tegra210_mvc_platform_probe, - .remove_new = tegra210_mvc_platform_remove, + .remove = tegra210_mvc_platform_remove, }; module_platform_driver(tegra210_mvc_driver) diff --git a/sound/soc/tegra/tegra210_ope.c b/sound/soc/tegra/tegra210_ope.c index 136ed17f3650..e2bc604e8b79 100644 --- a/sound/soc/tegra/tegra210_ope.c +++ b/sound/soc/tegra/tegra210_ope.c @@ -407,7 +407,7 @@ static struct platform_driver tegra210_ope_driver = { .pm = &tegra210_ope_pm_ops, }, .probe = tegra210_ope_probe, - .remove_new = tegra210_ope_remove, + .remove = tegra210_ope_remove, }; module_platform_driver(tegra210_ope_driver) diff --git a/sound/soc/tegra/tegra210_sfc.c b/sound/soc/tegra/tegra210_sfc.c index 028747c44f37..e16bbb44cc77 100644 --- a/sound/soc/tegra/tegra210_sfc.c +++ b/sound/soc/tegra/tegra210_sfc.c @@ -3631,7 +3631,7 @@ static struct platform_driver tegra210_sfc_driver = { .pm = &tegra210_sfc_pm_ops, }, .probe = tegra210_sfc_platform_probe, - .remove_new = tegra210_sfc_platform_remove, + .remove = tegra210_sfc_platform_remove, }; module_platform_driver(tegra210_sfc_driver) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index d2e8078e444a..c9b52f54cea8 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -608,7 +608,7 @@ static const struct dev_pm_ops tegra30_ahub_pm_ops = { static struct platform_driver tegra30_ahub_driver = { .probe = tegra30_ahub_probe, - .remove_new = tegra30_ahub_remove, + .remove = tegra30_ahub_remove, .driver = { .name = DRV_NAME, .of_match_table = tegra30_ahub_of_match, diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index a8ff51d12edb..0d3badfbe143 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -560,7 +560,7 @@ static struct platform_driver tegra30_i2s_driver = { .pm = &tegra30_i2s_pm_ops, }, .probe = tegra30_i2s_platform_probe, - .remove_new = tegra30_i2s_platform_remove, + .remove = tegra30_i2s_platform_remove, }; module_platform_driver(tegra30_i2s_driver); diff --git a/sound/soc/tegra/tegra_audio_graph_card.c b/sound/soc/tegra/tegra_audio_graph_card.c index feba9d42bbc5..8b48813c2c59 100644 --- a/sound/soc/tegra/tegra_audio_graph_card.c +++ b/sound/soc/tegra/tegra_audio_graph_card.c @@ -248,7 +248,7 @@ static struct platform_driver tegra_audio_graph_card = { .of_match_table = graph_of_tegra_match, }, .probe = tegra_audio_graph_probe, - .remove_new = simple_util_remove, + .remove = simple_util_remove, }; module_platform_driver(tegra_audio_graph_card); diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c index 76bda188e992..94645f275495 100644 --- a/sound/soc/ti/ams-delta.c +++ b/sound/soc/ti/ams-delta.c @@ -595,7 +595,7 @@ static struct platform_driver ams_delta_driver = { .name = DRV_NAME, }, .probe = ams_delta_probe, - .remove_new = ams_delta_remove, + .remove = ams_delta_remove, }; module_platform_driver(ams_delta_driver); diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 0f15a743c798..d682b98d6848 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -920,7 +920,7 @@ MODULE_DEVICE_TABLE(of, davinci_i2s_match); static struct platform_driver davinci_mcbsp_driver = { .probe = davinci_i2s_probe, - .remove_new = davinci_i2s_remove, + .remove = davinci_i2s_remove, .driver = { .name = "davinci-mcbsp", .of_match_table = of_match_ptr(davinci_i2s_match), diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 2b1ed91a736c..a0b8cca31cba 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -2535,7 +2535,7 @@ static const struct dev_pm_ops davinci_mcasp_pm_ops = { static struct platform_driver davinci_mcasp_driver = { .probe = davinci_mcasp_probe, - .remove_new = davinci_mcasp_remove, + .remove = davinci_mcasp_remove, .driver = { .name = "davinci-mcasp", .pm = &davinci_mcasp_pm_ops, diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c index 2110ffe5281c..411970399271 100644 --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -1430,7 +1430,7 @@ static struct platform_driver asoc_mcbsp_driver = { }, .probe = asoc_mcbsp_probe, - .remove_new = asoc_mcbsp_remove, + .remove = asoc_mcbsp_remove, }; module_platform_driver(asoc_mcbsp_driver); diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c index 01cc3b961999..a041ce8e23a4 100644 --- a/sound/soc/uniphier/aio-ld11.c +++ b/sound/soc/uniphier/aio-ld11.c @@ -347,7 +347,7 @@ static struct platform_driver uniphier_aio_driver = { .of_match_table = of_match_ptr(uniphier_aio_of_match), }, .probe = uniphier_aio_probe, - .remove_new = uniphier_aio_remove, + .remove = uniphier_aio_remove, }; module_platform_driver(uniphier_aio_driver); diff --git a/sound/soc/uniphier/aio-pxs2.c b/sound/soc/uniphier/aio-pxs2.c index fba13a212bdb..889f64b2c01f 100644 --- a/sound/soc/uniphier/aio-pxs2.c +++ b/sound/soc/uniphier/aio-pxs2.c @@ -256,7 +256,7 @@ static struct platform_driver uniphier_aio_driver = { .of_match_table = of_match_ptr(uniphier_aio_of_match), }, .probe = uniphier_aio_probe, - .remove_new = uniphier_aio_remove, + .remove = uniphier_aio_remove, }; module_platform_driver(uniphier_aio_driver); diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index d90b3e4b0104..662e45882c90 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -560,7 +560,7 @@ static struct platform_driver evea_codec_driver = { .of_match_table = of_match_ptr(evea_of_match), }, .probe = evea_probe, - .remove_new = evea_remove, + .remove = evea_remove, }; module_platform_driver(evea_codec_driver); diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index e0ab4534fe3e..ae6d326167d1 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -157,7 +157,7 @@ static struct platform_driver snd_soc_mop500_driver = { .of_match_table = snd_soc_mop500_match, }, .probe = mop500_probe, - .remove_new = mop500_remove, + .remove = mop500_remove, }; module_platform_driver(snd_soc_mop500_driver); diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 3fd13e8dd110..a2dd739fdf2d 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -816,7 +816,7 @@ static struct platform_driver msp_i2s_driver = { .of_match_table = ux500_msp_i2s_match, }, .probe = ux500_msp_drv_probe, - .remove_new = ux500_msp_drv_remove, + .remove = ux500_msp_drv_remove, }; module_platform_driver(msp_i2s_driver); diff --git a/sound/soc/xilinx/xlnx_formatter_pcm.c b/sound/soc/xilinx/xlnx_formatter_pcm.c index 158fc21a86c1..17ef05309469 100644 --- a/sound/soc/xilinx/xlnx_formatter_pcm.c +++ b/sound/soc/xilinx/xlnx_formatter_pcm.c @@ -713,7 +713,7 @@ MODULE_DEVICE_TABLE(of, xlnx_formatter_pcm_of_match); static struct platform_driver xlnx_formatter_pcm_driver = { .probe = xlnx_formatter_pcm_probe, - .remove_new = xlnx_formatter_pcm_remove, + .remove = xlnx_formatter_pcm_remove, .driver = { .name = DRV_NAME, .of_match_table = xlnx_formatter_pcm_of_match, diff --git a/sound/soc/xilinx/xlnx_spdif.c b/sound/soc/xilinx/xlnx_spdif.c index d52d5fc7b5b8..7febb3830dc2 100644 --- a/sound/soc/xilinx/xlnx_spdif.c +++ b/sound/soc/xilinx/xlnx_spdif.c @@ -325,7 +325,7 @@ static struct platform_driver xlnx_spdif_driver = { .of_match_table = xlnx_spdif_of_match, }, .probe = xlnx_spdif_probe, - .remove_new = xlnx_spdif_remove, + .remove = xlnx_spdif_remove, }; module_platform_driver(xlnx_spdif_driver); diff --git a/sound/soc/xtensa/xtfpga-i2s.c b/sound/soc/xtensa/xtfpga-i2s.c index 6e2b72d7a65d..4eaa9011405f 100644 --- a/sound/soc/xtensa/xtfpga-i2s.c +++ b/sound/soc/xtensa/xtfpga-i2s.c @@ -635,7 +635,7 @@ static const struct dev_pm_ops xtfpga_i2s_pm_ops = { static struct platform_driver xtfpga_i2s_driver = { .probe = xtfpga_i2s_probe, - .remove_new = xtfpga_i2s_remove, + .remove = xtfpga_i2s_remove, .driver = { .name = "xtfpga-i2s", .of_match_table = of_match_ptr(xtfpga_i2s_of_match), base-commit: 100cc857359b5d731407d1038f7e76cd0e871d94 -- 2.45.2

2 1

[PATCH] ASoC: tlv320aic31xx: Add support for loading filter coefficients
by Romain Gantois 09 Sep '24

09 Sep '24

The TLV320DAC3100 Audio DAC has 25 built-in digital audio processing blocks (PRBs). Each of these PRBs has a static filter structure with programmable coefficients. Once a PRB is selected for use by the DAC, its filter coefficients can be configured via a dedicated set of registers. Define a new optional firmware which can be loaded by the TLV320DAC driver. This firmware describes a full set of filter coefficients for all blocks used by the various PRBs. The firmware's binary format is heavily inspired by the one used in the peb2466 driver. It includes a version marker to allow for potential evolutions of the format. Note that adaptive filtering is not supported i.e. filter coefficients are loaded once before power-on and then cannot be changed while the DAC is powered. This is why only page A coefficients are modified. Page B coefficients are only used for adaptive filtering. Signed-off-by: Romain Gantois <romain.gantois(a)bootlin.com> --- sound/soc/codecs/tlv320aic31xx.c | 100 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 100 insertions(+) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 2f94cfda0e33..7e624c4b77b6 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -12,6 +12,7 @@ * and mono/stereo Class-D speaker driver. */ +#include <asm/unaligned.h> #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> @@ -22,6 +23,7 @@ #include <linux/gpio/consumer.h> #include <linux/regulator/consumer.h> #include <linux/acpi.h> +#include <linux/firmware.h> #include <linux/of.h> #include <linux/slab.h> #include <sound/core.h> @@ -1638,6 +1640,98 @@ static const struct i2c_device_id aic31xx_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); +static int tlv320dac3100_fw_load(struct aic31xx_priv *aic31xx, + const u8 *data, size_t size) +{ + int ret, reg; + u16 val16; + + /* + * Coefficients firmware binary structure. Multi-byte values are big-endian. + * + * @0, 16bits: Magic (0xB30C) + * @2, 16bits: Version (0x0100 for version 1.0) + * @4, 8bits: DAC Processing Block Selection + * @5, 62 16-bit values: Page 8 buffer A DAC programmable filter coefficients + * @129, 12 16-bit values: Page 9 Buffer A DAC programmable filter coefficients + * + * Filter coefficients are interpreted as two's complement values + * ranging from -32 768 to 32 767. For more details on filter coefficients, + * please refer to the TLV320DAC3100 datasheet, tables 6-120 and 6-123. + */ + + if (size != 153) { + dev_err(aic31xx->dev, "firmware size is %zu, expected 153 bytes\n", size); + return -EINVAL; + } + + /* Check magic */ + val16 = get_unaligned_be16(data); + if (val16 != 0xb30c) { + dev_err(aic31xx->dev, "fw magic is 0x%04x expected 0xb30c\n", val16); + return -EINVAL; + } + data += 2; + + /* Check version */ + val16 = get_unaligned_be16(data); + if (val16 != 0x0100) { + dev_err(aic31xx->dev, "invalid firmware version 0x%04x! expected 1", val16); + return -EINVAL; + } + data += 2; + + ret = regmap_write(aic31xx->regmap, AIC31XX_DACPRB, *data); + if (ret) { + dev_err(aic31xx->dev, "failed to write PRB index: err %d\n", ret); + return ret; + } + data += 1; + + /* Page 8 Buffer A coefficients */ + for (reg = 2; reg < 126; reg++) { + ret = regmap_write(aic31xx->regmap, AIC31XX_REG(8, reg), *data); + if (ret) { + dev_err(aic31xx->dev, + "failed to write page 8 filter coefficient %d: err %d\n", reg, ret); + return ret; + } + data++; + } + + /* Page 9 Buffer A coefficients */ + for (reg = 2; reg < 26; reg++) { + ret = regmap_write(aic31xx->regmap, AIC31XX_REG(9, reg), *data); + if (ret) { + dev_err(aic31xx->dev, + "failed to write page 9 filter coefficient %d: err %d\n", reg, ret); + return ret; + } + data++; + } + + dev_info(aic31xx->dev, "done loading DAC filter coefficients\n"); + + return ret; +} + +static int tlv320dac3100_load_coeffs(struct aic31xx_priv *aic31xx, + const char *fw_name) +{ + const struct firmware *fw; + int ret; + + ret = request_firmware(&fw, fw_name, aic31xx->dev); + if (ret) + return ret; + + ret = tlv320dac3100_fw_load(aic31xx, fw->data, fw->size); + + release_firmware(fw); + + return ret; +} + static int aic31xx_i2c_probe(struct i2c_client *i2c) { struct aic31xx_priv *aic31xx; @@ -1727,6 +1821,12 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c) } } + if (aic31xx->codec_type == DAC3100) { + ret = tlv320dac3100_load_coeffs(aic31xx, "tlv320dac3100-coeffs.bin"); + if (ret) + dev_warn(aic31xx->dev, "Did not load any filter coefficients\n"); + } + if (aic31xx->codec_type & DAC31XX_BIT) return devm_snd_soc_register_component(&i2c->dev, &soc_codec_driver_aic31xx, --- base-commit: 8400291e289ee6b2bf9779ff1c83a291501f017b change-id: 20240806-tlv320-filter-189edd66496c Best regards, -- Romain Gantois <romain.gantois(a)bootlin.com>

2 1

[PATCH] soundwire: stream: Revert "soundwire: stream: fix programming slave ports for non-continous port maps"
by Krzysztof Kozlowski 09 Sep '24

09 Sep '24

This reverts commit ab8d66d132bc8f1992d3eb6cab8d32dda6733c84 because it breaks codecs using non-continuous masks in source and sink ports. The commit missed the point that port numbers are not used as indices for iterating over prop.sink_ports or prop.source_ports. Soundwire core and existing codecs expect that the array passed as prop.sink_ports and prop.source_ports is continuous. The port mask still might be non-continuous, but that's unrelated. Reported-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Closes: https://lore.kernel.org/all/b6c75eee-761d-44c8-8413-2a5b34ee2f98@linux.inte… Fixes: ab8d66d132bc ("soundwire: stream: fix programming slave ports for non-continous port maps") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- I will need to fix my codec drivers, but that's independent. --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index f275143d7b18..7aa4900dcf31 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - unsigned long mask; + u8 num_ports; int i; if (direction == SDW_DATA_DIR_TX) { - mask = slave->prop.source_ports; + num_ports = hweight32(slave->prop.source_ports); dpn_prop = slave->prop.src_dpn_prop; } else { - mask = slave->prop.sink_ports; + num_ports = hweight32(slave->prop.sink_ports); dpn_prop = slave->prop.sink_dpn_prop; } - for_each_set_bit(i, &mask, 32) { + for (i = 0; i < num_ports; i++) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

5 6

[PATCH] ASoC: tas2781: fix to save the dsp bin file name into the correct array in case name_prefix is not NULL
by Shenghao Ding 09 Sep '24

09 Sep '24

fix to save the dsp bin file name into the correct array, coef_binaryname, instead of rca_binaryname, in case name_prefix is not NULL. Fixes: 55f0a1fa6ea4 ("ASoC: tas2781: Add name_prefix as the prefix name of DSP firmwares and calibrated data files") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 59fcce7fef7b..3bc10d615fd3 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -498,7 +498,7 @@ static void tasdevice_fw_ready(const struct firmware *fmw, */ tas_priv->fw_state = TASDEVICE_RCA_FW_OK; if (tas_priv->name_prefix) - scnprintf(tas_priv->rca_binaryname, 64, "%s-%s_coef.bin", + scnprintf(tas_priv->coef_binaryname, 64, "%s-%s_coef.bin", tas_priv->name_prefix, tas_priv->dev_name); else scnprintf(tas_priv->coef_binaryname, 64, "%s_coef.bin", -- 2.34.1

3 3

[PATCH] soundwire: stream: fix programming slave ports for non-continous port maps
by Krzysztof Kozlowski 09 Sep '24

09 Sep '24

Two bitmasks in 'struct sdw_slave_prop' - 'source_ports' and 'sink_ports' - define which ports to program in sdw_program_slave_port_params(). The masks are used to get the appropriate data port properties ('struct sdw_get_slave_dpn_prop') from an array. Bitmasks can be non-continuous or can start from index different than 0, thus when looking for matching port property for given port, we must iterate over mask bits, not from 0 up to number of ports. This fixes allocation and programming slave ports, when a source or sink masks start from further index. Fixes: f8101c74aa54 ("soundwire: Add Master and Slave port programming") Cc: <stable(a)vger.kernel.org> Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- drivers/soundwire/stream.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 7aa4900dcf31..f275143d7b18 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1291,18 +1291,18 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, unsigned int port_num) { struct sdw_dpn_prop *dpn_prop; - u8 num_ports; + unsigned long mask; int i; if (direction == SDW_DATA_DIR_TX) { - num_ports = hweight32(slave->prop.source_ports); + mask = slave->prop.source_ports; dpn_prop = slave->prop.src_dpn_prop; } else { - num_ports = hweight32(slave->prop.sink_ports); + mask = slave->prop.sink_ports; dpn_prop = slave->prop.sink_dpn_prop; } - for (i = 0; i < num_ports; i++) { + for_each_set_bit(i, &mask, 32) { if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.43.0

6 15

[PATCH v2] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property
by Kuninori Morimoto 09 Sep '24

09 Sep '24

At least if rsnd is using DPCM connection with Audio-Graph-Card2, fw_devlink might doesn't have enough information to break the cycle (Same problem might occur with Multi-CPU/Codec or Codec2Codec). In such case, rsnd driver will not be probed. Add post-init-providers support to break the link cycle. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - remove basic description/type - add detail description - update git-log Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9def..458877ce4aa42 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,13 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: | + At least if rsnd is using DPCM connection with Audio-Graph-Card2, fw_devlink might doesn't + have enough information to break the cycle. rsnd driver will not be probed in such case. + Add post-init-providers property to indicate which link in the cycle to break. + Same problem might occur with Multi-CPU/Codec or Codec2Codec. + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base -- 2.43.0

2 1

How to handle complex Codec2Codec
by Kuninori Morimoto 09 Sep '24

09 Sep '24

Hi ALSA ML ALSA SoC is supporting Codec2Codec connection, and Audio Graph Card2 is supporting it. I think its image is... +------+ +--- |Codec1| <-- | +------+ | +------+ +--> |Codec2| --> +------+ This is OK, but how to handle below ? +----------------------+ |CPU | +------+ | +--[rate convert]<--| <-- |Codec1| <-- | | | +------+ | | | +------+ | +------------------>| --> |Codec2| --> +----------------------+ +------+ I think there are 2 issues. (1) How to handle [CPU] part on Codec2Codec ? (2) How to handle [rate convert] on Codec2Codec ? I guess (1) can be handled if [CPU] was worked as platform, but do you have any other idea/solution for it ? I think (2) [rate convert] needs DPCM connection, but how to use it with Codec2Codec ? Or, can we handle Codec2Codec via DPCM somehow ?? FE BE +------+ +------+ --> |Codec1| -> |Codec2| -> +------+ +------+ If we can do this, both (1) and (2) can be solved, like below ? In such case, [Codec1] need to work as [capture] when playback. But I'm not sure how to implement it... FE BE +------+ +-----------------+ +-----+ +------+ --> |Codec1|-|CPU[rate convert]| -> |dummy|-|Codec2| -> +------+ +-----------------+ +-----+ +------+ Thank you for your help !! Best regards --- Kuninori Morimoto

2 8

Re: [PATCH] ALSA: Add logitech Audio profile quirk
by Joshua Pius 06 Sep '24

06 Sep '24

1 3

[PATCH] ASoC: codecs: Use devm_clk_get_enabled() helpers
by ying zuxin 06 Sep '24

06 Sep '24

From: ying zuxin <11154159(a)vivo.com> The devm_clk_get_enabled() helpers: - call devm_clk_get() - call clk_prepare_enable() and register what is needed in order to call clk_disable_unprepare() when needed, as a managed resource. This simplifies the code and avoids the calls to clk_disable_unprepare(). Signed-off-by: ying zuxin <11154159(a)vivo.com> --- sound/soc/codecs/peb2466.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/soc/codecs/peb2466.c b/sound/soc/codecs/peb2466.c index 76ee7e3f4d9b..b67cfad4fc32 100644 --- a/sound/soc/codecs/peb2466.c +++ b/sound/soc/codecs/peb2466.c @@ -1975,12 +1975,9 @@ static int peb2466_spi_probe(struct spi_device *spi) if (IS_ERR(peb2466->reset_gpio)) return PTR_ERR(peb2466->reset_gpio); - peb2466->mclk = devm_clk_get(&peb2466->spi->dev, "mclk"); + peb2466->mclk = devm_clk_get_enabled(&peb2466->spi->dev, "mclk"); if (IS_ERR(peb2466->mclk)) return PTR_ERR(peb2466->mclk); - ret = clk_prepare_enable(peb2466->mclk); - if (ret) - return ret; if (peb2466->reset_gpio) { gpiod_set_value_cansleep(peb2466->reset_gpio, 1); @@ -2031,17 +2028,9 @@ static int peb2466_spi_probe(struct spi_device *spi) return 0; failed: - clk_disable_unprepare(peb2466->mclk); return ret; } -static void peb2466_spi_remove(struct spi_device *spi) -{ - struct peb2466 *peb2466 = spi_get_drvdata(spi); - - clk_disable_unprepare(peb2466->mclk); -} - static const struct of_device_id peb2466_of_match[] = { { .compatible = "infineon,peb2466", }, { } @@ -2061,7 +2050,6 @@ static struct spi_driver peb2466_spi_driver = { }, .id_table = peb2466_id_table, .probe = peb2466_spi_probe, - .remove = peb2466_spi_remove, }; module_spi_driver(peb2466_spi_driver); -- 2.39.0

3 2

[PATCH] ASoC: makes rtd->initialized bit field
by Kuninori Morimoto 06 Sep '24

06 Sep '24

rtd->initialized is used to know whether soc_init_pcm_runtime() was correctly fined, and used to call snd_soc_link_exit(). We don't need to have it as bool, let's make it bit-field same as other flags. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Cc: Cezary Rojewski <cezary.rojewski(a)intel.com> --- include/sound/soc.h | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 81ef380b2e06..e6e359c1a2ac 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1207,8 +1207,7 @@ struct snd_soc_pcm_runtime { /* bit field */ unsigned int pop_wait:1; unsigned int fe_compr:1; /* for Dynamic PCM */ - - bool initialized; + unsigned int initialized:1; /* CPU/Codec/Platform */ int num_components; -- 2.43.0

2 1

[PATCH 1/2] pcm: dmix: Fix resume with multiple instances
by Takashi Iwai 06 Sep '24

06 Sep '24

The fix for PCM dmix suspend/resume checks spcm->info bit of SND_PCM_INFO_RESUME for applying a workaround for drivers with the full resume support. This assumed that scpm->info is exposed from the underlying slave PCM device. The above is true for the first opened instance, but for the second opened instance, it's a copy from the saved data in shmem. And, we dropped SND_PCM_INFO_RESUME bit there to assure not to expose the full resume capability to applications. This resulted in the inconsistencies, and when the second instance is resumed at first, it misses the snd_pcm_resume() call, hence the driver doesn't react properly any longer. For addressing it, we keep SND_PCM_INFO_RESUME bit in shmptr->s.info bits as is, while dropping the bit exposed to apps in snd_pcm_direct_hw_refine() and *_hw_params() callbacks. Fixes: 6d1d620eadf3 ("pcm: dmix: resume workaround for buggy driver") Reported-and-tested-by: Chancel Liu <chancel.liu(a)nxp.com> Closes: https://lore.kernel.org/DB9PR04MB94988752ED7C43B399E0BC00E3942@DB9PR04MB949… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index e53e59238119..17e677f60b19 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -1018,6 +1018,7 @@ int snd_pcm_direct_hw_refine(snd_pcm_t *pcm, snd_pcm_hw_params_t *params) } dshare->timer_ticks = hw_param_interval(params, SND_PCM_HW_PARAM_PERIOD_SIZE)->max / dshare->slave_period_size; params->info = dshare->shmptr->s.info; + params->info &= ~SND_PCM_INFO_RESUME; #ifdef REFINE_DEBUG snd_output_puts(log, "DMIX REFINE (end):\n"); snd_pcm_hw_params_dump(params, log); @@ -1031,6 +1032,7 @@ int snd_pcm_direct_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) snd_pcm_direct_t *dmix = pcm->private_data; params->info = dmix->shmptr->s.info; + params->info &= ~SND_PCM_INFO_RESUME; params->rate_num = dmix->shmptr->s.rate; params->rate_den = 1; params->fifo_size = 0; @@ -1183,8 +1185,6 @@ static void save_slave_setting(snd_pcm_direct_t *dmix, snd_pcm_t *spcm) COPY_SLAVE(buffer_time); COPY_SLAVE(sample_bits); COPY_SLAVE(frame_bits); - - dmix->shmptr->s.info &= ~SND_PCM_INFO_RESUME; } #undef COPY_SLAVE -- 2.43.0

1 1

Suspend/resume Issue on pcm_dmix.c in alsa-lib
by Chancel Liu 06 Sep '24

06 Sep '24

Hi Takashi Iwai, Jaroslav Kysela We found an issue on dmix in alsa-lib when do suspend and resume. It can be easily reproduced by following steps: 1. Run two dmix clients in parallel. (Only one client doesn't has such issue) ~# aplay xxx1.wav & ~# aplay xxx2.wav & Here I attach the asound.conf we're using. ~# cat /etc/asound.conf defaults.pcm.rate_converter "linear" pcm.dmix_44100{ type dmix ipc_key 5678293 ipc_key_add_uid yes slave{ pcm "hw:0,0" period_time 40000 format S16_LE rate 44100 } } pcm.asymed{ type asym playback.pcm "dmix_44100" capture.pcm "dsnoop_44100" } pcm.!default{ type plug route_policy "average" slave.pcm "asymed" } 2. Let linux enter into suspend and then resume(Repeat this step if not reproduced) 3. After resume, aplay will get stuck in snd_pcm_wait(). The GDB shows: (gdb) bt #0 0x0000fffff7da9264 in __GI___poll (fds=fds@entry=0xfffffffff480, nfds=nfds@entry=1, timeout=timeout@entry=240) at /usr/src/debug/glibc/2.39+git/sysdeps/unix/sysv/linux/poll.c:41 #1 0x0000fffff7edf468 in poll (__timeout=240, __nfds=1, __fds=0xfffffffff480) #2 snd1_pcm_wait_nocheck (pcm=pcm@entry=0xaaaaaaad2cb0, timeout=240, timeout@entry=-10001) at pcm.c:2993 #3 0x0000fffff7ee54a0 in snd1_pcm_write_areas (pcm=pcm@entry=0xaaaaaaad2cb0, areas=areas@entry=0xfffffffff560, offset=<optimized out>, offset@entry=0, size=<optimized out>, size@entry=1768, func=func@entry=0xfffff7ef5190 <snd_pcm_plugin_write_areas>) at pcm.c:7699 #4 0x0000fffff7ef5020 in snd_pcm_plugin_writei (pcm=0xaaaaaaad2cb0, buffer=<optimized out>, size=1768) at pcm_plugin.c:354 It seems that sometimes after suspend and resume there's no available space for data written into buffer. Then aplay keeps stuck in snd_pcm_wait(). I checked the hw_ptr of dmix and found that hw_ptr is always 0 after resume. I don't have a solution now so I turn to you for help. The version of alsa-lib is v1.2.11. Could you please help check it? Regards, Chancel Liu

3 12

[GIT PULL] ASoC fixes for v6.11-rc6
by Mark Brown 06 Sep '24

06 Sep '24

The following changes since commit bb4485562f5907708f1c218b5d70dce04165d1e1: ASoC: cs-amp-lib: Ignore empty UEFI calibration entries (2024-08-22 17:52:26 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc6 for you to fetch changes up to 77212f300bfd6fb3edaabd1daf863cabb521854a: ASoC: codecs: lpass-va-macro: set the default codec version for sm8250 (2024-08-30 13:54:24 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A larger set of fixes than I'd like at this point, but mainly due to people working on fixing module autoloading by adding missing exports of ID tables rather than anything particularly concerning. There are some other runtime fixes and quirks, and a tweak to the ABI definition for SOF which ensures that a struct layout doesn't vary depending on the architecture of the host. ---------------------------------------------------------------- Chen-Yu Tsai (1): ASoc: SOF: topology: Clear SOF link platform name upon unload Hans de Goede (2): ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards harder ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict Laurentiu Mihalcea (1): ASoC: SOF: ipc: replace "enum sof_comp_type" field with "uint32_t" Liao Chen (4): ASoC: intel: fix module autoloading ASoC: google: fix module autoloading ASoC: tda7419: fix module autoloading ASoC: fix module autoloading Mark Brown (1): ASoC: fix module autoloading Markuss Broks (1): ASoC: amd: yc: Add a quirk for MSI Bravo 17 (D7VEK) Matteo Martelli (1): ASoC: sunxi: sun4i-i2s: fix LRCLK polarity in i2s mode Mohan Kumar (1): ASoC: tegra: Fix CBB error during probe() Srinivas Kandagatla (1): ASoC: codecs: lpass-va-macro: set the default codec version for sm8250 Zhang Yi (1): ASoC: mediatek: mt8188-mt6359: Modify key robelin (1): ASoC: dapm: Fix UAF for snd_soc_pcm_runtime object include/sound/sof/topology.h | 2 +- include/uapi/sound/sof/abi.h | 2 +- sound/soc/amd/yc/acp6x-mach.c | 7 ++ sound/soc/codecs/chv3-codec.c | 1 + sound/soc/codecs/lpass-va-macro.c | 11 +- sound/soc/codecs/tda7419.c | 1 + sound/soc/google/chv3-i2s.c | 1 + sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - sound/soc/intel/keembay/kmb_platform.c | 1 + sound/soc/mediatek/mt8188/mt8188-mt6359.c | 17 ++- sound/soc/soc-dapm.c | 1 + sound/soc/sof/topology.c | 2 + sound/soc/sunxi/sun4i-i2s.c | 143 +++++++++++----------- sound/soc/tegra/tegra210_ahub.c | 10 +- 23 files changed, 127 insertions(+), 91 deletions(-)

2 1

[PATCH] ASoC: Intel: skl_hda_dsp_generic: convert comma to semicolon
by Chen Ni 05 Sep '24

05 Sep '24

Replace comma between expressions with semicolons. Using a ',' in place of a ';' can have unintended side effects. Although that is not the case here, it is seems best to use ';' unless ',' is intended. Found by inspection. No functional change intended. Compile tested only. Signed-off-by: Chen Ni <nichen(a)iscas.ac.cn> --- sound/soc/intel/boards/skl_hda_dsp_generic.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/skl_hda_dsp_generic.c b/sound/soc/intel/boards/skl_hda_dsp_generic.c index 225867bb3310..8e104874d58c 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_generic.c +++ b/sound/soc/intel/boards/skl_hda_dsp_generic.c @@ -217,14 +217,14 @@ static int skl_hda_audio_probe(struct platform_device *pdev) return -ENOMEM; card = &ctx->card; - card->name = "hda-dsp", - card->owner = THIS_MODULE, - card->dai_link = skl_hda_be_dai_links, - card->dapm_widgets = skl_hda_widgets, - card->dapm_routes = skl_hda_map, - card->add_dai_link = skl_hda_add_dai_link, - card->fully_routed = true, - card->late_probe = skl_hda_card_late_probe, + card->name = "hda-dsp"; + card->owner = THIS_MODULE; + card->dai_link = skl_hda_be_dai_links; + card->dapm_widgets = skl_hda_widgets; + card->dapm_routes = skl_hda_map; + card->add_dai_link = skl_hda_add_dai_link; + card->fully_routed = true; + card->late_probe = skl_hda_card_late_probe; snd_soc_card_set_drvdata(card, ctx); -- 2.25.1

2 1

[PATCH 1/2 4.19.y] ALSA: usb-audio: Sanity checks for each pipe and EP types
by Takashi Iwai 05 Sep '24

05 Sep '24

[ Upstream commit 801ebf1043ae7b182588554cc9b9ad3c14bc2ab5 ] The recent USB core code performs sanity checks for the given pipe and EP types, and it can be hit by manipulated USB descriptors by syzbot. For making syzbot happier, this patch introduces a local helper for a sanity check in the driver side and calls it at each place before the message handling, so that we can avoid the WARNING splats. Reported-by: syzbot+d952e5e28f5fb7718d23(a)syzkaller.appspotmail.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/usb/helper.c | 17 +++++++++++++++++ sound/usb/helper.h | 1 + sound/usb/quirks.c | 14 +++++++++++--- 3 files changed, 29 insertions(+), 3 deletions(-) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index 7712e2b84183..b1cc9499c57e 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -76,6 +76,20 @@ void *snd_usb_find_csint_desc(void *buffer, int buflen, void *after, u8 dsubtype return NULL; } +/* check the validity of pipe and EP types */ +int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe) +{ + static const int pipetypes[4] = { + PIPE_CONTROL, PIPE_ISOCHRONOUS, PIPE_BULK, PIPE_INTERRUPT + }; + struct usb_host_endpoint *ep; + + ep = usb_pipe_endpoint(dev, pipe); + if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) + return -EINVAL; + return 0; +} + /* * Wrapper for usb_control_msg(). * Allocates a temp buffer to prevent dmaing from/to the stack. @@ -88,6 +102,9 @@ int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, void *buf = NULL; int timeout; + if (snd_usb_pipe_sanity_check(dev, pipe)) + return -EINVAL; + if (size > 0) { buf = kmemdup(data, size, GFP_KERNEL); if (!buf) diff --git a/sound/usb/helper.h b/sound/usb/helper.h index f5b4c6647e4d..5e8a18b4e7b9 100644 --- a/sound/usb/helper.h +++ b/sound/usb/helper.h @@ -7,6 +7,7 @@ unsigned int snd_usb_combine_bytes(unsigned char *bytes, int size); void *snd_usb_find_desc(void *descstart, int desclen, void *after, u8 dtype); void *snd_usb_find_csint_desc(void *descstart, int desclen, void *after, u8 dsubtype); +int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe); int snd_usb_ctl_msg(struct usb_device *dev, unsigned int pipe, __u8 request, __u8 requesttype, __u16 value, __u16 index, void *data, __u16 size); diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 43cbaaff163f..087eef5e249d 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -743,11 +743,13 @@ static int snd_usb_novation_boot_quirk(struct usb_device *dev) static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) { int err, actual_length; - /* "midi send" enable */ static const u8 seq[] = { 0x4e, 0x73, 0x52, 0x01 }; + void *buf; - void *buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); + if (snd_usb_pipe_sanity_check(dev, usb_sndintpipe(dev, 0x05))) + return -EINVAL; + buf = kmemdup(seq, ARRAY_SIZE(seq), GFP_KERNEL); if (!buf) return -ENOMEM; err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x05), buf, @@ -772,7 +774,11 @@ static int snd_usb_accessmusic_boot_quirk(struct usb_device *dev) static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) { - int ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), + int ret; + + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; + ret = usb_control_msg(dev, usb_sndctrlpipe(dev, 0), 0xaf, USB_TYPE_VENDOR | USB_RECIP_DEVICE, 1, 0, NULL, 0, 1000); @@ -879,6 +885,8 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) dev_dbg(&dev->dev, "Waiting for Axe-Fx III to boot up...\n"); + if (snd_usb_pipe_sanity_check(dev, usb_sndctrlpipe(dev, 0))) + return -EINVAL; /* If the Axe-Fx III has not fully booted, it will timeout when trying * to enable the audio streaming interface. A more generous timeout is * used here to detect when the Axe-Fx III has finished booting as the -- 2.45.2

4 5

[PATCH] ASoC: atmel: mchp-i2s-mcc: Improve maxburst calculation for better performance
by Andrei Simion 05 Sep '24

05 Sep '24

From: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> The period size represents the size of the DMA descriptor. To ensure all DMA descriptors start from a well-aligned address, the period size must be divided by (sample size * maxburst), not just by maxburst. This adjustment allows for computing a higher maxburst value, thereby increasing the performance of the DMA transfer. Previously, snd_pcm_lib_period_bytes() returned 0 because the runtime HW parameters are computed after the hw_params() callbacks are used. To address this, we now use params_*() functions to compute the period size accurately. This change optimizes the DMA transfer performance by ensuring proper alignment and efficient use of maxburst values. Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> [andrei.simion(a)microchip.com: Reword commit message and commit title. Add macros with values for maximum DMA chunk size allowed. Add DMA_BURST_ALIGNED preprocessor function to check the alignment of the DMA burst] Signed-off-by: Andrei Simion <andrei.simion(a)microchip.com> --- sound/soc/atmel/mchp-i2s-mcc.c | 38 +++++++++++++++++++++++++++++----- 1 file changed, 33 insertions(+), 5 deletions(-) diff --git a/sound/soc/atmel/mchp-i2s-mcc.c b/sound/soc/atmel/mchp-i2s-mcc.c index 193dd7acceb0..35a56b266b8d 100644 --- a/sound/soc/atmel/mchp-i2s-mcc.c +++ b/sound/soc/atmel/mchp-i2s-mcc.c @@ -221,6 +221,15 @@ #define MCHP_I2SMCC_MAX_CHANNELS 8 #define MCHP_I2MCC_TDM_SLOT_WIDTH 32 +/* + * ---- DMA chunk size allowed ---- + */ +#define MCHP_I2SMCC_DMA_8_WORD_CHUNK 8 +#define MCHP_I2SMCC_DMA_4_WORD_CHUNK 4 +#define MCHP_I2SMCC_DMA_2_WORD_CHUNK 2 +#define MCHP_I2SMCC_DMA_1_WORD_CHUNK 1 +#define DMA_BURST_ALIGNED(_p, _s, _w) !(_p % (_s * _w)) + static const struct regmap_config mchp_i2s_mcc_regmap_config = { .reg_bits = 32, .reg_stride = 4, @@ -504,12 +513,30 @@ static int mchp_i2s_mcc_is_running(struct mchp_i2s_mcc_dev *dev) return !!(sr & (MCHP_I2SMCC_SR_TXEN | MCHP_I2SMCC_SR_RXEN)); } +static inline int mchp_i2s_mcc_period_to_maxburst(int period_size, int sample_size) +{ + int p_size = period_size; + int s_size = sample_size; + + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_8_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_8_WORD_CHUNK; + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_4_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_4_WORD_CHUNK; + if (DMA_BURST_ALIGNED(p_size, s_size, MCHP_I2SMCC_DMA_2_WORD_CHUNK)) + return MCHP_I2SMCC_DMA_2_WORD_CHUNK; + return MCHP_I2SMCC_DMA_1_WORD_CHUNK; +} + static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { unsigned long rate = 0; struct mchp_i2s_mcc_dev *dev = snd_soc_dai_get_drvdata(dai); + int sample_bytes = params_physical_width(params) / 8; + int period_bytes = params_period_size(params) * + params_channels(params) * sample_bytes; + int maxburst; u32 mra = 0; u32 mrb = 0; unsigned int channels = params_channels(params); @@ -519,9 +546,9 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, int ret; bool is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); - dev_dbg(dev->dev, "%s() rate=%u format=%#x width=%u channels=%u\n", + dev_dbg(dev->dev, "%s() rate=%u format=%#x width=%u channels=%u period_bytes=%d\n", __func__, params_rate(params), params_format(params), - params_width(params), params_channels(params)); + params_width(params), params_channels(params), period_bytes); switch (dev->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: @@ -630,11 +657,12 @@ static int mchp_i2s_mcc_hw_params(struct snd_pcm_substream *substream, * We must have the same burst size configured * in the DMA transfer and in out IP */ - mrb |= MCHP_I2SMCC_MRB_DMACHUNK(channels); + maxburst = mchp_i2s_mcc_period_to_maxburst(period_bytes, sample_bytes); + mrb |= MCHP_I2SMCC_MRB_DMACHUNK(maxburst); if (is_playback) - dev->playback.maxburst = 1 << (fls(channels) - 1); + dev->playback.maxburst = maxburst; else - dev->capture.maxburst = 1 << (fls(channels) - 1); + dev->capture.maxburst = maxburst; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: base-commit: fdadd93817f124fd0ea6ef251d4a1068b7feceba -- 2.34.1

2 1

[PATCH 2/2 4.19.y] ALSA: usb-audio: Fix gpf in snd_usb_pipe_sanity_check
by Hillf Danton 05 Sep '24

05 Sep '24

[ Upstream commit 5d78e1c2b7f4be00bbe62141603a631dc7812f35 ] syzbot found the following crash on: general protection fault: 0000 [#1] SMP KASAN RIP: 0010:snd_usb_pipe_sanity_check+0x80/0x130 sound/usb/helper.c:75 Call Trace: snd_usb_motu_microbookii_communicate.constprop.0+0xa0/0x2fb sound/usb/quirks.c:1007 snd_usb_motu_microbookii_boot_quirk sound/usb/quirks.c:1051 [inline] snd_usb_apply_boot_quirk.cold+0x163/0x370 sound/usb/quirks.c:1280 usb_audio_probe+0x2ec/0x2010 sound/usb/card.c:576 usb_probe_interface+0x305/0x7a0 drivers/usb/core/driver.c:361 really_probe+0x281/0x650 drivers/base/dd.c:548 .... It was introduced in commit 801ebf1043ae for checking pipe and endpoint types. It is fixed by adding a check of the ep pointer in question. BugLink: https://syzkaller.appspot.com/bug?extid=d59c4387bfb6eced94e2 Reported-by: syzbot <syzbot+d59c4387bfb6eced94e2(a)syzkaller.appspotmail.com> Fixes: 801ebf1043ae ("ALSA: usb-audio: Sanity checks for each pipe and EP types") Cc: Andrey Konovalov <andreyknvl(a)google.com> Signed-off-by: Hillf Danton <hdanton(a)sina.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/usb/helper.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/helper.c b/sound/usb/helper.c index b1cc9499c57e..029489b490ca 100644 --- a/sound/usb/helper.c +++ b/sound/usb/helper.c @@ -85,7 +85,7 @@ int snd_usb_pipe_sanity_check(struct usb_device *dev, unsigned int pipe) struct usb_host_endpoint *ep; ep = usb_pipe_endpoint(dev, pipe); - if (usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) + if (!ep || usb_pipetype(pipe) != pipetypes[usb_endpoint_type(&ep->desc)]) return -EINVAL; return 0; } -- 2.45.2

2 1

[PATCH 0/2 4.19.y] ALSA: usb-audio: sanity checks for pipes
by Dan Carpenter 05 Sep '24

05 Sep '24

Backporting the sanity checks on pipes seems like a good idea. These are basically the same as Takashi Iwai's patches upstream. The difference is that upstream added two sanity checks to code that doesn't exist in 4.19. I had talked about backporting fcc2cc1f3561 ("USB: move snd_usb_pipe_sanity_check into the USB core") but that's just a refactor and not a bug fix. Hillf Danton (1): ALSA: usb-audio: Fix gpf in snd_usb_pipe_sanity_check Takashi Iwai (1): ALSA: usb-audio: Sanity checks for each pipe and EP types sound/usb/helper.c | 17 +++++++++++++++++ sound/usb/helper.h | 1 + sound/usb/quirks.c | 14 +++++++++++--- 3 files changed, 29 insertions(+), 3 deletions(-) -- 2.45.2

1 0

[PATCH] ASoC/HDA cs35l56: Remove redundant IRQ handling
by Simon Trimmer 05 Sep '24

05 Sep '24

The IRQ handling in the cs35l56 driver was purely informational. It was not necessary to support the HDA or ASoC driver functionality and added unnecessary complexity to the drivers. As the IRQ signal GPIO line could be connected and shared with other components the handling is replaced with a regmap patch to ensure the cs35l56 IRQ sources are masked and will not generate interrupts that go unhandled. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- v1 - As this changes both the ASoC and HDA cs35l56 device drivers we considered a series of patches that stepped through the removal of the common code but thought that the co-ordination would be unnecessarily complicated so we initially offer a combined patch that that applies to both Mark and Takashi's for-next branches so that either route can be taken. include/sound/cs35l56.h | 4 -- sound/pci/hda/cs35l56_hda.c | 53 +-------------- sound/pci/hda/cs35l56_hda_i2c.c | 9 +-- sound/pci/hda/cs35l56_hda_spi.c | 9 +-- sound/soc/codecs/cs35l56-i2c.c | 2 - sound/soc/codecs/cs35l56-sdw.c | 104 +----------------------------- sound/soc/codecs/cs35l56-shared.c | 101 +---------------------------- sound/soc/codecs/cs35l56-spi.c | 2 - sound/soc/codecs/cs35l56.c | 92 ++------------------------ sound/soc/codecs/cs35l56.h | 10 --- 10 files changed, 14 insertions(+), 372 deletions(-) diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index 94e8185c4795..acc43d85f5df 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -257,8 +257,6 @@ struct cs35l56_base { struct device *dev; struct regmap *regmap; - int irq; - struct mutex irq_lock; u8 type; u8 rev; bool init_done; @@ -298,8 +296,6 @@ int cs35l56_wait_for_firmware_boot(struct cs35l56_base *cs35l56_base); void cs35l56_wait_control_port_ready(void); void cs35l56_wait_min_reset_pulse(void); void cs35l56_system_reset(struct cs35l56_base *cs35l56_base, bool is_soundwire); -int cs35l56_irq_request(struct cs35l56_base *cs35l56_base, int irq); -irqreturn_t cs35l56_irq(int irq, void *data); int cs35l56_is_fw_reload_needed(struct cs35l56_base *cs35l56_base); int cs35l56_runtime_suspend_common(struct cs35l56_base *cs35l56_base); int cs35l56_runtime_resume_common(struct cs35l56_base *cs35l56_base, bool is_soundwire); diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 31cc92bac89a..8ec06a3f4765 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -626,8 +626,6 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) goto err_fw_release; } - mutex_lock(&cs35l56->base.irq_lock); - /* * If the firmware hasn't been patched it must be shutdown before * doing a full patch and reset afterwards. If it is already @@ -637,14 +635,14 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) if (firmware_missing && (wmfw_firmware || coeff_firmware)) { ret = cs35l56_firmware_shutdown(&cs35l56->base); if (ret) - goto err; + goto err_fw_release; } ret = cs_dsp_power_up(&cs35l56->cs_dsp, wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename, "misc"); if (ret) { dev_dbg(cs35l56->base.dev, "%s: cs_dsp_power_up ret %d\n", __func__, ret); - goto err; + goto err_fw_release; } if (wmfw_filename) @@ -687,8 +685,6 @@ static void cs35l56_hda_fw_load(struct cs35l56_hda *cs35l56) err_powered_up: if (!cs35l56->base.fw_patched) cs_dsp_power_down(&cs35l56->cs_dsp); -err: - mutex_unlock(&cs35l56->base.irq_lock); err_fw_release: cs35l56_hda_release_firmware_files(wmfw_firmware, wmfw_filename, coeff_firmware, coeff_filename); @@ -778,15 +774,6 @@ static int cs35l56_hda_system_suspend(struct device *dev) cs35l56->suspended = true; - /* - * The interrupt line is normally shared, but after we start suspending - * we can't check if our device is the source of an interrupt, and can't - * clear it. Prevent this race by temporarily disabling the parent irq - * until we reach _no_irq. - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - return pm_runtime_force_suspend(dev); } @@ -806,34 +793,6 @@ static int cs35l56_hda_system_suspend_late(struct device *dev) return 0; } -static int cs35l56_hda_system_suspend_no_irq(struct device *dev) -{ - struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); - - /* Handlers are now disabled so the parent IRQ can safely be re-enabled. */ - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - - return 0; -} - -static int cs35l56_hda_system_resume_no_irq(struct device *dev) -{ - struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); - - /* - * WAKE interrupts unmask if the CS35L56 hibernates, which can cause - * spurious interrupts, and the interrupt line is normally shared. - * We can't check if our device is the source of an interrupt, and can't - * clear it, until it has fully resumed. Prevent this race by temporarily - * disabling the parent irq until we complete resume(). - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - - return 0; -} - static int cs35l56_hda_system_resume_early(struct device *dev) { struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); @@ -856,11 +815,8 @@ static int cs35l56_hda_system_resume(struct device *dev) struct cs35l56_hda *cs35l56 = dev_get_drvdata(dev); int ret; - /* Undo pm_runtime_force_suspend() before re-enabling the irq */ + /* Undo pm_runtime_force_suspend() */ ret = pm_runtime_force_resume(dev); - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - if (ret) return ret; @@ -987,7 +943,6 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int hid, int id) { int ret; - mutex_init(&cs35l56->base.irq_lock); dev_set_drvdata(cs35l56->base.dev, cs35l56); INIT_WORK(&cs35l56->dsp_work, cs35l56_hda_dsp_work); @@ -1120,8 +1075,6 @@ const struct dev_pm_ops cs35l56_hda_pm_ops = { SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend, cs35l56_hda_system_resume) LATE_SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend_late, cs35l56_hda_system_resume_early) - NOIRQ_SYSTEM_SLEEP_PM_OPS(cs35l56_hda_system_suspend_no_irq, - cs35l56_hda_system_resume_no_irq) }; EXPORT_SYMBOL_NS_GPL(cs35l56_hda_pm_ops, SND_HDA_SCODEC_CS35L56); diff --git a/sound/pci/hda/cs35l56_hda_i2c.c b/sound/pci/hda/cs35l56_hda_i2c.c index 40f2f97944d5..822c1251f703 100644 --- a/sound/pci/hda/cs35l56_hda_i2c.c +++ b/sound/pci/hda/cs35l56_hda_i2c.c @@ -34,14 +34,7 @@ static int cs35l56_hda_i2c_probe(struct i2c_client *clt) return ret; } - ret = cs35l56_hda_common_probe(cs35l56, id->driver_data, clt->addr); - if (ret) - return ret; - ret = cs35l56_irq_request(&cs35l56->base, clt->irq); - if (ret < 0) - cs35l56_hda_remove(cs35l56->base.dev); - - return ret; + return cs35l56_hda_common_probe(cs35l56, id->driver_data, clt->addr); } static void cs35l56_hda_i2c_remove(struct i2c_client *clt) diff --git a/sound/pci/hda/cs35l56_hda_spi.c b/sound/pci/hda/cs35l56_hda_spi.c index 7f02155fe61e..37a159546e59 100644 --- a/sound/pci/hda/cs35l56_hda_spi.c +++ b/sound/pci/hda/cs35l56_hda_spi.c @@ -34,14 +34,7 @@ static int cs35l56_hda_spi_probe(struct spi_device *spi) return ret; } - ret = cs35l56_hda_common_probe(cs35l56, id->driver_data, spi_get_chipselect(spi, 0)); - if (ret) - return ret; - ret = cs35l56_irq_request(&cs35l56->base, spi->irq); - if (ret < 0) - cs35l56_hda_remove(cs35l56->base.dev); - - return ret; + return cs35l56_hda_common_probe(cs35l56, id->driver_data, spi_get_chipselect(spi, 0)); } static void cs35l56_hda_spi_remove(struct spi_device *spi) diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 2bd2ff75cd50..4419a94c5e73 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -41,8 +41,6 @@ static int cs35l56_i2c_probe(struct i2c_client *client) return ret; ret = cs35l56_init(cs35l56); - if (ret == 0) - ret = cs35l56_irq_request(&cs35l56->base, client->irq); if (ret < 0) cs35l56_remove(cs35l56); diff --git a/sound/soc/codecs/cs35l56-sdw.c b/sound/soc/codecs/cs35l56-sdw.c index 7c9a17fe2195..134631eff861 100644 --- a/sound/soc/codecs/cs35l56-sdw.c +++ b/sound/soc/codecs/cs35l56-sdw.c @@ -266,75 +266,14 @@ static void cs35l56_sdw_init(struct sdw_slave *peripheral) } ret = cs35l56_init(cs35l56); - if (ret < 0) { + if (ret < 0) regcache_cache_only(cs35l56->base.regmap, true); - goto out; - } - - /* - * cs35l56_init can return with !init_done if it triggered - * a soft reset. - */ - if (cs35l56->base.init_done) { - /* Enable SoundWire interrupts */ - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - } out: pm_runtime_mark_last_busy(cs35l56->base.dev); pm_runtime_put_autosuspend(cs35l56->base.dev); } -static int cs35l56_sdw_interrupt(struct sdw_slave *peripheral, - struct sdw_slave_intr_status *status) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); - - /* SoundWire core holds our pm_runtime when calling this function. */ - - dev_dbg(cs35l56->base.dev, "int control_port=%#x\n", status->control_port); - - if ((status->control_port & SDW_SCP_INT1_IMPL_DEF) == 0) - return 0; - - /* - * Prevent bus manager suspending and possibly issuing a - * bus-reset before the queued work has run. - */ - pm_runtime_get_noresume(cs35l56->base.dev); - - /* - * Mask and clear until it has been handled. The read of GEN_INT_STAT_1 - * is required as per the SoundWire spec for interrupt status bits - * to clear. GEN_INT_MASK_1 masks the _inputs_ to GEN_INT_STAT1. - * None of the interrupts are time-critical so use the - * power-efficient queue. - */ - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - queue_work(system_power_efficient_wq, &cs35l56->sdw_irq_work); - - return 0; -} - -static void cs35l56_sdw_irq_work(struct work_struct *work) -{ - struct cs35l56_private *cs35l56 = container_of(work, - struct cs35l56_private, - sdw_irq_work); - - cs35l56_irq(-1, &cs35l56->base); - - /* unmask interrupts */ - if (!cs35l56->sdw_irq_no_unmask) - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - - pm_runtime_put_autosuspend(cs35l56->base.dev); -} - static int cs35l56_sdw_read_prop(struct sdw_slave *peripheral) { struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); @@ -349,7 +288,7 @@ static int cs35l56_sdw_read_prop(struct sdw_slave *peripheral) prop->sink_ports = BIT(CS35L56_SDW1_PLAYBACK_PORT); prop->paging_support = true; prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; - prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY | SDW_SCP_INT1_IMPL_DEF; + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; /* DP1 - playback */ ports[0].num = CS35L56_SDW1_PLAYBACK_PORT; @@ -406,7 +345,6 @@ static int __maybe_unused cs35l56_sdw_clk_stop(struct sdw_slave *peripheral, static const struct sdw_slave_ops cs35l56_sdw_ops = { .read_prop = cs35l56_sdw_read_prop, - .interrupt_callback = cs35l56_sdw_interrupt, .update_status = cs35l56_sdw_update_status, #ifdef DEBUG .clk_stop = cs35l56_sdw_clk_stop, @@ -465,10 +403,6 @@ static int __maybe_unused cs35l56_sdw_runtime_resume(struct device *dev) if (ret) return ret; - /* Re-enable SoundWire interrupts */ - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - return 0; } @@ -479,32 +413,9 @@ static int __maybe_unused cs35l56_sdw_system_suspend(struct device *dev) if (!cs35l56->base.init_done) return 0; - /* - * Disable SoundWire interrupts. - * Flush - don't cancel because that could leave an unbalanced pm_runtime_get. - */ - cs35l56->sdw_irq_no_unmask = true; - flush_work(&cs35l56->sdw_irq_work); - - /* Mask interrupts and flush in case sdw_irq_work was queued again */ - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - flush_work(&cs35l56->sdw_irq_work); - return cs35l56_system_suspend(dev); } -static int __maybe_unused cs35l56_sdw_system_resume(struct device *dev) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); - - cs35l56->sdw_irq_no_unmask = false; - /* runtime_resume re-enables the interrupt */ - - return cs35l56_system_resume(dev); -} - static int cs35l56_sdw_probe(struct sdw_slave *peripheral, const struct sdw_device_id *id) { struct device *dev = &peripheral->dev; @@ -517,7 +428,6 @@ static int cs35l56_sdw_probe(struct sdw_slave *peripheral, const struct sdw_devi cs35l56->base.dev = dev; cs35l56->sdw_peripheral = peripheral; - INIT_WORK(&cs35l56->sdw_irq_work, cs35l56_sdw_irq_work); dev_set_drvdata(dev, cs35l56); @@ -542,13 +452,6 @@ static int cs35l56_sdw_remove(struct sdw_slave *peripheral) { struct cs35l56_private *cs35l56 = dev_get_drvdata(&peripheral->dev); - /* Disable SoundWire interrupts */ - cs35l56->sdw_irq_no_unmask = true; - cancel_work_sync(&cs35l56->sdw_irq_work); - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - cs35l56_remove(cs35l56); return 0; @@ -556,9 +459,8 @@ static int cs35l56_sdw_remove(struct sdw_slave *peripheral) static const struct dev_pm_ops cs35l56_sdw_pm = { SET_RUNTIME_PM_OPS(cs35l56_sdw_runtime_suspend, cs35l56_sdw_runtime_resume, NULL) - SYSTEM_SLEEP_PM_OPS(cs35l56_sdw_system_suspend, cs35l56_sdw_system_resume) + SYSTEM_SLEEP_PM_OPS(cs35l56_sdw_system_suspend, cs35l56_system_resume) LATE_SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend_late, cs35l56_system_resume_early) - /* NOIRQ stage not needed, SoundWire doesn't use a hard IRQ */ }; static const struct sdw_device_id cs35l56_sdw_id[] = { diff --git a/sound/soc/codecs/cs35l56-shared.c b/sound/soc/codecs/cs35l56-shared.c index e45e9ae01bc6..55adbc3d26ae 100644 --- a/sound/soc/codecs/cs35l56-shared.c +++ b/sound/soc/codecs/cs35l56-shared.c @@ -37,6 +37,7 @@ static const struct reg_sequence cs35l56_patch[] = { { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, + { CS35L56_IRQ1_MASK_20, 0x15fff19c }, /* These are not reset by a soft-reset, so patch to defaults. */ { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, @@ -70,12 +71,6 @@ static const struct reg_default cs35l56_reg_defaults[] = { { CS35L56_SWIRE_DP3_CH2_INPUT, 0x00000019 }, { CS35L56_SWIRE_DP3_CH3_INPUT, 0x00000029 }, { CS35L56_SWIRE_DP3_CH4_INPUT, 0x00000028 }, - { CS35L56_IRQ1_MASK_1, 0x83ffffff }, - { CS35L56_IRQ1_MASK_2, 0xffff7fff }, - { CS35L56_IRQ1_MASK_4, 0xe0ffffff }, - { CS35L56_IRQ1_MASK_8, 0xfc000fff }, - { CS35L56_IRQ1_MASK_18, 0x1f7df0ff }, - { CS35L56_IRQ1_MASK_20, 0x15c00000 }, { CS35L56_MAIN_RENDER_USER_MUTE, 0x00000000 }, { CS35L56_MAIN_RENDER_USER_VOLUME, 0x00000000 }, { CS35L56_MAIN_POSTURE_NUMBER, 0x00000000 }, @@ -329,91 +324,6 @@ void cs35l56_system_reset(struct cs35l56_base *cs35l56_base, bool is_soundwire) } EXPORT_SYMBOL_NS_GPL(cs35l56_system_reset, SND_SOC_CS35L56_SHARED); -int cs35l56_irq_request(struct cs35l56_base *cs35l56_base, int irq) -{ - int ret; - - if (irq < 1) - return 0; - - ret = devm_request_threaded_irq(cs35l56_base->dev, irq, NULL, cs35l56_irq, - IRQF_ONESHOT | IRQF_SHARED | IRQF_TRIGGER_LOW, - "cs35l56", cs35l56_base); - if (!ret) - cs35l56_base->irq = irq; - else - dev_err(cs35l56_base->dev, "Failed to get IRQ: %d\n", ret); - - return ret; -} -EXPORT_SYMBOL_NS_GPL(cs35l56_irq_request, SND_SOC_CS35L56_SHARED); - -irqreturn_t cs35l56_irq(int irq, void *data) -{ - struct cs35l56_base *cs35l56_base = data; - unsigned int status1 = 0, status8 = 0, status20 = 0; - unsigned int mask1, mask8, mask20; - unsigned int val; - int rv; - - irqreturn_t ret = IRQ_NONE; - - if (!cs35l56_base->init_done) - return IRQ_NONE; - - mutex_lock(&cs35l56_base->irq_lock); - - rv = pm_runtime_resume_and_get(cs35l56_base->dev); - if (rv < 0) { - dev_err(cs35l56_base->dev, "irq: failed to get pm_runtime: %d\n", rv); - goto err_unlock; - } - - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_STATUS, &val); - if ((val & CS35L56_IRQ1_STS_MASK) == 0) { - dev_dbg(cs35l56_base->dev, "Spurious IRQ: no pending interrupt\n"); - goto err; - } - - /* Ack interrupts */ - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_EINT_1, &status1); - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_MASK_1, &mask1); - status1 &= ~mask1; - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_EINT_1, status1); - - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_EINT_8, &status8); - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_MASK_8, &mask8); - status8 &= ~mask8; - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_EINT_8, status8); - - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_EINT_20, &status20); - regmap_read(cs35l56_base->regmap, CS35L56_IRQ1_MASK_20, &mask20); - status20 &= ~mask20; - /* We don't want EINT20 but they default to unmasked: force mask */ - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_MASK_20, 0xffffffff); - - dev_dbg(cs35l56_base->dev, "%s: %#x %#x\n", __func__, status1, status8); - - /* Check to see if unmasked bits are active */ - if (!status1 && !status8 && !status20) - goto err; - - if (status1 & CS35L56_AMP_SHORT_ERR_EINT1_MASK) - dev_crit(cs35l56_base->dev, "Amp short error\n"); - - if (status8 & CS35L56_TEMP_ERR_EINT1_MASK) - dev_crit(cs35l56_base->dev, "Overtemp error\n"); - - ret = IRQ_HANDLED; - -err: - pm_runtime_put(cs35l56_base->dev); -err_unlock: - mutex_unlock(&cs35l56_base->irq_lock); - - return ret; -} -EXPORT_SYMBOL_NS_GPL(cs35l56_irq, SND_SOC_CS35L56_SHARED); int cs35l56_is_fw_reload_needed(struct cs35l56_base *cs35l56_base) { @@ -763,15 +673,6 @@ int cs35l56_hw_init(struct cs35l56_base *cs35l56_base) cs35l56_base->type, cs35l56_base->secured ? "s" : "", cs35l56_base->rev, otpid, fw_ver >> 16, (fw_ver >> 8) & 0xff, fw_ver & 0xff, !fw_missing); - /* Wake source and *_BLOCKED interrupts default to unmasked, so mask them */ - regmap_write(cs35l56_base->regmap, CS35L56_IRQ1_MASK_20, 0xffffffff); - regmap_update_bits(cs35l56_base->regmap, CS35L56_IRQ1_MASK_1, - CS35L56_AMP_SHORT_ERR_EINT1_MASK, - 0); - regmap_update_bits(cs35l56_base->regmap, CS35L56_IRQ1_MASK_8, - CS35L56_TEMP_ERR_EINT1_MASK, - 0); - return 0; } EXPORT_SYMBOL_NS_GPL(cs35l56_hw_init, SND_SOC_CS35L56_SHARED); diff --git a/sound/soc/codecs/cs35l56-spi.c b/sound/soc/codecs/cs35l56-spi.c index b07b798b0b45..1382b8903ea7 100644 --- a/sound/soc/codecs/cs35l56-spi.c +++ b/sound/soc/codecs/cs35l56-spi.c @@ -39,8 +39,6 @@ static int cs35l56_spi_probe(struct spi_device *spi) return ret; ret = cs35l56_init(cs35l56); - if (ret == 0) - ret = cs35l56_irq_request(&cs35l56->base, spi->irq); if (ret < 0) cs35l56_remove(cs35l56); diff --git a/sound/soc/codecs/cs35l56.c b/sound/soc/codecs/cs35l56.c index 757ade6373ed..cf2aee6cda6c 100644 --- a/sound/soc/codecs/cs35l56.c +++ b/sound/soc/codecs/cs35l56.c @@ -12,7 +12,6 @@ #include <linux/delay.h> #include <linux/err.h> #include <linux/gpio/consumer.h> -#include <linux/interrupt.h> #include <linux/math.h> #include <linux/module.h> #include <linux/pm.h> @@ -702,23 +701,9 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing { int ret; - /* - * Disable SoundWire interrupts to prevent race with IRQ work. - * Setting sdw_irq_no_unmask prevents the handler re-enabling - * the SoundWire interrupt. - */ - if (cs35l56->sdw_peripheral) { - cs35l56->sdw_irq_no_unmask = true; - flush_work(&cs35l56->sdw_irq_work); - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, 0); - sdw_read_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1); - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_STAT_1, 0xFF); - flush_work(&cs35l56->sdw_irq_work); - } - ret = cs35l56_firmware_shutdown(&cs35l56->base); if (ret) - goto err; + return; /* * Use wm_adsp to load and apply the firmware patch and coefficient files, @@ -728,11 +713,9 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing ret = wm_adsp_power_up(&cs35l56->dsp, !!firmware_missing); if (ret) { dev_dbg(cs35l56->base.dev, "%s: wm_adsp_power_up ret %d\n", __func__, ret); - goto err; + return; } - mutex_lock(&cs35l56->base.irq_lock); - reinit_completion(&cs35l56->init_completion); cs35l56->soft_resetting = true; @@ -748,10 +731,10 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing msecs_to_jiffies(5000))) { dev_err(cs35l56->base.dev, "%s: init_completion timed out (SDW)\n", __func__); - goto err_unlock; + return; } } else if (cs35l56_init(cs35l56)) { - goto err_unlock; + return; } regmap_clear_bits(cs35l56->base.regmap, CS35L56_PROTECTION_STATUS, @@ -760,16 +743,6 @@ static void cs35l56_patch(struct cs35l56_private *cs35l56, bool firmware_missing if (cs35l56_write_cal(cs35l56) == 0) cs35l56_mbox_send(&cs35l56->base, CS35L56_MBOX_CMD_AUDIO_REINIT); - -err_unlock: - mutex_unlock(&cs35l56->base.irq_lock); -err: - /* Re-enable SoundWire interrupts */ - if (cs35l56->sdw_peripheral) { - cs35l56->sdw_irq_no_unmask = false; - sdw_write_no_pm(cs35l56->sdw_peripheral, CS35L56_SDW_GEN_INT_MASK_1, - CS35L56_SDW_INT_MASK_CODEC_IRQ); - } } static void cs35l56_dsp_work(struct work_struct *work) @@ -956,15 +929,6 @@ int cs35l56_system_suspend(struct device *dev) if (cs35l56->component) flush_work(&cs35l56->dsp_work); - /* - * The interrupt line is normally shared, but after we start suspending - * we can't check if our device is the source of an interrupt, and can't - * clear it. Prevent this race by temporarily disabling the parent irq - * until we reach _no_irq. - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - return pm_runtime_force_suspend(dev); } EXPORT_SYMBOL_GPL(cs35l56_system_suspend); @@ -991,40 +955,6 @@ int cs35l56_system_suspend_late(struct device *dev) } EXPORT_SYMBOL_GPL(cs35l56_system_suspend_late); -int cs35l56_system_suspend_no_irq(struct device *dev) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); - - dev_dbg(dev, "system_suspend_no_irq\n"); - - /* Handlers are now disabled so the parent IRQ can safely be re-enabled. */ - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - - return 0; -} -EXPORT_SYMBOL_GPL(cs35l56_system_suspend_no_irq); - -int cs35l56_system_resume_no_irq(struct device *dev) -{ - struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); - - dev_dbg(dev, "system_resume_no_irq\n"); - - /* - * WAKE interrupts unmask if the CS35L56 hibernates, which can cause - * spurious interrupts, and the interrupt line is normally shared. - * We can't check if our device is the source of an interrupt, and can't - * clear it, until it has fully resumed. Prevent this race by temporarily - * disabling the parent irq until we complete resume(). - */ - if (cs35l56->base.irq) - disable_irq(cs35l56->base.irq); - - return 0; -} -EXPORT_SYMBOL_GPL(cs35l56_system_resume_no_irq); - int cs35l56_system_resume_early(struct device *dev) { struct cs35l56_private *cs35l56 = dev_get_drvdata(dev); @@ -1065,11 +995,8 @@ int cs35l56_system_resume(struct device *dev) */ cs35l56_wait_control_port_ready(); - /* Undo pm_runtime_force_suspend() before re-enabling the irq */ + /* Undo pm_runtime_force_suspend() */ ret = pm_runtime_force_resume(dev); - if (cs35l56->base.irq) - enable_irq(cs35l56->base.irq); - if (ret) return ret; @@ -1267,7 +1194,6 @@ int cs35l56_common_probe(struct cs35l56_private *cs35l56) int ret; init_completion(&cs35l56->init_completion); - mutex_init(&cs35l56->base.irq_lock); cs35l56->base.cal_index = -1; cs35l56->speaker_id = -ENOENT; @@ -1428,13 +1354,6 @@ void cs35l56_remove(struct cs35l56_private *cs35l56) { cs35l56->base.init_done = false; - /* - * WAKE IRQs unmask if CS35L56 hibernates so free the handler to - * prevent it racing with remove(). - */ - if (cs35l56->base.irq) - devm_free_irq(cs35l56->base.dev, cs35l56->base.irq, &cs35l56->base); - flush_workqueue(cs35l56->dsp_wq); destroy_workqueue(cs35l56->dsp_wq); @@ -1454,7 +1373,6 @@ EXPORT_NS_GPL_DEV_PM_OPS(cs35l56_pm_ops_i2c_spi, SND_SOC_CS35L56_CORE) = { SET_RUNTIME_PM_OPS(cs35l56_runtime_suspend_i2c_spi, cs35l56_runtime_resume_i2c_spi, NULL) SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend, cs35l56_system_resume) LATE_SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend_late, cs35l56_system_resume_early) - NOIRQ_SYSTEM_SLEEP_PM_OPS(cs35l56_system_suspend_no_irq, cs35l56_system_resume_no_irq) }; #endif diff --git a/sound/soc/codecs/cs35l56.h b/sound/soc/codecs/cs35l56.h index 8a987ec01507..0ebae11c1cd0 100644 --- a/sound/soc/codecs/cs35l56.h +++ b/sound/soc/codecs/cs35l56.h @@ -16,10 +16,6 @@ #include <sound/cs35l56.h> #include "wm_adsp.h" -#define CS35L56_SDW_GEN_INT_STAT_1 0xc0 -#define CS35L56_SDW_GEN_INT_MASK_1 0xc1 -#define CS35L56_SDW_INT_MASK_CODEC_IRQ BIT(0) - #define CS35L56_SDW_INVALID_BUS_SCALE 0xf #define CS35L56_RX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) @@ -38,8 +34,6 @@ struct cs35l56_private { struct snd_soc_component *component; struct regulator_bulk_data supplies[CS35L56_NUM_BULK_SUPPLIES]; struct sdw_slave *sdw_peripheral; - struct work_struct sdw_irq_work; - bool sdw_irq_no_unmask; bool soft_resetting; bool sdw_attached; struct completion init_completion; @@ -57,12 +51,8 @@ extern const struct dev_pm_ops cs35l56_pm_ops_i2c_spi; int cs35l56_system_suspend(struct device *dev); int cs35l56_system_suspend_late(struct device *dev); -int cs35l56_system_suspend_no_irq(struct device *dev); -int cs35l56_system_resume_no_irq(struct device *dev); int cs35l56_system_resume_early(struct device *dev); int cs35l56_system_resume(struct device *dev); -irqreturn_t cs35l56_irq(int irq, void *data); -int cs35l56_irq_request(struct cs35l56_base *cs35l56_base, int irq); int cs35l56_common_probe(struct cs35l56_private *cs35l56); int cs35l56_init(struct cs35l56_private *cs35l56); void cs35l56_remove(struct cs35l56_private *cs35l56); -- 2.43.0

2 5

[PATCH v26 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 04 Sep '24

04 Sep '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userspace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platform card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v26: - Cleaned up drivers based on errors from checkpatch - Fixed several typos using codespell - Removed any vendor specific notation from USB SND offload mixer patch Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addressed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnecessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userspsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (31): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: Allow for secondary interrupter to set IMOD usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 429 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 34 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 54 +- drivers/usb/host/xhci-sideband.c | 425 ++++ drivers/usb/host/xhci.c | 45 +- drivers/usb/host/xhci.h | 19 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 71 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 135 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 194 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 391 ++++ sound/soc/qcom/sm8250.c | 24 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 30 + sound/soc/soc-usb.c | 338 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 4 +- sound/usb/card.c | 110 + sound/usb/card.h | 17 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1968 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 863 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 164 ++ 53 files changed, 5820 insertions(+), 56 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

2 64

[PATCH] ASoC: dt-bindings: renesas,rsnd: add post-init-providers property
by Kuninori Morimoto 04 Sep '24

04 Sep '24

sometimes, fwnode dependency cycles can be broken, and in such case, rsnd driver will never be probed. To avoid such case, we would like to use post-init-providers to avoid it. Enable post-init-providers property on rsnd. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/renesas,rsnd.yaml | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml index 07ec6247d9def..cb44478d9e420 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -112,6 +112,10 @@ properties: description: List of necessary clock names. # details are defined below + post-init-providers: + description: ignore fw_devlink dependency cycles and avoid driver probe dead-lock + $ref: /schemas/types.yaml#/definitions/phandle-array + # ports is below port: $ref: audio-graph-port.yaml#/definitions/port-base -- 2.43.0

3 2

[PATCH 00/12] Add i2s/tdm support for acp7.0 and acp7.1 platforms
by Venkata Prasad Potturu 04 Sep '24

04 Sep '24

1. Refactor acp generic driver to support all platforms. 2. Add i2s/tdm and support for acp7.0 and acp7.1 platforms. Venkata Prasad Potturu (12): ASoC: amd: acp: Refactor TDM slots selction based on acp revision id ASoC: amd: acp: Refactor I2S dai driver ASoC: amd: acp: Update pcm hardware capabilities for acp6.3 platform ASoC: amd: acp: Add I2S TDM support for acp6.3 platform ASoC: amd: acp: Update pcm hardware capabilities for acp7.0 platform ASoC: amd: acp: Add I2S master clock generation support for acp7.0 platform ASoC: amd: acp: Set i2s clock for acp7.0 platform ASoC: amd: acp: Modify max channels and sample rate support for acp70 dai driver ASoC: amd: acp: Add I2S TDM support for acp7.0 platform ASoC: amd: acp: Add pte configuration for ACP7.0 platform ASoC: amd: acp: Add i2s master clock generation support for acp7.1 platform ASoC: amd: acp: Add I2S TDM support for acp7.1 platform sound/soc/amd/acp/acp-i2s.c | 188 ++++++++++++++++++++------ sound/soc/amd/acp/acp-legacy-common.c | 32 ++--- sound/soc/amd/acp/acp-pdm.c | 7 +- sound/soc/amd/acp/acp-platform.c | 117 ++++++++++++++-- sound/soc/amd/acp/acp70.c | 73 +++++++--- sound/soc/amd/acp/amd.h | 24 ++-- sound/soc/amd/acp/chip_offset_byte.h | 84 +++++++----- 7 files changed, 387 insertions(+), 138 deletions(-) -- 2.39.2

2 13

[PATCH] ALSA: hda/realtek: Support mute LED on HP Laptop 14-dq2xxx
by Maximilien Perreault 04 Sep '24

04 Sep '24

The mute LED on this HP laptop uses ALC236 and requires a quirk to function. This patch enables the existing quirk for the device. Signed-off-by: Maximilien Perreault <maximilienperreault(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f67190a8c..6f96c0c67 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10183,6 +10183,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x87f5, "HP", ALC287_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x87f6, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), SND_PCI_QUIRK(0x103c, 0x87f7, "HP Spectre x360 14", ALC245_FIXUP_HP_X360_AMP), + SND_PCI_QUIRK(0x103c, 0x87fd, "HP Laptop 14-dq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x87fe, "HP Laptop 15s-fq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x8805, "HP ProBook 650 G8 Notebook PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x880d, "HP EliteBook 830 G8 Notebook PC", ALC285_FIXUP_HP_GPIO_LED), -- 2.46.0

2 1

[PATCH] soc: qcom: apr: Fix some parenthesis alignment coding style issues
by Riyan Dhiman 04 Sep '24

04 Sep '24

Adhere to Linux kernel coding style. Issue reported by checkpatch: - CHECK: Alignment should match open parenthesis It has no functional changes. Signed-off-by: Riyan Dhiman <riyandhiman14(a)gmail.com> --- drivers/soc/qcom/apr.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/drivers/soc/qcom/apr.c b/drivers/soc/qcom/apr.c index 4fbff3a890e2..db1326bdb170 100644 --- a/drivers/soc/qcom/apr.c +++ b/drivers/soc/qcom/apr.c @@ -88,7 +88,7 @@ void gpr_free_port(gpr_port_t *port) EXPORT_SYMBOL_GPL(gpr_free_port); gpr_port_t *gpr_alloc_port(struct apr_device *gdev, struct device *dev, - gpr_port_cb cb, void *priv) + gpr_port_cb cb, void *priv) { struct packet_router *pr = dev_get_drvdata(gdev->dev.parent); gpr_port_t *port; @@ -159,7 +159,7 @@ static void apr_dev_release(struct device *dev) } static int apr_callback(struct rpmsg_device *rpdev, void *buf, - int len, void *priv, u32 addr) + int len, void *priv, u32 addr) { struct packet_router *apr = dev_get_drvdata(&rpdev->dev); struct apr_rx_buf *abuf; @@ -222,9 +222,9 @@ static int apr_do_rx_callback(struct packet_router *apr, struct apr_rx_buf *abuf } if (hdr->src_domain >= APR_DOMAIN_MAX || - hdr->dest_domain >= APR_DOMAIN_MAX || - hdr->src_svc >= APR_SVC_MAX || - hdr->dest_svc >= APR_SVC_MAX) { + hdr->dest_domain >= APR_DOMAIN_MAX || + hdr->src_svc >= APR_SVC_MAX || + hdr->dest_svc >= APR_SVC_MAX) { dev_err(apr->dev, "APR: Wrong APR header\n"); return -EINVAL; } -- 2.46.0

2 1

Qestion: of property has dead-lock to call .probe()
by Kuninori Morimoto 04 Sep '24

04 Sep '24

Hi Saravana, Rob Cc Mark/ALSA-ML, Geert, Laurent I got dead-lock issue on drivers/of/property.c to call .probe() I'm now using v6.11-rc5. I'm using 3 devices, and these are connected via OF-Graph. These are connected like this so far. Let's call it as [Style-A] [Style-A] +---+ +---+ |(A)+--+ | +-+-+ | | | |(B)| +-+-+ | | |(C)+--+ | +---+ +---+ These are using port/endpoint to connect. It works well for now, no issue I have so far. (B) is connector for (A)-(C) data, thus, (A) and (C) should be probed first, and (B) needs probe later (For ALSA member, this (B) is Audio Graph Card2). Here, (A)-(C) can connect directly for some data. Now it is using both data connection (= (A)-(B)-(C) and (A)-(C)) Now, I want to disconnect (A)-(C) connection, like below Let's call it as [Style-B] [Style-B] +---+ +---+ |(A)+--+ | +---+ | | |(B)| +---+ | | |(C)+--+ | +---+ +---+ Then, it seems dead-lock happen. In my debug, it seems... - (B) is handled as supplier for (A). - (B) probe() is called, but it needs (A) info which is not yet probed. So it returns -EPROBE_DEFER. - Because (B) is not probed, (A) probe() never called In [Style-A], it seems __fwnode_link_cycle() is called for (A)-(B) connection, so the dead-lock will be solved. But it is not called for [Style-B]. Because of that (B) is always handled as supplier for (A). If I used below patch, and use "non-supplier" property on (B), this dead-lock issue was solved. But I know this is not a good solution. I think it is very normal connection, not super special. How can I solve this issue on correct way ? Or how can I indicate you my issue more detail ? I can add debug patch and test it if you can indicate it to me. ----------------------------------------------- diff --git a/drivers/of/property.c b/drivers/of/property.c index def152c61049..2f08210c2ea4 100644 --- a/drivers/of/property.c +++ b/drivers/of/property.c @@ -1502,11 +1502,22 @@ static struct device_node *parse_remote_endpoint(struct device_node *np, const char *prop_name, int index) { + struct device_node *node; + /* Return NULL for index > 0 to signify end of remote-endpoints. */ if (index > 0 || strcmp(prop_name, "remote-endpoint")) return NULL; - return of_graph_get_remote_port_parent(np); + node = of_graph_get_remote_port_parent(np); + + /* + * There is clearly non-supplier node which is connected via "remote-endpoint". + * Ignore it, otherwise dead-lock might occur + */ + if (of_property_present(node, "non-supplier")) + return NULL; + + return node; } ----------------------------------------------- Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[PATCH] ASoC: tlv320aic32x4: Add multi endpoint support
by Marek Vasut 04 Sep '24

04 Sep '24

Support multiple endpoints on TLV320AIC32x4 codec port when used in of_graph context. This patch allows to share the codec port between two CPU DAIs. Example: Custom STM32MP157C board uses TLV320AIC32x4 audio codec. This codec is connected to two serial audio interfaces, which are configured either as rx or tx. From AsoC point of view the topolgy is the following: // 2 CPU DAIs (SAI2A/B), 1 Codec (TLV320AIC32x4) Playback: CPU-A-DAI(slave) -> (master)CODEC-DAI/port0 Record: CPU-B-DAI(slave) <- (master)CODEC-DAI/port0 In the DT two endpoints have to be associated to the codec port: tlv320aic32x4_port: port { tlv320aic32x4_tx_endpoint: endpoint@0 { remote-endpoint = <&sai2a_endpoint>; }; tlv320aic32x4_rx_endpoint: endpoint@1 { remote-endpoint = <&sai2b_endpoint>; }; }; However, when the audio graph card parses the codec nodes, it expects to find DAI interface indexes matching the endpoints indexes. The current patch forces the use of DAI id 0 for both endpoints, which allows to share the codec DAI between the two CPU DAIs for playback and capture streams respectively. Signed-off-by: Marek Vasut <marex(a)denx.de> --- Cc: Baojun Xu <baojun.xu(a)ti.com> Cc: Charles Keepax <ckeepax(a)opensource.cirrus.com> Cc: Christoph Niedermaier <cniedermaier(a)dh-electronics.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kevin Lu <kevin-lu(a)ti.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Marek Vasut <marex(a)denx.de> Cc: Mark Brown <broonie(a)kernel.org> Cc: Philipp Zabel <p.zabel(a)pengutronix.de> Cc: Shenghao Ding <shenghao-ding(a)ti.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: kernel(a)dh-electronics.com Cc: linux-sound(a)vger.kernel.org --- sound/soc/codecs/tlv320aic32x4.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 5c0c81da06dba..54ea4bc58c276 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -1073,6 +1073,13 @@ static int aic32x4_component_probe(struct snd_soc_component *component) return 0; } +static int aic32x4_of_xlate_dai_id(struct snd_soc_component *component, + struct device_node *endpoint) +{ + /* return dai id 0, whatever the endpoint index */ + return 0; +} + static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .probe = aic32x4_component_probe, .set_bias_level = aic32x4_set_bias_level, @@ -1082,6 +1089,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4 = { .num_dapm_widgets = ARRAY_SIZE(aic32x4_dapm_widgets), .dapm_routes = aic32x4_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, @@ -1203,6 +1211,7 @@ static const struct snd_soc_component_driver soc_component_dev_aic32x4_tas2505 = .num_dapm_widgets = ARRAY_SIZE(aic32x4_tas2505_dapm_widgets), .dapm_routes = aic32x4_tas2505_dapm_routes, .num_dapm_routes = ARRAY_SIZE(aic32x4_tas2505_dapm_routes), + .of_xlate_dai_id = aic32x4_of_xlate_dai_id, .suspend_bias_off = 1, .idle_bias_on = 1, .use_pmdown_time = 1, -- 2.45.2

2 1

[PATCH] ASOC: Intel: fix parenthesis alignment and space after cast coding style issue
by Riyan Dhiman 03 Sep '24

03 Sep '24

Adhere to Linux kernel coding style. Reported by checkpatch: - CHECK: Alignment should match open parenthesis - CHECK: No space is necessary after a cast It has no functional changes. Signed-off-by: Riyan Dhiman <riyandhiman14(a)gmail.com> --- sound/soc/intel/atom/sst/sst.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/atom/sst/sst.c b/sound/soc/intel/atom/sst/sst.c index e0357d257c6c..38d29d545b52 100644 --- a/sound/soc/intel/atom/sst/sst.c +++ b/sound/soc/intel/atom/sst/sst.c @@ -50,7 +50,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) union sst_imr_reg_mrfld imr; struct ipc_post *msg = NULL; unsigned int size; - struct intel_sst_drv *drv = (struct intel_sst_drv *) context; + struct intel_sst_drv *drv = (struct intel_sst_drv *)context; irqreturn_t retval = IRQ_HANDLED; /* Interrupt arrived, check src */ @@ -60,7 +60,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) /* Clear done bit */ spin_lock(&drv->ipc_spin_lock); header.full = sst_shim_read64(drv->shim, - drv->ipc_reg.ipcx); + drv->ipc_reg.ipcx); header.p.header_high.part.done = 0; sst_shim_write64(drv->shim, drv->ipc_reg.ipcx, header.full); @@ -92,7 +92,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) size = header.p.header_low_payload; if (sst_validate_mailbox_size(size)) { memcpy_fromio(msg->mailbox_data, - drv->mailbox + drv->mailbox_recv_offset, size); + drv->mailbox + drv->mailbox_recv_offset, size); } else { dev_err(drv->dev, "Mailbox not copied, payload size is: %u\n", size); @@ -114,7 +114,7 @@ static irqreturn_t intel_sst_interrupt_mrfld(int irq, void *context) static irqreturn_t intel_sst_irq_thread_mrfld(int irq, void *context) { - struct intel_sst_drv *drv = (struct intel_sst_drv *) context; + struct intel_sst_drv *drv = (struct intel_sst_drv *)context; struct ipc_post *__msg, *msg; unsigned long irq_flags; @@ -146,8 +146,8 @@ static int sst_save_dsp_context_v2(struct intel_sst_drv *sst) int ret = 0; ret = sst_prepare_and_post_msg(sst, SST_TASK_ID_MEDIA, IPC_CMD, - IPC_PREP_D3, PIPE_RSVD, 0, NULL, NULL, - true, true, false, true); + IPC_PREP_D3, PIPE_RSVD, 0, NULL, NULL, + true, true, false, true); if (ret < 0) { dev_err(sst->dev, "not suspending FW!!, Err: %d\n", ret); @@ -228,7 +228,7 @@ static void sst_init_locks(struct intel_sst_drv *ctx) * we are fine with using unsigned short as dev_id type. */ int sst_alloc_drv_context(struct intel_sst_drv **ctx, - struct device *dev, unsigned short dev_id) + struct device *dev, unsigned short dev_id) { *ctx = devm_kzalloc(dev, sizeof(struct intel_sst_drv), GFP_KERNEL); if (!(*ctx)) @@ -242,7 +242,7 @@ int sst_alloc_drv_context(struct intel_sst_drv **ctx, EXPORT_SYMBOL_GPL(sst_alloc_drv_context); static ssize_t firmware_version_show(struct device *dev, - struct device_attribute *attr, char *buf) + struct device_attribute *attr, char *buf) { struct intel_sst_drv *ctx = dev_get_drvdata(dev); @@ -302,7 +302,7 @@ int sst_context_init(struct intel_sst_drv *ctx) ctx->ipc_reg.ipcd = SST_IPCD + ctx->pdata->ipc_info->ipc_offset; dev_info(ctx->dev, "Got drv data max stream %d\n", - ctx->info.max_streams); + ctx->info.max_streams); for (i = 1; i <= ctx->info.max_streams; i++) { struct stream_info *stream = &ctx->streams[i]; @@ -325,7 +325,7 @@ int sst_context_init(struct intel_sst_drv *ctx) sst_shim_write64(ctx->shim, SST_IMRX, 0xFFFF0038); ctx->qos = devm_kzalloc(ctx->dev, - sizeof(struct pm_qos_request), GFP_KERNEL); + sizeof(struct pm_qos_request), GFP_KERNEL); if (!ctx->qos) { ret = -ENOMEM; goto do_free_mem; -- 2.46.0

1 0

ucm2: sof-soundwire: add RT1320 amplifier
by GitHub pull_request - opened 03 Sep '24

03 Sep '24

alsa-project/alsa-ucm-conf pull request #439 was opened from shumingfan: Add support for rt1320 amplifier Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/439 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/439.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2] ASoC: tas2781: Add Calibration Kcontrols for Chromebook
by Shenghao Ding 03 Sep '24

03 Sep '24

Add calibration related kcontrol for speaker impedance calibration and speaker leakage check for Chromebook Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - Add channel no into the log for error debug - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/202407131… - renamae tasdevice_chn_switch to put it in a single line - With inverted check this entire function becomes neater in tasdev_chn_switch. - Add registers for TAS2563 and TAS2781 calibration. - Add cali_data_backup for regsiter restore after calibration. - Add is_user_space_calidata to store the flag where the calibrated from, user space or bin file from driver parsing. - tasdevice_chn_switch for chip switch among multiple chips(tas2563 or tas2781) - Add loading the calibrated values from user space in tasdev_load_calibrated_data - Correct no dsp no work, it can still work in bypass mode. - Add calibrated register setting for tas2563&tas2781. - Define tas2781_cali_control and tas2563_cali_control for calibration --- include/sound/tas2781.h | 68 +++ sound/soc/codecs/tas2781-comlib.c | 26 + sound/soc/codecs/tas2781-fmwlib.c | 60 +- sound/soc/codecs/tas2781-i2c.c | 961 +++++++++++++++++++++++++++++- 4 files changed, 1104 insertions(+), 11 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index dbda552398b5..a9ccbbd1945c 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -49,12 +49,59 @@ /*I2C Checksum */ #define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +/* XM_340 */ +#define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x02, 0x4c) +/* XM_341 */ +#define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x02, 0x64) + /* Volume control */ #define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) #define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) +#define TAS2563_IDLE TASDEVICE_REG(0x00, 0x00, 0x3e) +#define TAS2563_PRM_R0_REG TASDEVICE_REG(0x00, 0x0f, 0x34) + +#define TAS2563_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x02, 0x70) +#define TAS2563_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x02, 0x48) + +#define TAS2563_PRM_ENFF_REG TASDEVICE_REG(0x00, 0x0d, 0x54) +#define TAS2563_PRM_DISTCK_REG TASDEVICE_REG(0x00, 0x0d, 0x58) +#define TAS2563_PRM_TE_SCTHR_REG TASDEVICE_REG(0x00, 0x0f, 0x60) +#define TAS2563_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x0d, 0x74) +#define TAS2563_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x0d, 0x7c) +/* prm_Int_B0 */ +#define TAS2563_TE_TA1_REG TASDEVICE_REG(0x00, 0x10, 0x0c) +/* prm_Int_A1 */ +#define TAS2563_TE_TA1_AT_REG TASDEVICE_REG(0x00, 0x10, 0x10) +/* prm_TE_Beta */ +#define TAS2563_TE_TA2_REG TASDEVICE_REG(0x00, 0x0f, 0x64) +/* prm_TE_Beta1 */ +#define TAS2563_TE_AT_REG TASDEVICE_REG(0x00, 0x0f, 0x68) +/* prm_TE_1_Beta1 */ +#define TAS2563_TE_DT_REG TASDEVICE_REG(0x00, 0x0f, 0x70) + +#define TAS2781_PRM_INT_MASK_REG TASDEVICE_REG(0x00, 0x00, 0x3b) +#define TAS2781_PRM_CLK_CFG_REG TASDEVICE_REG(0x00, 0x00, 0x5c) +#define TAS2781_PRM_RSVD_REG TASDEVICE_REG(0x00, 0x01, 0x19) +#define TAS2781_PRM_TEST_57_REG TASDEVICE_REG(0x00, 0xfd, 0x39) +#define TAS2781_PRM_TEST_62_REG TASDEVICE_REG(0x00, 0xfd, 0x3e) +#define TAS2781_PRM_PVDD_UVLO_REG TASDEVICE_REG(0x00, 0x00, 0x71) +#define TAS2781_PRM_CHNL_0_REG TASDEVICE_REG(0x00, 0x00, 0x03) +#define TAS2781_PRM_NG_CFG0_REG TASDEVICE_REG(0x00, 0x00, 0x35) +#define TAS2781_PRM_IDLE_CH_DET_REG TASDEVICE_REG(0x00, 0x00, 0x66) +#define TAS2781_PRM_PLT_FLAG_REG TASDEVICE_REG(0x00, 0x14, 0x38) +#define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) +#define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) + +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) +#define TAS2781_TEST_PAGE_UNLOCK 0x0D + +#define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) +#define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) +#define TAS2781_RUNTIME_RE_REG TASDEVICE_REG(0x64, 0x63, 0x44) + #define TASDEVICE_CMD_SING_W 0x1 #define TASDEVICE_CMD_BURST 0x2 #define TASDEVICE_CMD_DELAY 0x3 @@ -70,7 +117,15 @@ enum device_catlog_id { OTHERS }; +struct bulk_reg_val { + int reg; + unsigned char val[4]; + unsigned char val_len; + bool is_locked; +}; + struct tasdevice { + struct bulk_reg_val *cali_data_backup; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -81,9 +136,19 @@ struct tasdevice { bool is_loaderr; }; +struct cali_reg { + unsigned int r0_reg; + unsigned int r0_low_reg; + unsigned int invr0_reg; + unsigned int pow_reg; + unsigned int tlimit_reg; +}; + struct calidata { unsigned char *data; unsigned long total_sz; + struct cali_reg cali_reg_array; + unsigned int cali_dat_sz_per_dev; }; struct tasdevice_priv { @@ -119,6 +184,7 @@ struct tasdevice_priv { bool force_fwload_status; bool playback_started; bool isacpi; + bool is_user_space_calidata; unsigned int global_addr; int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, @@ -145,6 +211,8 @@ int tasdevice_init(struct tasdevice_priv *tas_priv); void tasdevice_remove(struct tasdevice_priv *tas_priv); int tasdevice_save_calibration(struct tasdevice_priv *tas_priv); void tasdevice_apply_calibration(struct tasdevice_priv *tas_priv); +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn); int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *value); int tasdevice_dev_write(struct tasdevice_priv *tas_priv, diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 664c371796d6..1e0b3aa95749 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -88,6 +88,32 @@ static int tasdevice_change_chn_book(struct tasdevice_priv *tas_priv, return ret; } +int tasdev_chn_switch(struct tasdevice_priv *tas_priv, + unsigned short chn) +{ + struct i2c_client *client = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = &tas_priv->tasdevice[chn]; + struct regmap *map = tas_priv->regmap; + int ret; + + if (client->addr != tasdev->dev_addr) { + client->addr = tasdev->dev_addr; + /* All devices share the same regmap, clear the page + * inside regmap once switching to another device. + * Register 0 at any pages and any books inside tas2781 + * is the same one for page-switching. + */ + ret = regmap_write(map, TASDEVICE_PAGE_SELECT, 0); + if (ret < 0) { + dev_err(tas_priv->dev, "%s, E=%d\n", __func__, ret); + return ret; + } + return 1; + } + return 0; +} +EXPORT_SYMBOL_GPL(tasdev_chn_switch); + int tasdevice_dev_read(struct tasdevice_priv *tas_priv, unsigned short chn, unsigned int reg, unsigned int *val) { diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index f3a7605f0710..3de0132c345d 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2151,20 +2151,61 @@ static int tasdevice_load_data(struct tasdevice_priv *tas_priv, static void tasdev_load_calibrated_data(struct tasdevice_priv *priv, int i) { + struct tasdevice_fw *cal_fmw = priv->tasdevice[i].cali_data_fmw; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *data = cali_data->data; struct tasdevice_calibration *cal; - struct tasdevice_fw *cal_fmw; + int k = i * (cali_data->cali_dat_sz_per_dev + 1); + int rc; - cal_fmw = priv->tasdevice[i].cali_data_fmw; + /* Load the calibrated data from cal bin file */ + if (!priv->is_user_space_calidata && cal_fmw) { + cal = cal_fmw->calibrations; - /* No calibrated data for current devices, playback will go ahead. */ - if (!cal_fmw) + if (cal) + load_calib_data(priv, &cal->dev_data); return; - - cal = cal_fmw->calibrations; - if (!cal) + } + if (!priv->is_user_space_calidata) + return; + /* load calibrated data from user space */ + if (data[k] != i) { + dev_err(priv->dev, "%s: no cal-data for dev %d from usr-spc\n", + __func__, i); return; + } + k++; - load_calib_data(priv, &cal->dev_data); + rc = tasdevice_dev_bulk_write(priv, i, p->r0_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_reg bulk_wr err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->r0_low_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_low_reg err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->invr0_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d invr0_reg err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->pow_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d pow_reg bulk_wr err = %d\n", i, rc); + return; + } + k += 4; + rc = tasdevice_dev_bulk_write(priv, i, p->tlimit_reg, &(data[k]), 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d tlimit_reg err = %d\n", i, rc); + return; + } } int tasdevice_select_tuningprm_cfg(void *context, int prm_no, @@ -2259,9 +2300,10 @@ int tasdevice_select_tuningprm_cfg(void *context, int prm_no, tas_priv->tasdevice[i].cur_conf = cfg_no; } } - } else + } else { dev_dbg(tas_priv->dev, "%s: Unneeded loading dsp conf %d\n", __func__, cfg_no); + } status |= cfg_info[rca_conf_no]->active_dev; diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 59fcce7fef7b..ebd71c7eaa59 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -33,6 +33,143 @@ #include <sound/tas2781-tlv.h> #include <asm/unaligned.h> +static const struct bulk_reg_val tas2563_cali_start_reg[] = { + { + .reg = TAS2563_IDLE, + .val_len = 1, + .val = { 0x00 }, + }, + { + .reg = TAS2563_PRM_ENFF_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_DISTCK_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_TE_SCTHR_REG, + .val_len = 4, + .val = { 0x7f, 0xff, 0xff, 0xff }, + }, + { + .reg = TAS2563_PRM_PLT_FLAG_REG, + .val_len = 4, + .val = { 0x40, 0x00, 0x00, 0x00 }, + }, + { + .reg = TAS2563_PRM_SINEGAIN_REG, + .val_len = 4, + .val = { 0x0a, 0x3d, 0x70, 0xa4 }, + }, + { + .reg = TAS2563_TE_TA1_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA1_AT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_TA2_REG, + .val_len = 4, + .val = { 0x00, 0x06, 0xd3, 0x72 }, + }, + { + .reg = TAS2563_TE_AT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, + { + .reg = TAS2563_TE_DT_REG, + .val_len = 4, + .val = { 0x00, 0x36, 0x91, 0x5e }, + }, +}; + +static const struct bulk_reg_val tas2781_cali_start_reg[] = { + { + .reg = TAS2781_PRM_INT_MASK_REG, + .val = { 0xfe }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CLK_CFG_REG, + .val = { 0xdd }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_RSVD_REG, + .val = { 0x20 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_TEST_57_REG, + .val = { 0x14 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_TEST_62_REG, + .val = { 0x45 }, + .val_len = 1, + .is_locked = true + }, + { + .reg = TAS2781_PRM_PVDD_UVLO_REG, + .val = { 0x03 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_CHNL_0_REG, + .val = { 0xA8 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_NG_CFG0_REG, + .val = { 0xb9 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = TAS2781_PRM_IDLE_CH_DET_REG, + .val = { 0x92 }, + .val_len = 1, + .is_locked = false + }, + { + .reg = 0, + .val = { 0x00, 0x00, 0x00, 0x56 }, // 1mA + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_PLT_FLAG_REG, + .val = { 0x40, 0x00, 0x00, 0x00 }, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN_REG, + .val_len = 4, + .is_locked = false + }, + { + .reg = TAS2781_PRM_SINEGAIN2_REG, + .val_len = 4, + .is_locked = false + }, +}; + static const struct i2c_device_id tasdevice_id[] = { { "tas2563", TAS2563 }, { "tas2781", TAS2781 }, @@ -141,6 +278,556 @@ static int tasdev_force_fwload_put(struct snd_kcontrol *kcontrol, return change; } +static int tasdev_cali_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned char *data = cali_data->data; + unsigned int i = 0; + unsigned int j, k; + int rc; + + guard(mutex)(&priv->codec_lock); + if (!priv->is_user_space_calidata) + return -1; + + if (!p->r0_reg) + return -1; + + dst[i++] = bytes_ext->max; + dst[i++] = 'r'; + + dst[i++] = TASDEVICE_BOOK_ID(p->r0_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->r0_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->r0_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->r0_low_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->r0_low_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->r0_low_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->invr0_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->invr0_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->invr0_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->pow_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->pow_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->pow_reg); + + dst[i++] = TASDEVICE_BOOK_ID(p->tlimit_reg); + dst[i++] = TASDEVICE_PAGE_ID(p->tlimit_reg); + dst[i++] = TASDEVICE_PAGE_REG(p->tlimit_reg); + + for (j = 0, k = 0; j < priv->ndev; j++) { + if (j == data[k]) { + dst[i++] = j; + k++; + } else { + dev_err(priv->dev, "chn %d device %u not match\n", + j, data[k]); + k += 21; + continue; + } + rc = tasdevice_dev_bulk_read(priv, j, p->r0_reg, &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_reg bulk_rd err = %d\n", + j, rc); + i += 20; + k += 20; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d r0_data is not same\n", j); + k += 4; + i += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->r0_low_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d r0_low bulk_rd err = %d\n", + j, rc); + i += 16; + k += 16; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d r0_low is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->invr0_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d invr0 bulk_rd err = %d\n", + j, rc); + i += 12; + k += 12; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d invr0 is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->pow_reg, &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d pow_reg bulk_rd err = %d\n", + j, rc); + i += 8; + k += 8; + continue; + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d pow_reg is not same\n", j); + i += 4; + k += 4; + rc = tasdevice_dev_bulk_read(priv, j, p->tlimit_reg, + &dst[i], 4); + if (rc < 0) { + dev_err(priv->dev, "chn %d tlimit bulk_rd err = %d\n", + j, rc); + } + rc = memcmp(&dst[i], &data[k], 4); + if (rc != 0) + dev_dbg(priv->dev, "chn %d tlimit is not same\n", j); + i += 4; + k += 4; + } + return 0; +} + +static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, + unsigned char *dst) +{ + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int rc = -1; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[0] = i; + rc = tasdevice_dev_bulk_read(tas_priv, i, reg, &dst[1], + 4); + break; + } + } + + return rc; +} + +static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, + int *reg, unsigned char *dat) +{ + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int j; + + if (p == NULL) + return; + + /* Store the current setting from the chip */ + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_read(tas_priv, i, p[j].reg, + (int *)&p[j].val[0]); + } else { + switch (p[j].reg) { + case 0: + if (!reg[0]) + continue; + p[j].reg = reg[0]; + break; + case TAS2781_PRM_PLT_FLAG_REG: + p[j].reg = reg[1]; + break; + case TAS2781_PRM_SINEGAIN_REG: + p[j].reg = reg[2]; + break; + case TAS2781_PRM_SINEGAIN2_REG: + p[j].reg = reg[3]; + break; + } + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } + + /* Update the setting for calibration */ + for (j = 0; j < sum - 2; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + tas2781_cali_start_reg[j].val[0]); + } else { + if (!p[j].reg) + continue; + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + (unsigned char *) + tas2781_cali_start_reg[j].val, 4); + } + } + + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, &dat[1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, &dat[5], 4); +} + +static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dat = ucontrol->value.bytes.data; + int i, reg[4]; + int j = 0; + + guard(mutex)(&priv->codec_lock); + if (priv->chip_id != TAS2781 || bytes_ext->max != dat[0] || + dat[1] != 'r') { + dev_err(priv->dev, "%s: package fmt or chipid incorrect\n", + __func__); + return 0; + } + j += 2; + /* refresh pilot tone and SineGain register */ + for (i = 0; i < ARRAY_SIZE(reg); i++) { + reg[i] = TASDEVICE_REG(dat[j], dat[j + 1], dat[j + 2]); + j += 3; + } + + for (i = 0; i < priv->ndev; i++) { + int k = i * 9 + j; + + if (dat[k] != i) { + dev_err(priv->dev, "%s:no cal-setting for dev %d\n", + __func__, i); + continue; + } + sngl_calib_start(priv, i, reg, dat + k); + } + return 1; +} + +static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) { + if (p[j].is_locked) + tasdevice_dev_write(tas_priv, i, + TAS2781_TEST_UNLOCK_REG, + TAS2781_TEST_PAGE_UNLOCK); + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + } else { + if (!p[j].reg) + continue; + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } + } +} + +static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct bulk_reg_val *q = (struct bulk_reg_val *)tas2563_cali_start_reg; + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int rc = 1; + int i, j; + + guard(mutex)(&tas_priv->codec_lock); + if (tas_priv->chip_id != TAS2563) { + rc = -1; + goto out; + } + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_read(tas_priv, + i, p[j].reg, + (unsigned int *)&p[j].val[0]); + else + tasdevice_dev_bulk_read(tas_priv, + i, p[j].reg, p[j].val, 4); + } + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + q[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + q[j].val, 4); + } + } +out: + return rc; +} + +static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) +{ + const int sum = ARRAY_SIZE(tas2563_cali_start_reg); + int i, j; + + for (i = 0; i < tas_priv->ndev; i++) { + struct tasdevice *tasdev = tas_priv->tasdevice; + struct bulk_reg_val *p = tasdev[i].cali_data_backup; + + if (p == NULL) + continue; + + for (j = 0; j < sum; j++) { + if (p[j].val_len == 1) + tasdevice_dev_write(tas_priv, i, p[j].reg, + p[j].val[0]); + else + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, + p[j].val, 4); + } + } +} + +static int tasdev_calib_stop_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + + guard(mutex)(&priv->codec_lock); + if (priv->chip_id == TAS2563) + tas2563_calib_stop_put(priv); + else + tas2781_calib_stop_put(priv); + + return 1; +} + +static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct calidata *cali_data = &priv->cali_data; + struct cali_reg *p = &cali_data->cali_reg_array; + unsigned char *src = ucontrol->value.bytes.data; + unsigned char *dst = cali_data->data; + int rc = 1, i = 0; + int j; + + guard(mutex)(&priv->codec_lock); + if (src[0] != bytes_ext->max || src[1] != 'r') { + dev_err(priv->dev, "%s: pkg fmt invalid\n", __func__); + return 0; + } + for (j = 0; j < priv->ndev; j++) { + if (src[17 + j * 21] != j) { + dev_err(priv->dev, "%s: pkg fmt invalid\n", __func__); + return 0; + } + } + i += 2; + priv->is_user_space_calidata = true; + + p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + + memcpy(dst, &src[i], cali_data->total_sz); + return rc; +} + +static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + struct tasdevice *tasdev = tas_priv->tasdevice; + unsigned char *dst = ucontrol->value.bytes.data; + int i, val, rc = -1; + + dst[0] = bytes_ext->max; + guard(mutex)(&tas_priv->codec_lock); + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + /* First byte is the device index. */ + dst[1] = i; + rc = tasdevice_dev_read(tas_priv, i, + TAS2781_RUNTIME_LATCH_RE_REG, &val); + if (rc < 0) + dev_err(tas_priv->dev, "%s, get value error\n", + __func__); + else + dst[2] = val; + + break; + } + } + + return rc; +} + +static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG_TF; + else + reg = TAS2563_RUNTIME_RE_REG_TF; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + if (tas_priv->chip_id == TAS2781) + reg = TAS2781_RUNTIME_RE_REG; + else + reg = TAS2563_RUNTIME_RE_REG; + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_r0_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct calidata *cali_data = &tas_priv->cali_data; + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + + if (tas_priv->chip_id == TAS2563) + reg = TAS2563_PRM_R0_REG; + else if (cali_data->cali_reg_array.r0_reg) + reg = cali_data->cali_reg_array.r0_reg; + else + return -1; + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A1_REG; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct soc_bytes_ext *bytes_ext = + (struct soc_bytes_ext *) kcontrol->private_value; + unsigned char *dst = ucontrol->value.bytes.data; + unsigned int reg = TASDEVICE_XM_A2_REG; + int rc = -1; + + guard(mutex)(&tas_priv->codec_lock); + dst[0] = bytes_ext->max; + rc = calib_data_get(tas_priv, reg, &dst[1]); + + return rc; +} + +static int tasdev_nop_get( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + return 0; +} + static int tas2563_digital_gain_get( struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -241,6 +928,16 @@ static const struct snd_kcontrol_new tasdevice_snd_controls[] = { tasdev_force_fwload_get, tasdev_force_fwload_put), }; +static const struct snd_kcontrol_new tasdevice_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Stop", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tasdev_calib_stop_put), + SND_SOC_BYTES_EXT("Amp TF Data", 6, tasdev_tf_data_get, NULL), + SND_SOC_BYTES_EXT("Amp RE Data", 6, tasdev_re_data_get, NULL), + SND_SOC_BYTES_EXT("Amp R0 Data", 6, tasdev_r0_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA1 Data", 6, tasdev_XMA1_data_get, NULL), + SND_SOC_BYTES_EXT("Amp XMA2 Data", 6, tasdev_XMA2_data_get, NULL), +}; + static const struct snd_kcontrol_new tas2781_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Analog Gain", TAS2781_AMP_LEVEL, 1, 0, 20, 0, tas2781_amp_getvol, @@ -250,6 +947,10 @@ static const struct snd_kcontrol_new tas2781_snd_controls[] = { tas2781_digital_putvol, dvc_tlv), }; +static const struct snd_kcontrol_new tas2781_cali_controls[] = { + SND_SOC_BYTES_EXT("Amp Latch Data", 3, tas2781_latch_reg_get, NULL), +}; + static const struct snd_kcontrol_new tas2563_snd_controls[] = { SOC_SINGLE_RANGE_EXT_TLV("Speaker Digital Volume", TAS2563_DVC_LVL, 0, 0, ARRAY_SIZE(tas2563_dvc_table) - 1, 0, @@ -257,6 +958,11 @@ static const struct snd_kcontrol_new tas2563_snd_controls[] = { tas2563_dvc_tlv), }; +static const struct snd_kcontrol_new tas2563_cali_controls[] = { + SOC_SINGLE_EXT("Calibration Start", SND_SOC_NOPM, 0, 1, 0, + tasdev_nop_get, tas2563_calib_start_put), +}; + static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -274,6 +980,31 @@ static int tasdevice_set_profile_id(struct snd_kcontrol *kcontrol, return ret; } +static int tasdevice_info_active_num(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = tas_priv->ndev - 1; + + return 0; +} + +static int tasdevice_info_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = TAS2563; + uinfo->value.integer.max = TAS2781; + + return 0; +} + static int tasdevice_info_programs(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { @@ -330,6 +1061,17 @@ static int tasdevice_get_profile_id(struct snd_kcontrol *kcontrol, return 0; } +static int tasdevice_get_chip_id(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + + ucontrol->value.integer.value[0] = tas_priv->chip_id; + + return 0; +} + static int tasdevice_create_control(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *prof_ctrls; @@ -421,11 +1163,47 @@ static int tasdevice_configuration_put( return ret; } +static int tasdevice_active_num_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + struct i2c_client *clt = (struct i2c_client *)tas_priv->client; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) { + if (clt->addr == tasdev[i].dev_addr) { + ucontrol->value.integer.value[0] = i; + return 0; + } + } + + return -1; +} + +static int tasdevice_active_num_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *codec = snd_soc_kcontrol_component(kcontrol); + struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); + int dev_id = ucontrol->value.integer.value[0]; + int max = tas_priv->ndev - 1, rc; + + dev_id = clamp(dev_id, 0, max); + + guard(mutex)(&tas_priv->codec_lock); + rc = tasdev_chn_switch(tas_priv, dev_id); + + return rc; +} + static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) { struct snd_kcontrol_new *dsp_ctrls; - char *prog_name, *conf_name; - int nr_controls = 2; + char *active_dev_num, *chip_id; + char *conf_name, *prog_name; + int nr_controls = 4; int mix_index = 0; int ret; @@ -466,6 +1244,30 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_configuration_put; mix_index++; + active_dev_num = devm_kstrdup(tas_priv->dev, "Activate Tasdevice Num", + GFP_KERNEL); + if (!active_dev_num) { + ret = -ENOMEM; + goto out; + } + dsp_ctrls[mix_index].name = active_dev_num; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_active_num; + dsp_ctrls[mix_index].get = tasdevice_active_num_get; + dsp_ctrls[mix_index].put = tasdevice_active_num_put; + mix_index++; + + chip_id = devm_kstrdup(tas_priv->dev, "Tasdevice Chip Id", GFP_KERNEL); + if (!chip_id) { + ret = -ENOMEM; + goto out; + } + dsp_ctrls[mix_index].name = chip_id; + dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + dsp_ctrls[mix_index].info = tasdevice_info_chip_id; + dsp_ctrls[mix_index].get = tasdevice_get_chip_id; + mix_index++; + ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); @@ -473,6 +1275,150 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) return ret; } +static int tasdevice_create_cali_ctrls(struct tasdevice_priv *tas_priv) +{ + struct calidata *cali_data = &tas_priv->cali_data; + struct tasdevice *tasdev = tas_priv->tasdevice; + struct soc_bytes_ext *ext_cali_data; + struct snd_kcontrol_new *cali_ctrls; + unsigned int num_controls; + char *cali_name; + int rc, i; + + rc = snd_soc_add_component_controls(tas_priv->codec, + tasdevice_cali_controls, ARRAY_SIZE(tasdevice_cali_controls)); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add cali controls err rc = %d", + __func__, rc); + return rc; + } + + if (tas_priv->chip_id == TAS2781) { + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; + num_controls = ARRAY_SIZE(tas2781_cali_controls); + for (i = 0; i < tas_priv->ndev; i++) { + tasdev[i].cali_data_backup = + kmemdup(tas2781_cali_start_reg, + sizeof(tas2781_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_backup) + return -ENOMEM; + } + } else { + cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; + num_controls = ARRAY_SIZE(tas2563_cali_controls); + for (i = 0; i < tas_priv->ndev; i++) { + tasdev[i].cali_data_backup = + kmemdup(tas2563_cali_start_reg, + sizeof(tas2563_cali_start_reg), GFP_KERNEL); + if (!tasdev[i].cali_data_backup) + return -ENOMEM; + } + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls); + if (rc < 0) { + dev_err(tas_priv->dev, "%s: Add chip cali ctrls err rc = %d", + __func__, rc); + return rc; + } + + /* index for cali_ctrls */ + i = 0; + if (tas_priv->chip_id == TAS2781) + num_controls = 2; + else + num_controls = 1; + + /* + * Alloc kcontrol via devm_kzalloc, which don't manually + * free the kcontrol + */ + cali_ctrls = devm_kcalloc(tas_priv->dev, num_controls, + sizeof(cali_ctrls[0]), GFP_KERNEL); + if (!cali_ctrls) + return -ENOMEM; + + ext_cali_data = devm_kzalloc(tas_priv->dev, sizeof(*ext_cali_data), + GFP_KERNEL); + if (!ext_cali_data) + return -ENOMEM; + + cali_name = devm_kstrdup(tas_priv->dev, "Speaker Calibrated Data", + GFP_KERNEL); + if (!cali_name) + return -ENOMEM; + /* the number of calibrated data per tas2563/tas2781 */ + cali_data->cali_dat_sz_per_dev = 20; + /* + * Data structure for tas2563/tas2781 calibrated data: + * Pkg len (1 byte) + * Reg id (1 byte, constant 'r') + * book, page, register array for calibrated data (15 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info (1 byte) + * Calibrated data for Device #i (20 bytes) + * } + */ + ext_cali_data->max = tas_priv->ndev * + (cali_data->cali_dat_sz_per_dev + 1) + 1 + 15 + 1; + tas_priv->cali_data.total_sz = tas_priv->ndev * + (cali_data->cali_dat_sz_per_dev + 1); + tas_priv->cali_data.data = devm_kzalloc(tas_priv->dev, + ext_cali_data->max, GFP_KERNEL); + cali_ctrls[i].name = cali_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].get = tasdev_cali_data_get; + cali_ctrls[i].put = tasdev_cali_data_put; + cali_ctrls[i].private_value = (unsigned long)ext_cali_data; + i++; + + cali_data->data = devm_kzalloc(tas_priv->dev, cali_data->total_sz, + GFP_KERNEL); + if (!cali_data->data) + return -ENOMEM; + + if (tas_priv->chip_id == TAS2781) { + struct soc_bytes_ext *ext_cali_start; + char *cali_start_name; + + ext_cali_start = devm_kzalloc(tas_priv->dev, + sizeof(*ext_cali_start), GFP_KERNEL); + if (!ext_cali_start) + return -ENOMEM; + + cali_start_name = devm_kstrdup(tas_priv->dev, + "Calibration Start", GFP_KERNEL); + if (!cali_start_name) + return -ENOMEM; + /* + * package structure for tas2781 ftc start: + * Pkg len (1 byte) + * Reg id (1 byte, constant 'r') + * book, page, register for pilot threshold, pilot tone + * and sine gain (12 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info (1 byte) + * Sine gain for Device #i (8 bytes) + * } + */ + ext_cali_data->max = 14 + tas_priv->ndev * 9; + cali_ctrls[i].name = cali_start_name; + cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; + cali_ctrls[i].info = snd_soc_bytes_info_ext; + cali_ctrls[i].put = tas2781_calib_start_put; + cali_ctrls[i].get = tasdev_nop_get; + cali_ctrls[i].private_value = (unsigned long)ext_cali_start; + i++; + } + + rc = snd_soc_add_component_controls(tas_priv->codec, cali_ctrls, + num_controls < i ? num_controls : i); + + return rc; +} + static void tasdevice_fw_ready(const struct firmware *fmw, void *context) { @@ -519,6 +1465,12 @@ static void tasdevice_fw_ready(const struct firmware *fmw, goto out; } + ret = tasdevice_create_cali_ctrls(tas_priv); + if (ret) { + dev_err(tas_priv->dev, "cali controls error\n"); + goto out; + } + tas_priv->fw_state = TASDEVICE_DSP_FW_ALL_OK; /* If calibrated data occurs error, dsp will still works with default @@ -720,6 +1672,11 @@ static int tasdevice_codec_probe(struct snd_soc_component *codec) static void tasdevice_deinit(void *context) { struct tasdevice_priv *tas_priv = (struct tasdevice_priv *) context; + struct tasdevice *tasdev = tas_priv->tasdevice; + int i; + + for (i = 0; i < tas_priv->ndev; i++) + kfree(tasdev[i].cali_data_backup); tasdevice_config_info_remove(tas_priv); tasdevice_dsp_remove(tas_priv); -- 2.35.7

3 2

[PATCH v2 0/3] ASoC: dt-bindings: amlogic-sound-cards: document clocks property
by Neil Armstrong 02 Sep '24

02 Sep '24

Following an off-list discution with Jerome about fixing the following DTBs check errors: sound: Unevaluated properties are not allowed ('assigned-clock-parents', 'assigned-clock-rates', 'assigned-clocks' were unexpected) from schema $id: http://devicetree.org/schemas/sound/amlogic,axg-sound-card.yaml# sound: Unevaluated properties are not allowed ('assigned-clock-parents', 'assigned-clock-rates', 'assigned-clocks' were unexpected) from schema $id: http://devicetree.org/schemas/sound/amlogic,gx-sound-card.yaml# sound: 'anyOf' conditional failed, one must be fixed: 'clocks' is a required property '#clock-cells' is a required property from schema $id: http://devicetree.org/schemas/clock/clock.yaml# It has been agreed documenting the clock in the sound card is a better solution than moving them to a random clock controller or consumer node which is not related to the actual meaning of those root frequencies. The patchset adds the clocks proprty to the bindings and finally adds the properties to the DT files. Signed-off-by: Neil Armstrong <neil.armstrong(a)linaro.org> --- Changes in v2: - drop clock-names in bindings & commit messages - fix new boards - drop RFS - Link to v1: https://lore.kernel.org/r/20240614-topic-amlogic-upstream-bindings-fixes-au… --- Neil Armstrong (3): ASoC: dt-bindings: amlogic,axg-sound-card: document clocks property ASoC: dt-bindings: amlogic,gx-sound-card: document clocks property arm64: dts: amlogic: add clock and clock-names to sound cards .../devicetree/bindings/sound/amlogic,axg-sound-card.yaml | 7 +++++++ Documentation/devicetree/bindings/sound/amlogic,gx-sound-card.yaml | 7 +++++++ arch/arm64/boot/dts/amlogic/meson-axg-s400.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-fbx8am.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-radxa-zero.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-sei510.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-u200.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12a-x96-max.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-bananapi-cm4-cm4io.dts | 4 ++++ .../arm64/boot/dts/amlogic/meson-g12b-bananapi-cm4-mnt-reform2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-bananapi.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-dreambox.dtsi | 6 ++++++ arch/arm64/boot/dts/amlogic/meson-g12b-gsking-x.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-gtking-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-gtking.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-go-ultra.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-n2.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-odroid-n2l.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-radxa-zero2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-g12b-ugoos-am6.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gx-libretech-pc.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gx-p23x-q20x.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-kii-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-nanopi-k2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-nexbox-a95x.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-odroidc2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-p200.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-p201.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-vega-s95.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-wetek-hub.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxbb-wetek-play2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s805x-libretech-ac.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s805x-p241.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-khadas-vim.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-libretech-cc-v2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-libretech-cc.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-p212.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxl-s905x-vero4k.dts | 5 +++++ arch/arm64/boot/dts/amlogic/meson-gxm-khadas-vim2.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxm-nexbox-a1.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-gxm-rbox-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-khadas-vim3.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-libretech-cottonwood.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-a95xf3-air-gbit.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-a95xf3-air.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-bananapi-m2-pro.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-bananapi-m5.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-h96-max.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-odroid.dtsi | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-sei610.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-x96-air-gbit.dts | 4 ++++ arch/arm64/boot/dts/amlogic/meson-sm1-x96-air.dts | 4 ++++ 52 files changed, 217 insertions(+) --- base-commit: cb2361c78068ec47c3a6222d71f664b67665550e change-id: 20240614-topic-amlogic-upstream-bindings-fixes-audio-snd-card-2c642ccd2c6e Best regards, -- Neil Armstrong <neil.armstrong(a)linaro.org>

3 6

[tiwai-sound:for-next 12/12] sound/core/pcm_memory.c:218:29: warning: format specifies type 'size_t' (aka 'unsigned int') but the argument has type 'unsigned long'
by kernel test robot 02 Sep '24

02 Sep '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 43b42ed438bfff6bb5a51cc27a1658c03cd223fd commit: 43b42ed438bfff6bb5a51cc27a1658c03cd223fd [12/12] ALSA: pcm: Fix the previous conversion to kstrtoul() config: i386-buildonly-randconfig-002-20240902 (https://download.01.org/0day-ci/archive/20240902/202409020722.T2ymGxmq-lkp@…) compiler: clang version 18.1.5 (https://github.com/llvm/llvm-project 617a15a9eac96088ae5e9134248d8236e34b91b1) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240902/202409020722.T2ymGxmq-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409020722.T2ymGxmq-lkp@intel.com/ All warnings (new ones prefixed by >>): >> sound/core/pcm_memory.c:218:29: warning: format specifies type 'size_t' (aka 'unsigned int') but the argument has type 'unsigned long' [-Wformat] 215 | pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", | ~~~ | %lu 216 | substream->pcm->card->number, substream->pcm->device, 217 | substream->stream ? 'c' : 'p', substream->number, 218 | substream->pcm->name, size); | ^~~~ include/linux/printk.h:595:38: note: expanded from macro 'pr_debug' 595 | no_printk(KERN_DEBUG pr_fmt(fmt), ##__VA_ARGS__) | ~~~ ^~~~~~~~~~~ include/linux/printk.h:133:18: note: expanded from macro 'no_printk' 133 | _printk(fmt, ##__VA_ARGS__); \ | ~~~ ^~~~~~~~~~~ 1 warning generated. vim +218 sound/core/pcm_memory.c c7132aeb72ad11 Jaroslav Kysela 2006-10-06 174 ^1da177e4c3f41 Linus Torvalds 2005-04-16 175 /* ^1da177e4c3f41 Linus Torvalds 2005-04-16 176 * write callback for prealloc proc file ^1da177e4c3f41 Linus Torvalds 2005-04-16 177 * ^1da177e4c3f41 Linus Torvalds 2005-04-16 178 * accepts the preallocation size in kB. ^1da177e4c3f41 Linus Torvalds 2005-04-16 179 */ 877211f5e1b119 Takashi Iwai 2005-11-17 180 static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, 877211f5e1b119 Takashi Iwai 2005-11-17 181 struct snd_info_buffer *buffer) ^1da177e4c3f41 Linus Torvalds 2005-04-16 182 { 877211f5e1b119 Takashi Iwai 2005-11-17 183 struct snd_pcm_substream *substream = entry->private_data; d4cfb30fce0309 Takashi Iwai 2020-01-20 184 struct snd_card *card = substream->pcm->card; ^1da177e4c3f41 Linus Torvalds 2005-04-16 185 char line[64], str[64]; 43b42ed438bfff Takashi Iwai 2024-09-01 186 unsigned long size; ^1da177e4c3f41 Linus Torvalds 2005-04-16 187 struct snd_dma_buffer new_dmab; ^1da177e4c3f41 Linus Torvalds 2005-04-16 188 dd0da75b9a2768 Takashi Iwai 2024-02-27 189 guard(mutex)(&substream->pcm->open_mutex); ^1da177e4c3f41 Linus Torvalds 2005-04-16 190 if (substream->runtime) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 191 buffer->error = -EBUSY; dd0da75b9a2768 Takashi Iwai 2024-02-27 192 return; ^1da177e4c3f41 Linus Torvalds 2005-04-16 193 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 194 if (!snd_info_get_line(buffer, line, sizeof(line))) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 195 snd_info_get_str(str, line, sizeof(str)); 61bc4deff03318 Hongbo Li 2024-08-31 196 buffer->error = kstrtoul(str, 10, &size); 61bc4deff03318 Hongbo Li 2024-08-31 197 if (buffer->error != 0) 61bc4deff03318 Hongbo Li 2024-08-31 198 return; 61bc4deff03318 Hongbo Li 2024-08-31 199 size *= 1024; ^1da177e4c3f41 Linus Torvalds 2005-04-16 200 if ((size != 0 && size < 8192) || size > substream->dma_max) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 201 buffer->error = -EINVAL; dd0da75b9a2768 Takashi Iwai 2024-02-27 202 return; ^1da177e4c3f41 Linus Torvalds 2005-04-16 203 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 204 if (substream->dma_buffer.bytes == size) dd0da75b9a2768 Takashi Iwai 2024-02-27 205 return; ^1da177e4c3f41 Linus Torvalds 2005-04-16 206 memset(&new_dmab, 0, sizeof(new_dmab)); ^1da177e4c3f41 Linus Torvalds 2005-04-16 207 new_dmab.dev = substream->dma_buffer.dev; ^1da177e4c3f41 Linus Torvalds 2005-04-16 208 if (size > 0) { d4cfb30fce0309 Takashi Iwai 2020-01-20 209 if (do_alloc_pages(card, d4cfb30fce0309 Takashi Iwai 2020-01-20 210 substream->dma_buffer.dev.type, ^1da177e4c3f41 Linus Torvalds 2005-04-16 211 substream->dma_buffer.dev.dev, a25684a956468e Takashi Iwai 2021-10-17 212 substream->stream, ^1da177e4c3f41 Linus Torvalds 2005-04-16 213 size, &new_dmab) < 0) { ^1da177e4c3f41 Linus Torvalds 2005-04-16 214 buffer->error = -ENOMEM; dc85fc9d05d235 Amadeusz Sławiński 2021-03-18 215 pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", dc85fc9d05d235 Amadeusz Sławiński 2021-03-18 216 substream->pcm->card->number, substream->pcm->device, dc85fc9d05d235 Amadeusz Sławiński 2021-03-18 217 substream->stream ? 'c' : 'p', substream->number, dc85fc9d05d235 Amadeusz Sławiński 2021-03-18 @218 substream->pcm->name, size); dd0da75b9a2768 Takashi Iwai 2024-02-27 219 return; ^1da177e4c3f41 Linus Torvalds 2005-04-16 220 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 221 substream->buffer_bytes_max = size; ^1da177e4c3f41 Linus Torvalds 2005-04-16 222 } else { ^1da177e4c3f41 Linus Torvalds 2005-04-16 223 substream->buffer_bytes_max = UINT_MAX; ^1da177e4c3f41 Linus Torvalds 2005-04-16 224 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 225 if (substream->dma_buffer.area) d4cfb30fce0309 Takashi Iwai 2020-01-20 226 do_free_pages(card, &substream->dma_buffer); ^1da177e4c3f41 Linus Torvalds 2005-04-16 227 substream->dma_buffer = new_dmab; ^1da177e4c3f41 Linus Torvalds 2005-04-16 228 } else { ^1da177e4c3f41 Linus Torvalds 2005-04-16 229 buffer->error = -EINVAL; ^1da177e4c3f41 Linus Torvalds 2005-04-16 230 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 231 } ^1da177e4c3f41 Linus Torvalds 2005-04-16 232 :::::: The code at line 218 was first introduced by commit :::::: dc85fc9d05d23591ddfde400c817413765611ec7 ALSA: pcm: Add debug print on memory allocation failure :::::: TO: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> :::::: CC: Takashi Iwai <tiwai(a)suse.de> -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[RESEND PATCH v4] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 01 Sep '24

01 Sep '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. So far, the Software maintainer has not been confirmed. Once the maintainer was confimred, the guy will update his mail into audio converter section. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - Add Touch Screen support v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 34 +++++++++++++++++++++++++--------- 1 file changed, 25 insertions(+), 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index fe83ba7194ea..a2448dab3601 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22596,26 +22596,43 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22625,9 +22642,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

Re: [PATCH v7 0/2] ASoC: fsl-asoc-card: add S/PDIF controller support
by Shawn Guo 01 Sep '24

01 Sep '24

On Tue, Aug 20, 2024 at 01:28:25PM +0200, Elinor Montmasson wrote: > Elinor Montmasson (2): > arm64: dts: imx8m: update spdif sound card node properties > ARM: dts: imx6: update spdif sound card node properties Applied both, thanks!

1 0

[tiwai-sound:for-next 11/11] sound/core/pcm_memory.c:196:37: error: incompatible pointer types passing 'size_t *' (aka 'unsigned int *') to parameter of type 'unsigned long *'
by kernel test robot 01 Sep '24

01 Sep '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 61bc4deff033181992408f973b48fca08757d3ff commit: 61bc4deff033181992408f973b48fca08757d3ff [11/11] ALSA: pcm: replace simple_strtoul to kstrtoul config: hexagon-randconfig-002-20240901 (https://download.01.org/0day-ci/archive/20240901/202409010509.nOL6xTD7-lkp@…) compiler: clang version 20.0.0git (https://github.com/llvm/llvm-project 46fe36a4295f05d5d3731762e31fc4e6e99863e9) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240901/202409010509.nOL6xTD7-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409010509.nOL6xTD7-lkp@intel.com/ All errors (new ones prefixed by >>): In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:548:31: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 548 | val = __raw_readb(PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:561:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 561 | val = __le16_to_cpu((__le16 __force)__raw_readw(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:37:51: note: expanded from macro '__le16_to_cpu' 37 | #define __le16_to_cpu(x) ((__force __u16)(__le16)(x)) | ^ In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:574:61: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 574 | val = __le32_to_cpu((__le32 __force)__raw_readl(PCI_IOBASE + addr)); | ~~~~~~~~~~ ^ include/uapi/linux/byteorder/little_endian.h:35:51: note: expanded from macro '__le32_to_cpu' 35 | #define __le32_to_cpu(x) ((__force __u32)(__le32)(x)) | ^ In file included from sound/core/pcm_memory.c:7: In file included from include/linux/io.h:14: In file included from arch/hexagon/include/asm/io.h:328: include/asm-generic/io.h:585:33: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 585 | __raw_writeb(value, PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:595:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 595 | __raw_writew((u16 __force)cpu_to_le16(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ include/asm-generic/io.h:605:59: warning: performing pointer arithmetic on a null pointer has undefined behavior [-Wnull-pointer-arithmetic] 605 | __raw_writel((u32 __force)cpu_to_le32(value), PCI_IOBASE + addr); | ~~~~~~~~~~ ^ In file included from sound/core/pcm_memory.c:14: In file included from include/sound/pcm.h:15: In file included from include/linux/mm.h:2228: include/linux/vmstat.h:517:36: warning: arithmetic between different enumeration types ('enum node_stat_item' and 'enum lru_list') [-Wenum-enum-conversion] 517 | return node_stat_name(NR_LRU_BASE + lru) + 3; // skip "nr_" | ~~~~~~~~~~~ ^ ~~~ >> sound/core/pcm_memory.c:196:37: error: incompatible pointer types passing 'size_t *' (aka 'unsigned int *') to parameter of type 'unsigned long *' [-Werror,-Wincompatible-pointer-types] 196 | buffer->error = kstrtoul(str, 10, &size); | ^~~~~ include/linux/kstrtox.h:30:90: note: passing argument to parameter 'res' here 30 | static inline int __must_check kstrtoul(const char *s, unsigned int base, unsigned long *res) | ^ 7 warnings and 1 error generated. vim +196 sound/core/pcm_memory.c 174 175 /* 176 * write callback for prealloc proc file 177 * 178 * accepts the preallocation size in kB. 179 */ 180 static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, 181 struct snd_info_buffer *buffer) 182 { 183 struct snd_pcm_substream *substream = entry->private_data; 184 struct snd_card *card = substream->pcm->card; 185 char line[64], str[64]; 186 size_t size; 187 struct snd_dma_buffer new_dmab; 188 189 guard(mutex)(&substream->pcm->open_mutex); 190 if (substream->runtime) { 191 buffer->error = -EBUSY; 192 return; 193 } 194 if (!snd_info_get_line(buffer, line, sizeof(line))) { 195 snd_info_get_str(str, line, sizeof(str)); > 196 buffer->error = kstrtoul(str, 10, &size); 197 if (buffer->error != 0) 198 return; 199 size *= 1024; 200 if ((size != 0 && size < 8192) || size > substream->dma_max) { 201 buffer->error = -EINVAL; 202 return; 203 } 204 if (substream->dma_buffer.bytes == size) 205 return; 206 memset(&new_dmab, 0, sizeof(new_dmab)); 207 new_dmab.dev = substream->dma_buffer.dev; 208 if (size > 0) { 209 if (do_alloc_pages(card, 210 substream->dma_buffer.dev.type, 211 substream->dma_buffer.dev.dev, 212 substream->stream, 213 size, &new_dmab) < 0) { 214 buffer->error = -ENOMEM; 215 pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", 216 substream->pcm->card->number, substream->pcm->device, 217 substream->stream ? 'c' : 'p', substream->number, 218 substream->pcm->name, size); 219 return; 220 } 221 substream->buffer_bytes_max = size; 222 } else { 223 substream->buffer_bytes_max = UINT_MAX; 224 } 225 if (substream->dma_buffer.area) 226 do_free_pages(card, &substream->dma_buffer); 227 substream->dma_buffer = new_dmab; 228 } else { 229 buffer->error = -EINVAL; 230 } 231 } 232 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 11/11] sound/core/pcm_memory.c:196:51: error: passing argument 3 of 'kstrtoul' from incompatible pointer type
by kernel test robot 31 Aug '24

31 Aug '24

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 61bc4deff033181992408f973b48fca08757d3ff commit: 61bc4deff033181992408f973b48fca08757d3ff [11/11] ALSA: pcm: replace simple_strtoul to kstrtoul config: sh-allmodconfig (https://download.01.org/0day-ci/archive/20240901/202409010425.YPS7cWeJ-lkp@…) compiler: sh4-linux-gcc (GCC) 14.1.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20240901/202409010425.YPS7cWeJ-lkp@…) If you fix the issue in a separate patch/commit (i.e. not just a new version of the same patch/commit), kindly add following tags | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: https://lore.kernel.org/oe-kbuild-all/202409010425.YPS7cWeJ-lkp@intel.com/ All errors (new ones prefixed by >>): sound/core/pcm_memory.c: In function 'snd_pcm_lib_preallocate_proc_write': >> sound/core/pcm_memory.c:196:51: error: passing argument 3 of 'kstrtoul' from incompatible pointer type [-Wincompatible-pointer-types] 196 | buffer->error = kstrtoul(str, 10, &size); | ^~~~~ | | | size_t * {aka unsigned int *} In file included from include/linux/kernel.h:25, from arch/sh/include/asm/fixmap.h:13, from arch/sh/include/asm/pgtable.h:22, from include/linux/pgtable.h:6, from arch/sh/include/asm/io.h:21, from include/linux/io.h:14, from sound/core/pcm_memory.c:7: include/linux/kstrtox.h:30:90: note: expected 'long unsigned int *' but argument is of type 'size_t *' {aka 'unsigned int *'} 30 | static inline int __must_check kstrtoul(const char *s, unsigned int base, unsigned long *res) | ~~~~~~~~~~~~~~~^~~ vim +/kstrtoul +196 sound/core/pcm_memory.c 174 175 /* 176 * write callback for prealloc proc file 177 * 178 * accepts the preallocation size in kB. 179 */ 180 static void snd_pcm_lib_preallocate_proc_write(struct snd_info_entry *entry, 181 struct snd_info_buffer *buffer) 182 { 183 struct snd_pcm_substream *substream = entry->private_data; 184 struct snd_card *card = substream->pcm->card; 185 char line[64], str[64]; 186 size_t size; 187 struct snd_dma_buffer new_dmab; 188 189 guard(mutex)(&substream->pcm->open_mutex); 190 if (substream->runtime) { 191 buffer->error = -EBUSY; 192 return; 193 } 194 if (!snd_info_get_line(buffer, line, sizeof(line))) { 195 snd_info_get_str(str, line, sizeof(str)); > 196 buffer->error = kstrtoul(str, 10, &size); 197 if (buffer->error != 0) 198 return; 199 size *= 1024; 200 if ((size != 0 && size < 8192) || size > substream->dma_max) { 201 buffer->error = -EINVAL; 202 return; 203 } 204 if (substream->dma_buffer.bytes == size) 205 return; 206 memset(&new_dmab, 0, sizeof(new_dmab)); 207 new_dmab.dev = substream->dma_buffer.dev; 208 if (size > 0) { 209 if (do_alloc_pages(card, 210 substream->dma_buffer.dev.type, 211 substream->dma_buffer.dev.dev, 212 substream->stream, 213 size, &new_dmab) < 0) { 214 buffer->error = -ENOMEM; 215 pr_debug("ALSA pcmC%dD%d%c,%d:%s: cannot preallocate for size %zu\n", 216 substream->pcm->card->number, substream->pcm->device, 217 substream->stream ? 'c' : 'p', substream->number, 218 substream->pcm->name, size); 219 return; 220 } 221 substream->buffer_bytes_max = size; 222 } else { 223 substream->buffer_bytes_max = UINT_MAX; 224 } 225 if (substream->dma_buffer.area) 226 do_free_pages(card, &substream->dma_buffer); 227 substream->dma_buffer = new_dmab; 228 } else { 229 buffer->error = -EINVAL; 230 } 231 } 232 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

Fix TLV dB parser in case of used container
by GitHub pull_request - edited 31 Aug '24

31 Aug '24

alsa-project/alsa-lib pull request #409 was edited from arkq: In case when dB information does not appear as the only TLV type in the stream (it might be wrapped in a container, but the container can not have any other type), the TLV parser fails to get the dB TLV pointer. This commit fixes it by distinguishing between TLV parse error and dB information not being found in a container, so the parser can iterate over all elements in the container. Also, it fixes out-of-bounds read in case of malicious TLV record. EDIT: Other possibility to fix is to return `-ENOENT` in case where dB information was not found and handle that accordingly when iterating over items of a container. Request URL : https://github.com/alsa-project/alsa-lib/pull/409 Patch URL : https://github.com/alsa-project/alsa-lib/pull/409.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Fix TLV dB parser in case of used container
by GitHub pull_request - opened 31 Aug '24

31 Aug '24

alsa-project/alsa-lib pull request #409 was opened from arkq: In case when dB information does not appear as the only TLV type in the stream (it might be wrapped in a container, but the container can not have any other type), the TLV parser fails to get the dB TLV pointer. This commit fixes it by distinguishing between TLV parse error and dB information not being found in a container, so the parser can iterate over all elements in the container. Also, it fixes out-of-bounds read in case of malicious TLV record. Request URL : https://github.com/alsa-project/alsa-lib/pull/409 Patch URL : https://github.com/alsa-project/alsa-lib/pull/409.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] ASoC: codecs: Switch to use dev_err_probe()
by Yang Ruibin 30 Aug '24

30 Aug '24

Using dev_err_probe() instead of dev_err() in probe() simplifies the error path and standardizes the format of the error code. Signed-off-by: Yang Ruibin <11162571(a)vivo.com> --- sound/soc/codecs/ad1980.c | 8 +++----- sound/soc/codecs/adau1701.c | 19 +++++++------------ sound/soc/codecs/ssm2602.c | 6 ++---- 3 files changed, 12 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 3c1ae13c1..d2f679d9d 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -237,11 +237,9 @@ static int ad1980_soc_probe(struct snd_soc_component *component) u16 ext_status; ac97 = snd_soc_new_ac97_component(component, 0, 0); - if (IS_ERR(ac97)) { - ret = PTR_ERR(ac97); - dev_err(component->dev, "Failed to register AC97 component: %d\n", ret); - return ret; - } + if (IS_ERR(ac97)) + return dev_err_probe(component->dev, PTR_ERR(ac97), + "Failed to register AC97 component\n"); regmap = regmap_init_ac97(ac97, &ad1980_regmap_config); if (IS_ERR(regmap)) { diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 8bd6067df..f4c574447 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -677,10 +677,9 @@ static int adau1701_probe(struct snd_soc_component *component) ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(component->dev, "Failed to enable regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(component->dev, ret, + "Failed to enable regulators\n"); /* * Let the pll_clkdiv variable default to something that won't happen @@ -798,17 +797,13 @@ static int adau1701_i2c_probe(struct i2c_client *client) ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(dev, "Failed to get regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(dev, ret, "Failed to get regulators\n"); ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); - if (ret < 0) { - dev_err(dev, "Failed to enable regulators: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(dev, ret, "Failed to enable regulators\n"); adau1701->client = client; adau1701->regmap = devm_regmap_init(dev, NULL, client, diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c29324403..153eb55a3 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -605,10 +605,8 @@ static int ssm260x_component_probe(struct snd_soc_component *component) int ret; ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); - if (ret < 0) { - dev_err(component->dev, "Failed to issue reset: %d\n", ret); - return ret; - } + if (ret < 0) + return dev_err_probe(component->dev, ret, "Failed to issue reset\n"); regmap_register_patch(ssm2602->regmap, ssm2602_patch, ARRAY_SIZE(ssm2602_patch)); -- 2.34.1

2 1

[PATCH v2] ASoC: codecs: lpass-va-macro: set the default codec version for sm8250
by srinivas.kandagatla＠linaro.org 30 Aug '24

30 Aug '24

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> sm8250 and sc7280 have lpass codec version 1.0, as these are very old platforms, they do not have a reliable way to get the codec version from core_id registers. On codec versions below 2.0, even though the core_id registers are available to read, the values of these registers are not unique to be able to determine the version of the codec dynamically. Add the version info into of_data, so that driver does not need to use core_id registers to get version number for such situations. Fixes: 378918d59181 ("ASoC: codecs: lpass-macro: add helpers to get codec version") Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - updated commit text to add more details sound/soc/codecs/lpass-va-macro.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 8454193ed22a..e95d1f29ef18 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -228,11 +228,13 @@ struct va_macro { struct va_macro_data { bool has_swr_master; bool has_npl_clk; + int version; }; static const struct va_macro_data sm8250_va_data = { .has_swr_master = false, .has_npl_clk = false, + .version = LPASS_CODEC_VERSION_1_0, }; static const struct va_macro_data sm8450_va_data = { @@ -1587,7 +1589,14 @@ static int va_macro_probe(struct platform_device *pdev) goto err_npl; } - va_macro_set_lpass_codec_version(va); + /** + * old version of codecs do not have a reliable way to determine the + * version from registers, get them from soc specific data + */ + if (data->version) + lpass_macro_set_codec_version(data->version); + else /* read version from register */ + va_macro_set_lpass_codec_version(va); if (va->has_swr_master) { /* Set default CLK div to 1 */ -- 2.25.1

5 5

[PATCH v1] ASoC: fsl_ssi: Simplify with dev_err_probe()
by Shen Lichuan 30 Aug '24

30 Aug '24

Use dev_err_probe() to simplify the error path and unify a message template. Using this helper is totally fine even if err is known to never be -EPROBE_DEFER. The benefit compared to a normal dev_err() is the standardized format of the error code, it being emitted symbolically and the fact that the error code is returned which allows more compact error paths. Signed-off-by: Shen Lichuan <shenlichuan(a)vivo.com> --- sound/soc/fsl/fsl_ssi.c | 33 +++++++++++++++------------------ 1 file changed, 15 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c4c1d9c44056..51927f89241f 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1354,19 +1354,16 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, ssi->clk = devm_clk_get(dev, "ipg"); else ssi->clk = devm_clk_get(dev, NULL); - if (IS_ERR(ssi->clk)) { - ret = PTR_ERR(ssi->clk); - dev_err(dev, "failed to get clock: %d\n", ret); - return ret; - } + if (IS_ERR(ssi->clk)) + return dev_err_probe(dev, PTR_ERR(ssi->clk), + "failed to get clock\n"); /* Enable the clock since regmap will not handle it in this case */ if (!ssi->has_ipg_clk_name) { ret = clk_prepare_enable(ssi->clk); - if (ret) { - dev_err(dev, "clk_prepare_enable failed: %d\n", ret); - return ret; - } + if (ret) + return dev_err_probe(dev, ret, + "clk_prepare_enable failed\n"); } /* Do not error out for consumer cases that live without a baud clock */ @@ -1552,10 +1549,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) &regconfig); else ssi->regs = devm_regmap_init_mmio(dev, iomem, &regconfig); - if (IS_ERR(ssi->regs)) { - dev_err(dev, "failed to init register map\n"); - return PTR_ERR(ssi->regs); - } + if (IS_ERR(ssi->regs)) + return dev_err_probe(dev, PTR_ERR(ssi->regs), + "failed to init register map\n"); ssi->irq = platform_get_irq(pdev, 0); if (ssi->irq < 0) @@ -1607,7 +1603,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) mutex_init(&ssi->ac97_reg_lock); ret = snd_soc_set_ac97_ops_of_reset(&fsl_ssi_ac97_ops, pdev); if (ret) { - dev_err(dev, "failed to set AC'97 ops\n"); + dev_err_probe(dev, ret, "failed to set AC'97 ops\n"); goto error_ac97_ops; } } @@ -1615,7 +1611,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = devm_snd_soc_register_component(dev, &fsl_ssi_component, &ssi->cpu_dai_drv, 1); if (ret) { - dev_err(dev, "failed to register DAI: %d\n", ret); + dev_err_probe(dev, ret, "failed to register DAI\n"); goto error_asoc_register; } @@ -1623,7 +1619,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) ret = devm_request_irq(dev, ssi->irq, fsl_ssi_isr, 0, dev_name(dev), ssi); if (ret < 0) { - dev_err(dev, "failed to claim irq %u\n", ssi->irq); + dev_err_probe(dev, ret, "failed to claim irq %u\n", + ssi->irq); goto error_asoc_register; } } @@ -1649,8 +1646,8 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi->card_name, ssi->card_idx, NULL, 0); if (IS_ERR(ssi->card_pdev)) { ret = PTR_ERR(ssi->card_pdev); - dev_err(dev, "failed to register %s: %d\n", - ssi->card_name, ret); + dev_err_probe(dev, ret, "failed to register %s\n", + ssi->card_name); goto error_sound_card; } } -- 2.17.1

1 0

ALSA question
by Hai Nguyen 30 Aug '24

30 Aug '24

Hi, I have a C++ audio application that plays back audio data on the sound card, Digigram VX882e. https://www.digigram.com/support/vx882e-support/ The application runs on HPE ProLiant DL380 Gen9 Server. HPE ProLiant DL380 Gen9 Server It has worked well under RHEL 7 (kernel 3.10) and RHEL 8 (kernel 4.x). Asthe OS is upgraded to RHEL 9 (kernel 5.14), there is an buffer underrun. When investigating the issue, I run “speaker-test” utility. It fails with the error below. I'm just wonder if this has something to do with the changes in ALSA driver from kernel 4.x to kernel 5.14? If not, what could cause the buffer underrun issue? Your advice/help is highly appreciated, Thanks, Kevin Nguyen sudo speaker-test -D plughw:VX882e0 -c 2 speaker-test 1.2.9 Playback device is plughw:VX882e0 Stream parameters are 48000Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 8192 Period size range from 64 to 4096 Using max buffer size 8192 Periods = 4 was set period_size = 2048 was set buffer_size = 8192 0 - Front Left Write error: -5,Input/outputerror xrun_recovery failed:-5,Input/output error Transfer failed: Input/outputerror The Digigram card is detected with “aplay -l” command. sudo aplay -l **** List of PLAYBACK Hardware Devices **** card 0: VX882e0 [Digigram [PCM #0]], device 0: pcxhr 0[pcxhr 0] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 1: VX882e1 [Digigram [PCM #1]], device 0: pcxhr 1[pcxhr 1] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 2: VX882e2 [Digigram [PCM #2]], device 0: pcxhr 2 [pcxhr2] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 3: VX882e3 [Digigram [PCM #3]], device 0: pcxhr 3[pcxhr 3] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3

2 1

[PATCH v3] ASoC: codecs: wsa884x: Implement temperature reading and hwmon
by Krzysztof Kozlowski 29 Aug '24

29 Aug '24

Read temperature of the speaker and expose it via hwmon interface, which will be later used during calibration of speaker protection algorithms. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v3: 1. Replace Kconfig HWMON dependency in favor of IS_REACHABLE(CONFIG_HWMON) which solves Kernel Test Robot warning for !HWMON and HWMON=y && WSA884x=m Changes in v2: 1. Add missing dependency on HWMON for y!=m builds (Kernel Test Robot report: undefined reference to `devm_hwmon_device_register_with_info'). --- sound/soc/codecs/wsa884x.c | 201 +++++++++++++++++++++++++++++++++++++ 1 file changed, 201 insertions(+) diff --git a/sound/soc/codecs/wsa884x.c b/sound/soc/codecs/wsa884x.c index 8db1380d1f10..86df5152c547 100644 --- a/sound/soc/codecs/wsa884x.c +++ b/sound/soc/codecs/wsa884x.c @@ -5,11 +5,14 @@ */ #include <linux/bitfield.h> +#include <linux/cleanup.h> #include <linux/device.h> #include <linux/gpio/consumer.h> +#include <linux/hwmon.h> #include <linux/init.h> #include <linux/kernel.h> #include <linux/module.h> +#include <linux/mutex.h> #include <linux/pm_runtime.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> @@ -301,8 +304,28 @@ #define WSA884X_PA_FSM_MSK1 (WSA884X_DIG_CTRL0_BASE + 0x3b) #define WSA884X_PA_FSM_BYP_CTL (WSA884X_DIG_CTRL0_BASE + 0x3c) #define WSA884X_PA_FSM_BYP0 (WSA884X_DIG_CTRL0_BASE + 0x3d) +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK 0x01 +#define WSA884X_PA_FSM_BYP0_DC_CAL_EN_SHIFT 0 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK 0x02 +#define WSA884X_PA_FSM_BYP0_CLK_WD_EN_SHIFT 1 +#define WSA884X_PA_FSM_BYP0_BG_EN_MASK 0x04 +#define WSA884X_PA_FSM_BYP0_BG_EN_SHIFT 2 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_MASK 0x08 +#define WSA884X_PA_FSM_BYP0_BOOST_EN_SHIFT 3 +#define WSA884X_PA_FSM_BYP0_PA_EN_MASK 0x10 +#define WSA884X_PA_FSM_BYP0_PA_EN_SHIFT 4 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK 0x20 +#define WSA884X_PA_FSM_BYP0_D_UNMUTE_SHIFT 5 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK 0x40 +#define WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_SHIFT 6 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_MASK 0x80 +#define WSA884X_PA_FSM_BYP0_TSADC_EN_SHIFT 7 #define WSA884X_PA_FSM_BYP1 (WSA884X_DIG_CTRL0_BASE + 0x3e) #define WSA884X_TADC_VALUE_CTL (WSA884X_DIG_CTRL0_BASE + 0x50) +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK 0x01 +#define WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_SHIFT 0 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_MASK 0x02 +#define WSA884X_TADC_VALUE_CTL_VBAT_VALUE_RD_EN_SHIFT 1 #define WSA884X_TEMP_DETECT_CTL (WSA884X_DIG_CTRL0_BASE + 0x51) #define WSA884X_TEMP_DIN_MSB (WSA884X_DIG_CTRL0_BASE + 0x52) #define WSA884X_TEMP_DIN_LSB (WSA884X_DIG_CTRL0_BASE + 0x53) @@ -691,6 +714,17 @@ SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) +/* Two-point trimming for temperature calibration */ +#define WSA884X_T1_TEMP -10L +#define WSA884X_T2_TEMP 150L + +/* + * Device will report senseless data in many cases, so discard any measurements + * outside of valid range. + */ +#define WSA884X_LOW_TEMP_THRESHOLD 5 +#define WSA884X_HIGH_TEMP_THRESHOLD 45 + struct wsa884x_priv { struct regmap *regmap; struct device *dev; @@ -706,6 +740,13 @@ struct wsa884x_priv { int active_ports; int dev_mode; bool hw_init; + /* + * Protects temperature reading code (related to speaker protection) and + * fields: temperature and pa_on. + */ + struct mutex sp_lock; + unsigned int temperature; + bool pa_on; }; enum { @@ -1660,6 +1701,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_POST_PMU: + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = true; + mutex_unlock(&wsa884x->sp_lock); + wsa884x_spkr_post_pmu(component, wsa884x); snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, @@ -1671,6 +1716,10 @@ static int wsa884x_spkr_event(struct snd_soc_dapm_widget *w, snd_soc_component_write_field(component, WSA884X_PDM_WD_CTL, WSA884X_PDM_WD_CTL_PDM_WD_EN_MASK, 0x0); + + mutex_lock(&wsa884x->sp_lock); + wsa884x->pa_on = false; + mutex_unlock(&wsa884x->sp_lock); break; } @@ -1810,6 +1859,144 @@ static struct snd_soc_dai_driver wsa884x_dais[] = { }, }; +static int wsa884x_get_temp(struct wsa884x_priv *wsa884x, long *temp) +{ + unsigned int d1_msb = 0, d1_lsb = 0, d2_msb = 0, d2_lsb = 0; + unsigned int dmeas_msb = 0, dmeas_lsb = 0; + int d1, d2, dmeas; + unsigned int mask; + long val; + int ret; + + guard(mutex)(&wsa884x->sp_lock); + + if (wsa884x->pa_on) { + /* + * Reading temperature is possible only when Power Amplifier is + * off. Report last cached data. + */ + *temp = wsa884x->temperature; + return 0; + } + + ret = pm_runtime_resume_and_get(wsa884x->dev); + if (ret < 0) + return ret; + + mask = WSA884X_PA_FSM_BYP0_DC_CAL_EN_MASK | + WSA884X_PA_FSM_BYP0_CLK_WD_EN_MASK | + WSA884X_PA_FSM_BYP0_BG_EN_MASK | + WSA884X_PA_FSM_BYP0_D_UNMUTE_MASK | + WSA884X_PA_FSM_BYP0_SPKR_PROT_EN_MASK | + WSA884X_PA_FSM_BYP0_TSADC_EN_MASK; + /* + * Here and further do not care about read or update failures. + * For example, before turning on Power Amplifier for the first + * time, reading WSA884X_TEMP_DIN_MSB will always return 0. + * Instead, check if returned value is within reasonable + * thresholds. + */ + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, mask); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x0)); + + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_MSB, &dmeas_msb); + regmap_read(wsa884x->regmap, WSA884X_TEMP_DIN_LSB, &dmeas_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_TADC_VALUE_CTL, + WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, + FIELD_PREP(WSA884X_TADC_VALUE_CTL_TEMP_VALUE_RD_EN_MASK, 0x1)); + + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_1, &d1_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_2, &d1_lsb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_3, &d2_msb); + regmap_read(wsa884x->regmap, WSA884X_OTP_REG_4, &d2_lsb); + + regmap_update_bits(wsa884x->regmap, WSA884X_PA_FSM_BYP0, mask, 0x0); + + dmeas = (((dmeas_msb & 0xff) << 0x8) | (dmeas_lsb & 0xff)) >> 0x6; + d1 = (((d1_msb & 0xff) << 0x8) | (d1_lsb & 0xff)) >> 0x6; + d2 = (((d2_msb & 0xff) << 0x8) | (d2_lsb & 0xff)) >> 0x6; + + if (d1 == d2) { + /* Incorrect data in OTP? */ + ret = -EINVAL; + goto out; + } + + val = WSA884X_T1_TEMP + (((dmeas - d1) * (WSA884X_T2_TEMP - WSA884X_T1_TEMP))/(d2 - d1)); + + dev_dbg(wsa884x->dev, "Measured temp %ld (dmeas=%d, d1=%d, d2=%d)\n", + val, dmeas, d1, d2); + + if ((val > WSA884X_LOW_TEMP_THRESHOLD) && + (val < WSA884X_HIGH_TEMP_THRESHOLD)) { + wsa884x->temperature = val; + *temp = val; + ret = 0; + } else { + ret = -EAGAIN; + } + +out: + pm_runtime_mark_last_busy(wsa884x->dev); + pm_runtime_put_autosuspend(wsa884x->dev); + + return ret; +} + +static umode_t wsa884x_hwmon_is_visible(const void *data, + enum hwmon_sensor_types type, u32 attr, + int channel) +{ + if (type != hwmon_temp) + return 0; + + switch (attr) { + case hwmon_temp_input: + return 0444; + default: + break; + } + + return 0; +} + +static int wsa884x_hwmon_read(struct device *dev, + enum hwmon_sensor_types type, + u32 attr, int channel, long *temp) +{ + int ret; + + switch (attr) { + case hwmon_temp_input: + ret = wsa884x_get_temp(dev_get_drvdata(dev), temp); + break; + default: + ret = -EOPNOTSUPP; + break; + } + + return ret; +} + +static const struct hwmon_channel_info *const wsa884x_hwmon_info[] = { + HWMON_CHANNEL_INFO(temp, HWMON_T_INPUT), + NULL +}; + +static const struct hwmon_ops wsa884x_hwmon_ops = { + .is_visible = wsa884x_hwmon_is_visible, + .read = wsa884x_hwmon_read, +}; + +static const struct hwmon_chip_info wsa884x_hwmon_chip_info = { + .ops = &wsa884x_hwmon_ops, + .info = wsa884x_hwmon_info, +}; + static void wsa884x_reset_powerdown(void *data) { struct wsa884x_priv *wsa884x = data; @@ -1866,6 +2053,8 @@ static int wsa884x_probe(struct sdw_slave *pdev, if (!wsa884x) return -ENOMEM; + mutex_init(&wsa884x->sp_lock); + for (i = 0; i < WSA884X_SUPPLIES_NUM; i++) wsa884x->supplies[i].supply = wsa884x_supply_name[i]; @@ -1923,6 +2112,18 @@ static int wsa884x_probe(struct sdw_slave *pdev, regcache_cache_only(wsa884x->regmap, true); wsa884x->hw_init = true; + if (IS_REACHABLE(CONFIG_HWMON)) { + struct device *hwmon; + + hwmon = devm_hwmon_device_register_with_info(dev, "wsa884x", + wsa884x, + &wsa884x_hwmon_chip_info, + NULL); + if (IS_ERR(hwmon)) + return dev_err_probe(dev, PTR_ERR(hwmon), + "Failed to register hwmon sensor\n"); + } + pm_runtime_set_autosuspend_delay(dev, 3000); pm_runtime_use_autosuspend(dev); pm_runtime_mark_last_busy(dev); -- 2.43.0

2 1

Re: [PATCH] ALSA: hda/realtek: Autodetect Cirrus Logic companion amplifier bindings
by Takashi Iwai 29 Aug '24

29 Aug '24

On Thu, 29 Aug 2024 18:11:14 +0200, Simon Trimmer wrote: > > We do not need model specific HDA quirks to construct the component > binding details of Cirrus Logic companion amplifiers as this information > is already present in the ACPI. > > Quirks are then only required for special workarounds not described in > the ACPI such as internal configuration of the Realtek codecs. > > The codec pointer is now initialized in hda_component_manager_init() so > that we can detect when companion amplifiers are found in the ACPI but > the SSID invokes a quirk that also attempts to create the component > binding. > > Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> > --- > v1 - This patch applies to for-next with this additional patch: > b1faff28a2bd ("ALSA: hda: hda_component: Fix mutex crash if nothing ever binds") > which is currently only in the for-linus. OK, I now merged for-linus to for-next, and applied yours. > We're looking to do the same for the CS35L41 devices, but that will > come later because there are many more quirks to work through. That's understandable. Thanks! Takashi

1 0

[PATCH] ASoC: mediatek: mt8188-mt6359: Modify key
by Zhang Yi 29 Aug '24

29 Aug '24

In order to get the correct keys when using the ES8326.We will associate SND_JACK_BTN_1 to KEY_VOLUMEUP and SND_JACK_BTN_2 to KEY_VOLUMEDOWN when the ES8326 flag is recognized. Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 17 +++++++++++++---- 1 file changed, 13 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8188/mt8188-mt6359.c b/sound/soc/mediatek/mt8188/mt8188-mt6359.c index eba6f4c445ff..08ae962afeb9 100644 --- a/sound/soc/mediatek/mt8188/mt8188-mt6359.c +++ b/sound/soc/mediatek/mt8188/mt8188-mt6359.c @@ -734,6 +734,7 @@ static int mt8188_headset_codec_init(struct snd_soc_pcm_runtime *rtd) struct mtk_soc_card_data *soc_card_data = snd_soc_card_get_drvdata(rtd->card); struct snd_soc_jack *jack = &soc_card_data->card_data->jacks[MT8188_JACK_HEADSET]; struct snd_soc_component *component = snd_soc_rtd_to_codec(rtd, 0)->component; + struct mtk_platform_card_data *card_data = soc_card_data->card_data; int ret; ret = snd_soc_dapm_new_controls(&card->dapm, mt8188_nau8825_widgets, @@ -762,10 +763,18 @@ static int mt8188_headset_codec_init(struct snd_soc_pcm_runtime *rtd) return ret; } - snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); - snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); - snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); - snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + if (card_data->flags & ES8326_HS_PRESENT) { + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + } else { + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + } + ret = snd_soc_component_set_jack(component, jack, NULL); if (ret) { -- 2.17.1

2 1

[PATCH v1] ASoC: codecs: Simplify with dev_err_probe()
by Yang Ruibin 29 Aug '24

29 Aug '24

Using dev_err_probe() instead of dev_err() in probe() simplifies the error path and standardizes the format of the error code. Signed-off-by: Yang Ruibin <11162571(a)vivo.com> --- sound/soc/codecs/ad1980.c | 4 ++-- sound/soc/codecs/adau1701.c | 12 ++++++------ sound/soc/codecs/ssm2602.c | 4 ++-- 3 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c index 3c1ae13c1..25fd8bc4f 100644 --- a/sound/soc/codecs/ad1980.c +++ b/sound/soc/codecs/ad1980.c @@ -239,8 +239,8 @@ static int ad1980_soc_probe(struct snd_soc_component *component) ac97 = snd_soc_new_ac97_component(component, 0, 0); if (IS_ERR(ac97)) { ret = PTR_ERR(ac97); - dev_err(component->dev, "Failed to register AC97 component: %d\n", ret); - return ret; + return dev_err_probe(component->dev, ret, + "Failed to register AC97 component: %d\n", ret); } regmap = regmap_init_ac97(ac97, &ad1980_regmap_config); diff --git a/sound/soc/codecs/adau1701.c b/sound/soc/codecs/adau1701.c index 8bd6067df..9fe18b22b 100644 --- a/sound/soc/codecs/adau1701.c +++ b/sound/soc/codecs/adau1701.c @@ -678,8 +678,8 @@ static int adau1701_probe(struct snd_soc_component *component) ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); if (ret < 0) { - dev_err(component->dev, "Failed to enable regulators: %d\n", ret); - return ret; + return dev_err_probe(component->dev, ret, + "Failed to enable regulators: %d\n", ret); } /* @@ -799,15 +799,15 @@ static int adau1701_i2c_probe(struct i2c_client *client) ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(adau1701->supplies), adau1701->supplies); if (ret < 0) { - dev_err(dev, "Failed to get regulators: %d\n", ret); - return ret; + return dev_err_probe(dev, ret, + "Failed to get regulators: %d\n", ret); } ret = regulator_bulk_enable(ARRAY_SIZE(adau1701->supplies), adau1701->supplies); if (ret < 0) { - dev_err(dev, "Failed to enable regulators: %d\n", ret); - return ret; + return dev_err_probe(dev, ret, + "Failed to enable regulators: %d\n", ret); } adau1701->client = client; diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index c29324403..727bcccad 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -606,8 +606,8 @@ static int ssm260x_component_probe(struct snd_soc_component *component) ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); if (ret < 0) { - dev_err(component->dev, "Failed to issue reset: %d\n", ret); - return ret; + return dev_err_probe(component->dev, ret, + "Failed to issue reset: %d\n", ret); } regmap_register_patch(ssm2602->regmap, ssm2602_patch, -- 2.34.1

1 1

[PATCH -next] aoa: core: Using helper function for_each_child_of_node()
by Lin Ruifeng 29 Aug '24

29 Aug '24

Helper function for_each_child_of_node() can iterate through the DT node, so we don't need to use while loop. Signed-off-by: Lin Ruifeng <linruifeng4(a)huawei.com> --- sound/aoa/core/gpio-feature.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c index 39bb409b27f6..01060f7c3862 100644 --- a/sound/aoa/core/gpio-feature.c +++ b/sound/aoa/core/gpio-feature.c @@ -72,7 +72,7 @@ static struct device_node *get_gpio(char *name, gpio = of_find_node_by_name(NULL, "gpio"); if (!gpio) return NULL; - while ((np = of_get_next_child(gpio, np))) { + for_each_child_of_node(gpio, np) { audio_gpio = of_get_property(np, "audio-gpio", NULL); if (!audio_gpio) continue; -- 2.17.1

1 0

[PATCH 0/2] Add a driver for the Iron Device SMA1307 Amp
by Kiseok Jo 29 Aug '24

29 Aug '24

This adds basic audio support for the Iron Device SMA1307 amplifier Kiseok Jo (2): ASoC: sma1307: Add driver for Iron Device SMA1307 ASoC: sma1307: Add bindings for Iron Device SMA1307 amplifier .../bindings/sound/irondevice,sma1307.yaml | 67 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sma1307.c | 2630 +++++++++++++++++ sound/soc/codecs/sma1307.h | 456 +++ 5 files changed, 3163 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/irondevice,sma1307.yaml create mode 100644 sound/soc/codecs/sma1307.c create mode 100644 sound/soc/codecs/sma1307.h -- 2.39.2

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[PATCH v25 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 28 Aug '24

28 Aug '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v25: - Cleanups on typos mentioned within the xHCI layers - Modified the xHCI interrupter search if clients specify interrupter index - Moved mixer_usb_offload into its own module, so that other vendor offload USB modules can utilize it also. - Added support for USB audio devices that may have multiple PCM streams, as previous implementation only assumed a single PCM device. SOC USB will be able to handle an array of PCM indexes supported by the USB audio device. - Added some additional checks in the QC USB offload driver to check that device has at least one playback stream before allowing to bind - Reordered DT bindings to fix the error found by Rob's bot. The patch that added USB_RX was after the example was updated. - Updated comments within SOC USB to clarify terminology and to keep it consistent - Added SND_USB_JACK type for notifying of USB device audio connections Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (31): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: Allow for secondary interrupter to set IMOD usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ALSA: Add USB audio device jack type ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Check for support for requested audio format ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Add PCM format check API for USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 429 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 34 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 50 +- drivers/usb/host/xhci-sideband.c | 424 ++++ drivers/usb/host/xhci.c | 43 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/mod_devicetable.h | 2 +- include/linux/usb/xhci-sideband.h | 68 + include/sound/jack.h | 4 +- include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 134 ++ include/uapi/linux/input-event-codes.h | 3 +- sound/core/jack.c | 6 +- sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 389 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 337 +++ sound/usb/Kconfig | 26 + sound/usb/Makefile | 4 +- sound/usb/card.c | 109 + sound/usb/card.h | 16 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/mixer_usb_offload.c | 102 + sound/usb/mixer_usb_offload.h | 17 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1964 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 53 files changed, 5827 insertions(+), 53 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/mixer_usb_offload.c create mode 100644 sound/usb/mixer_usb_offload.h create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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[PATCH v2-resend 0/2] ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function
by Kuninori Morimoto 28 Aug '24

28 Aug '24

Hi Mark This is resend patch-set No driver is calling snd_soc_dpcm_can_be_xxx() functions. We don't need to have EXPORT_SYMBOL_GPL() for them. This patch-set makes it static function. v1 -> v2 - rebased to latest branch (for-6.12) Kuninori Morimoto (2): ASoC: soc-pcm: move snd_soc_dpcm_can_be_xxx() to top ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function include/sound/soc-dpcm.h | 18 ---- sound/soc/soc-pcm.c | 201 +++++++++++++++++++-------------------- 2 files changed, 99 insertions(+), 120 deletions(-) -- 2.43.0

2 3

ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function
by Kuninori Morimoto 28 Aug '24

28 Aug '24

Hi Mark No driver is calling snd_soc_dpcm_can_be_xxx() functions. We don't need to have EXPORT_SYMBOL_GPL() for them. This patch-set makes it static function. Thank you for your help !! Best regards --- Kuninori Morimoto

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[PATCH 00/18] ASoC: Intel: boards: updates for 6.11
by Pierre-Louis Bossart 28 Aug '24

28 Aug '24

More simplifications from Brent Lu for Chromebooks, Bard Liao for SoundWire, and support for new Dell SKUs w/ Cirrus Logic codecs added by Charles Keepax. Bard Liao (7): ASoC: Intel: sof_sdw_cs42l42: use dai parameter ASoC: Intel: sof_sdw_rt711: use dai parameter ASoC: Intel: sof_sdw_rt5682: use dai parameter ASoC: Intel: sof_sdw_rt700: use dai parameter ASoC: Intel: sof_sdw_rt_dmic: use from dai parameter ASoC: Intel: sof_sdw_rt_sdca_jack_common: use dai parameter ASoC: Intel: sof_sdw: remove get_codec_dai_by_name Brent Lu (7): ASoC: SOF: sof-audio: rename dai clock setting query function ASoC: SOF: sof-audio: add sof_dai_get_tdm_slots function ASoC: SOF: ipc3-topology: support tdm slot number query ASoC: SOF: ipc4-topology: support tdm slot number query ASoC: Intel: maxim-common: rewrite max_98373_hw_params function ASoC: Intel: sof_da7219: remove local max98373 ops ASoC: Intel: sof_da7219: disable max98373 speaker pins in late_probe Charles Keepax (4): ASoC: Intel: sof_sdw: Add missing controls for cs42l43/cs35l56 ASoC: Intel: soc-acpi: Add match entries for some cs42l43 laptops ASoC: Intel: sof_sdw: Add quirks for some new Dell laptops ASoC: Intel: sof_sdw: Add quirks from some new Dell laptops include/sound/sof.h | 1 + sound/soc/intel/boards/sof_board_helpers.h | 2 - sound/soc/intel/boards/sof_da7219.c | 57 ++------ sound/soc/intel/boards/sof_maxim_common.c | 84 ++++++++++-- sound/soc/intel/boards/sof_sdw.c | 128 +++++++++++++++--- sound/soc/intel/boards/sof_sdw_common.h | 4 - sound/soc/intel/boards/sof_sdw_cs42l42.c | 11 +- sound/soc/intel/boards/sof_sdw_rt5682.c | 11 +- sound/soc/intel/boards/sof_sdw_rt700.c | 11 +- sound/soc/intel/boards/sof_sdw_rt711.c | 11 +- sound/soc/intel/boards/sof_sdw_rt_dmic.c | 14 +- .../boards/sof_sdw_rt_sdca_jack_common.c | 11 +- .../intel/common/soc-acpi-intel-arl-match.c | 50 +++++++ .../intel/common/soc-acpi-intel-rpl-match.c | 50 +++++++ sound/soc/sof/ipc3-topology.c | 14 +- sound/soc/sof/ipc4-topology.c | 14 +- sound/soc/sof/sof-audio.c | 20 ++- sound/soc/sof/sof-audio.h | 9 +- 18 files changed, 338 insertions(+), 164 deletions(-) -- 2.43.0

4 21

[PATCH v7 0/4] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 28 Aug '24

28 Aug '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is v7 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.12 or v6.13 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871q5tnuok.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87bk4oqerx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/8734pctmte.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0ctwzr4.wl-kuninori.morimoto.gx@renesas.com v6 -> v7 - based on latest ASoC branch (for-6.12) v5 -> v6 - based on latest ASoC branch v4 -> v5 - typo fix: limition -> limitation - typo fix: catpure -> capture - include Intel patch v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (2): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: use playback/capture_only flags include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 13 files changed, 47 insertions(+), 88 deletions(-) -- 2.43.0

2 7

[PATCH v6 0/4] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 28 Aug '24

28 Aug '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is resend of v6 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.12 or v6.13 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871q5tnuok.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87bk4oqerx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/8734pctmte.wl-kuninori.morimoto.gx@renesas.com v5 -> v6 - based on latest ASoC branch v4 -> v5 - typo fix: limition -> limitation - typo fix: catpure -> capture - include Intel patch v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (2): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: use playback/capture_only flags include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 13 files changed, 47 insertions(+), 88 deletions(-) -- 2.43.0

2 5

[PATCH v5 0/4] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 28 Aug '24

28 Aug '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is v5 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.11 or v6.12 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871q5tnuok.wl-kuninori.morimoto.gx@renesas.com v4 -> v5 - typo fix: limition -> limitation - typo fix: catpure -> capture - include Intel patch v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (2): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() Pierre-Louis Bossart (1): ASoC: Intel: sof_sdw: use playback/capture_only flags include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/intel/boards/sof_sdw.c | 4 +- sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 13 files changed, 47 insertions(+), 88 deletions(-) -- 2.43.0

2 9

Specifying supported hw params (channels, format, etc.) by the plug plugin in alsa config?
by Pavel Hofman 27 Aug '24

27 Aug '24

Hi, Please is it possible to restrict the hw params space for the plug plugin by some alsa configuration, so that it reports/accepts only a specific number of channels/specific sample format? I was not able to figure it out, I apologize if it's trivial :-) Thanks a lot. With regards, Pavel.

2 4

[PATCH -next 0/4] ASoC: fix module autoloading
by Liao Chen 27 Aug '24

27 Aug '24

Hi all, This patchset aims to enable autoloading of some use modules. By registering MDT, the kernel is allowed to automatically bind modules to devices that match the specified compatible strings. Liao Chen (4): ASoC: intel: fix module autoloading ASoC: google: fix module autoloading ASoC: tda7419: fix module autoloading ASoC: fix module autoloading sound/soc/codecs/chv3-codec.c | 1 + sound/soc/codecs/tda7419.c | 1 + sound/soc/google/chv3-i2s.c | 1 + sound/soc/intel/keembay/kmb_platform.c | 1 + 4 files changed, 4 insertions(+) -- 2.34.1

2 5

[PATCH v2] ASoC: tas2781: replace devm_kzalloc and scnprintf with devm_kstrdup
by Shenghao Ding 26 Aug '24

26 Aug '24

Replace devm_kzalloc and scnprintf with devm_kstrdup. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - use devm_kstrdup instead of devm_kasprintf --- sound/soc/codecs/tas2781-i2c.c | 23 ++++++++++------------- 1 file changed, 10 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index eb8732b1bb99..59fcce7fef7b 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -346,13 +346,11 @@ static int tasdevice_create_control(struct tasdevice_priv *tas_priv) } /* Create a mixer item for selecting the active profile */ - name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - GFP_KERNEL); + name = devm_kstrdup(tas_priv->dev, "Speaker Profile Id", GFP_KERNEL); if (!name) { ret = -ENOMEM; goto out; } - scnprintf(name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, "Speaker Profile Id"); prof_ctrls[mix_index].name = name; prof_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; prof_ctrls[mix_index].info = tasdevice_info_profile; @@ -441,18 +439,13 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) goto out; } - /* Create a mixer item for selecting the active profile */ - prog_name = devm_kzalloc(tas_priv->dev, - SNDRV_CTL_ELEM_ID_NAME_MAXLEN, GFP_KERNEL); - conf_name = devm_kzalloc(tas_priv->dev, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, + /* Create mixer items for selecting the active Program and Config */ + prog_name = devm_kstrdup(tas_priv->dev, "Speaker Program Id", GFP_KERNEL); - if (!prog_name || !conf_name) { + if (!prog_name) { ret = -ENOMEM; goto out; } - - scnprintf(prog_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Program Id"); dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -460,8 +453,12 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) dsp_ctrls[mix_index].put = tasdevice_program_put; mix_index++; - scnprintf(conf_name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN, - "Speaker Config Id"); + conf_name = devm_kstrdup(tas_priv->dev, "Speaker Config Id", + GFP_KERNEL); + if (!conf_name) { + ret = -ENOMEM; + goto out; + } dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; -- 2.34.1

3 2

[PATCH] ASoC: MAINTAINERS: Drop incorrect tlv320aic31xx.txt path
by Krzysztof Kozlowski 26 Aug '24

26 Aug '24

tlv320aic31xx.txt was converted to DT schema (YAML) and new file is already matched by wildcard. This fixes get_maintainers.pl self-test warning: ./MAINTAINERS:22739: warning: no file matches F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt Fixes: e486feb7b8ec ("ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 6574b4910f8a..028186bb4e8d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22736,7 +22736,6 @@ F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt F: include/sound/tas2*.h F: include/sound/tlv320*.h F: include/sound/tpa6130a2-plat.h -- 2.43.0

2 1

[PATCH] MAINTAINERS: adjust file entries in TEXAS INSTRUMENTS AUDIO DRIVERS
by Lukas Bulwahn 26 Aug '24

26 Aug '24

From: Lukas Bulwahn <lukas.bulwahn(a)redhat.com> Commit e486feb7b8ec ("ASoC: dt-bindings: convert tlv320aic31xx.txt to yaml") converts tlv320aic31xx.txt to ti,tlv320dac3100.yaml, but misses to adjust the file entry in TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS. Hence, ./scripts/get_maintainer.pl --self-test=patterns complains about a broken reference. The pattern ti,tlv320*.yaml in the section already covers ti,tlv320dac3100.yaml and ti,tlv320adcx140.yaml. So, further file entries in this section are redundant. Adjust the file entries in TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS accordingly. Signed-off-by: Lukas Bulwahn <lukas.bulwahn(a)redhat.com> --- MAINTAINERS | 2 -- 1 file changed, 2 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 5dd88c28c0f7..6aa85a43d8c1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22753,8 +22753,6 @@ F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml -F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt F: include/sound/tas2*.h F: include/sound/tlv320*.h F: include/sound/tpa6130a2-plat.h -- 2.45.2

1 0

[PATCH v1] ASoC: tas2781: mark const variables tas2563_dvc_table as __maybe_unused
by Shenghao Ding 26 Aug '24

26 Aug '24

In case of tas2781, tas2563_dvc_table will be unused, so mark it as __maybe_unused. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- include/sound/tas2563-tlv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/tas2563-tlv.h b/include/sound/tas2563-tlv.h index faa3e194f73b..bb269b21f460 100644 --- a/include/sound/tas2563-tlv.h +++ b/include/sound/tas2563-tlv.h @@ -18,7 +18,7 @@ static const __maybe_unused DECLARE_TLV_DB_SCALE(tas2563_dvc_tlv, -12150, 50, 1); /* pow(10, db/20) * pow(2,30) */ -static const unsigned char tas2563_dvc_table[][4] = { +static const __maybe_unused unsigned char tas2563_dvc_table[][4] = { { 0X00, 0X00, 0X00, 0X00 }, /* -121.5db */ { 0X00, 0X00, 0X03, 0XBC }, /* -121.0db */ { 0X00, 0X00, 0X03, 0XF5 }, /* -120.5db */ -- 2.34.1

4 6

[DISCUSSION] Why doesn't ALSA autogenerate/autoconfigure the mono profile by default?
by Xerox Slayer 26 Aug '24

26 Aug '24

Sorry if this is the wrong place to discuss this, but from what I managed to find out, this is primarily the responsibility of ALSA. This was pointed out by a reddit user, and to be frank, I was kinda surprised that this is not done by default. The user in question is hearing impaired (can listen on only one ear), so the user has no benefit of actually listening to stereo or any other multichannel audio, thus wanted to just set everything to mono (dual mono I presume) and was puzzled when there was no such option in pulseaudio (which pulls the profiles from ALSA). I did some more investigation and as it turns out, this profile is autogenerated, but only if the hardware has such capability (as is the case with any other profile). You can enable it manually by editing config files (in which case I presume the channel downconversion is done in software), but it seemed odd that one has to go through such trouble just to listen to mono sound. Thus, I think the mono profile should be taken as a special case scenario and be autogenerated/configured by default, regardless if it's supported in hardware or not. Once again, I'm sorry if this is not the place to ask/discuss this, but would appreciate if someone pointed me to where I can actually do this.

2 2

[PATCH v4] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 25 Aug '24

25 Aug '24

Due to internal re-org, Kevin is no longer mnaintaining audio driver. Due to job change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. So far, the Software maintainer has not been confirmed. Once the maintainer was confimred, the guy will update his mail into audio converter section. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v4: - Add Touch Screen support v3: - Add Audio converter section - Set the section of LOW/MIDDLE-POWER AUDIO AMPLIFIER as Supported v2: - Add the detailed information of the maintained drivers. --- MAINTAINERS | 34 +++++++++++++++++++++++++--------- 1 file changed, 25 insertions(+), 9 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 878dcd23b331..ff363eac55b1 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -22591,26 +22591,43 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml +F: Documentation/devicetree/bindings/sound/tpa6130a2.txt +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml F: Documentation/devicetree/bindings/sound/tlv320aic31xx.txt -F: Documentation/devicetree/bindings/sound/tpa6130a2.txt -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -22620,9 +22637,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

[PATCH 0/2] Add SOF support for ACP7.0 based platform
by Vijendar Mukunda 24 Aug '24

24 Aug '24

This patch series adds SOF support for ACP7.0 based platform. Link: https://github.com/thesofproject/linux/pull/5139 Vijendar Mukunda (2): ASoC: amd: Add acpi machine id for acp7.0 version based platform ASoC: SOF: amd: add support for acp7.0 based platform sound/soc/amd/acp-config.c | 12 +++ sound/soc/amd/mach-config.h | 1 + sound/soc/sof/amd/Kconfig | 10 ++ sound/soc/sof/amd/Makefile | 4 +- sound/soc/sof/amd/acp-dsp-offset.h | 24 ++++- sound/soc/sof/amd/acp.c | 65 ++++++++++--- sound/soc/sof/amd/acp.h | 9 ++ sound/soc/sof/amd/acp70.c | 142 +++++++++++++++++++++++++++++ sound/soc/sof/amd/pci-acp70.c | 112 +++++++++++++++++++++++ 9 files changed, 365 insertions(+), 14 deletions(-) create mode 100644 sound/soc/sof/amd/acp70.c create mode 100644 sound/soc/sof/amd/pci-acp70.c -- 2.34.1

2 3

[PATCH] ASoC: Intel: soc-acpi-cht: Make Lenovo Yoga Tab 3 X90F DMI match less strict
by Hans de Goede 24 Aug '24

24 Aug '24

There are 2G and 4G RAM versions of the Lenovo Yoga Tab 3 X90F and it turns out that the 2G version has a DMI product name of "CHERRYVIEW D1 PLATFORM" where as the 4G version has "CHERRYVIEW C0 PLATFORM". The sys-vendor + product-version check are unique enough that the product-name check is not necessary. Drop the product-name check so that the existing DMI match for the 4G RAM version also matches the 2G RAM version. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/common/soc-acpi-intel-cht-match.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-cht-match.c b/sound/soc/intel/common/soc-acpi-intel-cht-match.c index 5e2ec60e2954..e4c3492a0c28 100644 --- a/sound/soc/intel/common/soc-acpi-intel-cht-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-cht-match.c @@ -84,7 +84,6 @@ static const struct dmi_system_id lenovo_yoga_tab3_x90[] = { /* Lenovo Yoga Tab 3 Pro YT3-X90, codec missing from DSDT */ .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Intel Corporation"), - DMI_MATCH(DMI_PRODUCT_NAME, "CHERRYVIEW D1 PLATFORM"), DMI_MATCH(DMI_PRODUCT_VERSION, "Blade3-10A-001"), }, }, -- 2.46.0

3 2

[PATCH 6.11 regression fix] ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards harder
by Hans de Goede 24 Aug '24

24 Aug '24

Since commit 13f58267cda3 ("ASoC: soc.h: don't create dummy Component via COMP_DUMMY()") dummy codecs declared like this: SND_SOC_DAILINK_DEF(dummy, DAILINK_COMP_ARRAY(COMP_DUMMY())); expand to: static struct snd_soc_dai_link_component dummy[] = { }; Which means that dummy is a zero sized array and thus dais[i].codecs should not be dereferenced *at all* since it points to the address of the next variable stored in the data section as the "dummy" variable has an address but no size, so even dereferencing dais[0] is already an out of bounds array reference. Which means that the if (dais[i].codecs->name) check added in commit 7d99a70b6595 ("ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards") relies on that the part of the next variable which the name member maps to just happens to be NULL. Which apparently so far it usually is, except when it isn't and then it results in crashes like this one: [ 28.795659] BUG: unable to handle page fault for address: 0000000000030011 ... [ 28.795780] Call Trace: [ 28.795787] <TASK> ... [ 28.795862] ? strcmp+0x18/0x40 [ 28.795872] 0xffffffffc150c605 [ 28.795887] platform_probe+0x40/0xa0 ... [ 28.795979] ? __pfx_init_module+0x10/0x10 [snd_soc_sst_bytcr_wm5102] Really fix things this time around by checking dais.num_codecs != 0. Fixes: 7d99a70b6595 ("ASoC: Intel: Boards: Fix NULL pointer deref in BYT/CHT boards") Cc: stable(a)vger.kernel.org Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/bytcr_wm5102.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- 9 files changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index dce6a2086f2a..6da1517c53c6 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -605,7 +605,7 @@ static int broxton_audio_probe(struct platform_device *pdev) int i; for (i = 0; i < ARRAY_SIZE(broxton_rt298_dais); i++) { - if (card->dai_link[i].codecs->name && + if (card->dai_link[i].num_codecs && !strncmp(card->dai_link[i].codecs->name, "i2c-INT343A:00", I2C_NAME_SIZE)) { if (!strncmp(card->name, "broxton-rt298", diff --git a/sound/soc/intel/boards/bytcht_cx2072x.c b/sound/soc/intel/boards/bytcht_cx2072x.c index c014d85a08b2..df3c2a7b64d2 100644 --- a/sound/soc/intel/boards/bytcht_cx2072x.c +++ b/sound/soc/intel/boards/bytcht_cx2072x.c @@ -241,7 +241,7 @@ static int snd_byt_cht_cx2072x_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_cx2072x_dais); i++) { - if (byt_cht_cx2072x_dais[i].codecs->name && + if (byt_cht_cx2072x_dais[i].num_codecs && !strcmp(byt_cht_cx2072x_dais[i].codecs->name, "i2c-14F10720:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcht_da7213.c b/sound/soc/intel/boards/bytcht_da7213.c index f4ac3ddd148b..08c598b7e1ee 100644 --- a/sound/soc/intel/boards/bytcht_da7213.c +++ b/sound/soc/intel/boards/bytcht_da7213.c @@ -245,7 +245,7 @@ static int bytcht_da7213_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(dailink); i++) { - if (dailink[i].codecs->name && + if (dailink[i].num_codecs && !strcmp(dailink[i].codecs->name, "i2c-DLGS7213:00")) { dai_index = i; break; diff --git a/sound/soc/intel/boards/bytcht_es8316.c b/sound/soc/intel/boards/bytcht_es8316.c index 2fcec2e02bb5..77b91ea4dc32 100644 --- a/sound/soc/intel/boards/bytcht_es8316.c +++ b/sound/soc/intel/boards/bytcht_es8316.c @@ -546,7 +546,7 @@ static int snd_byt_cht_es8316_mc_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_cht_es8316_dais); i++) { - if (byt_cht_es8316_dais[i].codecs->name && + if (byt_cht_es8316_dais[i].num_codecs && !strcmp(byt_cht_es8316_dais[i].codecs->name, "i2c-ESSX8316:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index a64d1989e28a..db4a33680d94 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1677,7 +1677,7 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_rt5640_dais); i++) { - if (byt_rt5640_dais[i].codecs->name && + if (byt_rt5640_dais[i].num_codecs && !strcmp(byt_rt5640_dais[i].codecs->name, "i2c-10EC5640:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 80c841b000a3..8514b79f389b 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -910,7 +910,7 @@ static int snd_byt_rt5651_mc_probe(struct platform_device *pdev) /* fix index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_rt5651_dais); i++) { - if (byt_rt5651_dais[i].codecs->name && + if (byt_rt5651_dais[i].num_codecs && !strcmp(byt_rt5651_dais[i].codecs->name, "i2c-10EC5651:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/bytcr_wm5102.c b/sound/soc/intel/boards/bytcr_wm5102.c index cccb5e90c0fe..e5a7cc606aa9 100644 --- a/sound/soc/intel/boards/bytcr_wm5102.c +++ b/sound/soc/intel/boards/bytcr_wm5102.c @@ -605,7 +605,7 @@ static int snd_byt_wm5102_mc_probe(struct platform_device *pdev) /* find index of codec dai */ for (i = 0; i < ARRAY_SIZE(byt_wm5102_dais); i++) { - if (byt_wm5102_dais[i].codecs->name && + if (byt_wm5102_dais[i].num_codecs && !strcmp(byt_wm5102_dais[i].codecs->name, "wm5102-codec")) { dai_index = i; diff --git a/sound/soc/intel/boards/cht_bsw_rt5645.c b/sound/soc/intel/boards/cht_bsw_rt5645.c index eb41b7115d01..1da9ceee4d59 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5645.c +++ b/sound/soc/intel/boards/cht_bsw_rt5645.c @@ -569,7 +569,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* set correct codec name */ for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) - if (cht_dailink[i].codecs->name && + if (cht_dailink[i].num_codecs && !strcmp(cht_dailink[i].codecs->name, "i2c-10EC5645:00")) { dai_index = i; diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index be2d1a8dbca8..d68e5bc755de 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -466,7 +466,7 @@ static int snd_cht_mc_probe(struct platform_device *pdev) /* find index of codec dai */ for (i = 0; i < ARRAY_SIZE(cht_dailink); i++) { - if (cht_dailink[i].codecs->name && + if (cht_dailink[i].num_codecs && !strcmp(cht_dailink[i].codecs->name, RT5672_I2C_DEFAULT)) { dai_index = i; break; -- 2.46.0

3 3

[GIT PULL] ASoC fixes for v6.11-rc3
by Mark Brown 23 Aug '24

23 Aug '24

The following changes since commit 72776774b55bb59b7b1b09117e915a5030110304: ASoC: cs35l56: Patch CS35L56_IRQ1_MASK_18 to the default value (2024-08-08 20:34:56 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.11-rc3 for you to fetch changes up to bb4485562f5907708f1c218b5d70dce04165d1e1: ASoC: cs-amp-lib: Ignore empty UEFI calibration entries (2024-08-22 17:52:26 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.11 A relatively large collection of fixes here, all driver specific and none of them particularly major, plus one MAINTAINERS update. There's been a bunch of work on module autoloading from several people. ---------------------------------------------------------------- Albert Jakieła (1): ASoC: SOF: mediatek: Add missing board compatible Dmitry Baryshkov (2): ASoC: codecs: lpass-macro: fix version strings returned for 1.x codecs ASoC: codecs: lpass-va-macro: warn on unknown version Hongbo Li (2): ASoC: allow module autoloading for table db1200_pids ASoC: allow module autoloading for table board_ids Krzysztof Kozlowski (2): ASoC: MAINTAINERS: Drop Banajit Goswami from Qualcomm sound drivers ASoC: codecs: wcd937x: Fix missing de-assert of reset GPIO Mark Brown (1): ASoC: codecs: warn on unknown codec version Richard Fitzgerald (2): ASoC: cs-amp-lib-test: Force test calibration blob entries to be valid ASoC: cs-amp-lib: Ignore empty UEFI calibration entries Vijendar Mukunda (3): ASoC: SOF: amd: move iram-dram fence register programming sequence ASoC: SOF: amd: Fix for incorrect acp error register offsets ASoC: SOF: amd: Fix for acp init sequence YR Yang (1): ASoC: mediatek: mt8188: Mark AFE_DAC_CON0 register as volatile Yuntao Liu (1): ASoC: amd: acp: fix module autoloading MAINTAINERS | 1 - sound/soc/amd/acp/acp-legacy-mach.c | 2 ++ sound/soc/amd/acp/acp-sof-mach.c | 2 ++ sound/soc/au1x/db1200.c | 1 + sound/soc/codecs/cs-amp-lib-test.c | 9 ++++++ sound/soc/codecs/cs-amp-lib.c | 7 +++- sound/soc/codecs/lpass-macro-common.h | 6 ++++ sound/soc/codecs/lpass-va-macro.c | 4 +++ sound/soc/codecs/wcd937x.c | 5 ++- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 1 + sound/soc/sof/amd/acp-dsp-offset.h | 6 ++-- sound/soc/sof/amd/acp.c | 52 ++++++++++++++++++++---------- sound/soc/sof/amd/acp.h | 9 ++++-- sound/soc/sof/amd/pci-acp63.c | 2 ++ sound/soc/sof/amd/pci-rmb.c | 2 ++ sound/soc/sof/amd/pci-rn.c | 2 ++ sound/soc/sof/mediatek/mt8195/mt8195.c | 3 ++ 17 files changed, 88 insertions(+), 26 deletions(-)

2 1

[PATCH v8 0/4] ASoC: grace time for DPCM cleanup
by Kuninori Morimoto 23 Aug '24

23 Aug '24

Hi Mark, Pierre-Louis, Jerome Cc each ASoC driver maintainer This is v8 of DPCM cleanup As we discussed in [1], we don't need to use dpcm_playback/capture flag, so we remove it. But we have been using it for 10 years, some driver might get damage. The most likely case is that the device/driver can use both playback/capture, but have only one flag, and not using xxx_only flag. [1/3] patch indicates warning in such case. These adds grace time for DPCM cleanup. I'm not sure when dpcm_xxx will be removed, and Codec check bypass will be error, but maybe v6.12 or v6.13 ? Please check each driver by that time. Previous patch-set try to check both CPU and Codec in DPCM, but we noticed that there are some special DAI which we can't handle today [2]. So I will escape it in this patch-set. [1] https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com [2] https://lore.kernel.org/all/3e67d62d-fe08-4f55-ab5b-ece8a57154f9@linux.inte… Link: https://lore.kernel.org/r/87edaym2cg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87wmo6dyxg.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87msole5wc.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/871q5tnuok.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87bk4oqerx.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/8734pctmte.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87r0ctwzr4.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87cymvlmki.wl-kuninori.morimoto.gx@renesas.com v8 -> v7 - rebase to latest ASoC branch (for-6.12) v6 -> v7 - based on latest ASoC branch (for-6.12) v5 -> v6 - based on latest ASoC branch v4 -> v5 - typo fix: limition -> limitation - typo fix: catpure -> capture - include Intel patch v3 -> v4 - don't check Codec on DPCM - include Jerome's dpcm_xxx update patch v2 -> v3 - tidyup typo (reuqsts -> requests) - add Tested-by on git-log v1 -> v2 - tidyup Codec check warning output condition Jerome Brunet (1): ASoC: amlogic: do not use dpcm_playback/capture flags Kuninori Morimoto (3): ASoC: soc-pcm: Indicate warning if dpcm_playback/capture were used for availability limition ASoC: remove snd_soc_dai_link_set_capabilities() ASoC: Intel: sof_sdw: use playback/capture_only flags include/sound/soc-dai.h | 1 - include/sound/soc.h | 1 + sound/soc/fsl/imx-card.c | 3 -- sound/soc/generic/audio-graph-card.c | 2 - sound/soc/generic/audio-graph-card2.c | 2 - sound/soc/generic/simple-card.c | 2 - sound/soc/meson/axg-card.c | 11 +++-- sound/soc/meson/gx-card.c | 1 - sound/soc/meson/meson-card-utils.c | 4 +- sound/soc/qcom/common.c | 1 - sound/soc/sdw_utils/soc_sdw_utils.c | 4 +- sound/soc/soc-dai.c | 38 ---------------- sound/soc/soc-pcm.c | 65 +++++++++++++++------------ 13 files changed, 47 insertions(+), 88 deletions(-) -- 2.43.0

2 5

ALC285 headphone no sound
by Kailang 22 Aug '24

22 Aug '24

Hi Takashi, I also changed with ALC215 ALC285 ALC289 ALC225 ALC295 ALC299 codec. It maybe has similar issue as ALC256. Let it has same procedure. Many Thanks. BR, Kailang > -----Original Message----- > From: Kailang > Sent: Thursday, August 22, 2024 11:14 AM > To: Takashi Iwai (tiwai(a)suse.de) <tiwai(a)suse.de> > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > Subject: ALC256 headphone no sound > > Hi Takashi, > > Attach patch will solve headphone issue for alc256. > Many Thanks. > > BR, > Kailang

2 1

Re: [PATCH] [RFC] ASoC: Conditional PCM support
by Jaroslav Kysela 22 Aug '24

22 Aug '24

On 21. 08. 24 12:18, Cezary Rojewski wrote: > Conditional PCM (condpcm) helps facilitate modern audio usecases such as > Echo Cancellations and Noise Reduction. These are not invoked by the > means of userspace application opening an endpoint (FrontEnd) but are a > "side effect" of selected PCMs running simultaneously e.g.: if both > Speaker (source) and Microphone Array (sink) are running, reference > data from the Speaker and take it into account when processing capture > for better voice command detection ratio. > > Which PCMs are needed for given conditional PCM to be spawned is > determinated by the driver when registering the condpcm. Perhaps the naming may be changed to "sidepcm" or "monitorpcm"/"monpcm". It looks better in my eyes. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

2 1

ALC256 headphone no sound
by Kailang 22 Aug '24

22 Aug '24

Hi Takashi, Attach patch will solve headphone issue for alc256. Many Thanks. BR, Kailang

2 2

[PATCH v2 0/2] ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function
by Kuninori Morimoto 22 Aug '24

22 Aug '24

Hi Mark No driver is calling snd_soc_dpcm_can_be_xxx() functions. We don't need to have EXPORT_SYMBOL_GPL() for them. This patch-set makes it static function. v1 -> v2 - rebased to latest branch (for-6.12) Kuninori Morimoto (2): ASoC: soc-pcm: move snd_soc_dpcm_can_be_xxx() to top ASoC: soc-pcm: makes snd_soc_dpcm_can_be_xxx() local function include/sound/soc-dpcm.h | 18 ---- sound/soc/soc-pcm.c | 201 +++++++++++++++++++-------------------- 2 files changed, 99 insertions(+), 120 deletions(-) -- 2.43.0

2 6

USB-Audio: Add support for Focusrite 4th Gen devices
by GitHub pull_request - opened 22 Aug '24

22 Aug '24

alsa-project/alsa-ucm-conf pull request #438 was opened from ChrisLane: Extending the existing support for Focusrite Scarlett Solo and 2i2 interfaces with support for the 4th Gen devices. I have a Focusrite Scarlett 2i2 which is what I have tested with and used to get the device ID. For the Solo device, I grabbed the device ID from [linux-hardware.org](https://linux-hardware.org/?id=usb:1235-8218). Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/438 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/438.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v24 00/34] Introduce QC USB SND audio offloading support
by Wesley Cheng 21 Aug '24

21 Aug '24

Requesting to see if we can get some Acked-By tags, and merge on usb-next. Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer ========================= soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP ============================== q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support ================================== XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver ======================================== qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... The current limitation is that the latest USB audio device that is identified will be automatically selected by the Q6USB BE DAI for offloading. Future patches can be added to possibly add for more flexibility, but until the userpace applications can be better defined, having these mechanisms will complicate the overall implementation. USB offload Kcontrols ===================== Part of the vendor offload package will have a mixer driver associated with it (mixer_usb_offload.c). This entity will be responsible for coordinating with SOC USB and the Q6USB backend DAI to fetch information about the sound card and PCM device indices associated with the offload path. The logic is done based on the current implementation of how paths are controlled within the QC ASoC implementation. QC ASoC Q6Routing ----------------- Within the Q6 ASOC design, the registered ASoC platform card will expose a set of kcontrols for enabling the BE DAI links to the FE DAI link. For example: tinymix -D 0 contents Number of controls: 1033 ctl type num name value ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 Off 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off Each of these kcontrols will enable the USB BE DAI link (q6usb) to be connected to a FE DAI link (q6asm). Since each of these controls are DAPM widgets, when it is enabled, the DAPM widget's "connect" flag is updated accordingly. USB Offload Mapping ------------------- Based on the Q6routing, the USB BE DAI link can determine which sound card and PCM device is enabled for offloading. Fetching the ASoC platform sound card's information is fairly straightforward, and the bulk of the work goes to finding the corresponding PCM device index. As mentioned above, the USB BE DAI can traverse the DAPM widgets to find the DAPM path that is related to the control for the "USB Mixer." Based on which "USB Mixer" is enabled, it can find the corresponding DAPM widget associated w/ the FE DAI link (Multimedia*). From there it can find the PCM device created for the Multimedia* stream. Only one BE DAI link can be enabled per FE DAI. For example, if the HDMI path is enabled for Multimedia1, the USB Mixer will be disabled and switched over. Examples of kcontrol -------------------- tinymix -D 0 contents Number of controls: 1033 ctl type num name ... 1025 BOOL 1 USB Mixer MultiMedia1 Off 1026 BOOL 1 USB Mixer MultiMedia2 On 1027 BOOL 1 USB Mixer MultiMedia3 Off 1028 BOOL 1 USB Mixer MultiMedia4 Off 1029 BOOL 1 USB Mixer MultiMedia5 Off 1030 BOOL 1 USB Mixer MultiMedia6 Off 1031 BOOL 1 USB Mixer MultiMedia7 Off 1032 BOOL 1 USB Mixer MultiMedia8 Off tinymix -D 2 contents Number of controls: 7 ctl type num name value 0 INT 2 Playback Channel Map 0, 0 (range 0->36) 1 BOOL 2 MDR-1ADAC Playback Switch On, On 2 BOOL 1 MDR-1ADAC Playback Switch On 3 INT 2 MDR-1ADAC Playback Volume 127, 127 (range 0->127) 4 INT 1 MDR-1ADAC Playback Volume 127 (range 0->127) 5 BOOL 1 Sony Internal Clock Validity On 6 INT 2 USB Offload Playback Route PCM#0 0, 1 (range -1->255) The example highlights that the userspace/application can utilize the offload path for the USB device on card#0 PCM device#1. When dealing with multiple USB audio devices, only the latest USB device identified is going to be selected for offload capable. tinymix -D 1 contents Number of controls: 9 ctl type num name value 0 INT 2 Capture Channel Map 0, 0 (range 0->36) 1 INT 2 Playback Channel Map 0, 0 (range 0->36) 2 BOOL 1 Headset Capture Switch On 3 INT 1 Headset Capture Volume 1 (range 0->4) 4 BOOL 1 Sidetone Playback Switch On 5 INT 1 Sidetone Playback Volume 4096 (range 0->8192) 6 BOOL 1 Headset Playback Switch On 7 INT 2 Headset Playback Volume 20, 20 (range 0->24) 8 INT 2 USB Offload Playback Route PCM#0 -1, -1 (range -1->255) "-1, -1" shows that this device has no route to the offload path. This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v24: - Simplified the kcontrols involved in determining how to utilize the offload path. - There is one kcontrol registered to each USB audio device that will output which card/pcm device it is mapped to for the offload route. - Removed kcontrols to track offload status and device selection. - Default to last USB audio device plugged in as offload capable. - kcontrol will reside on USB SND device. - Reworked the tracking of connected USB devices from the Q6USB BE DAI link. Previously, it was convoluted by doing it over an array, but moved to using a list made it much simpler. Logic is still unchanged in that the last USB headset plugged in will be selected for offloading. - Updated the USB SOC RST documentation accordingly with new kcontrol updates. - Added logic to fetch mapped ASoC card and pcm device index that the offload path is mapped to for the USB SND kcontrol (for offload route). - Re-ordered series to hopefully make reviews more readable by combining patches based on the layer modified (ie QC ASoC, ASoC, USB sound, and USB XHCI). Changes in v23: - Added MODULE_DESCRIPTION() fields to drivers that needed it. Changes in v22: - Removed components tag for the ASoC platform card, as the USB SND kcontrol for notifying userspace of offload capable card achieves similar results. - Due to the above, had to remove the review-by tag for the RST documentation, as changes were made to remove the components tag section. - Took in feedback to make the SOC USB add/remove ports void. - Fixed an issue w/ the USB SND kcontrol management for devices that have multi UAC interfaces. (would attempt to create the kcontrol more than once) - Modified SOC USB card and PCM index select to be based off the num_supported streams that is specified by the USB BE DAI. - Modified comments on selecting the latest USB headset for offloading. Changes in v21: - Added an offload jack disable path from the ASoC platform driver and SOC USB. - Refactored some of the existing SOC USB context look up APIs and created some new helpers to search for the USB context. - Renamed snd_soc_usb_find_format to snd_soc_usb_find_supported_format - Removed some XHCI sideband calls that would allow clients to actually enable the IRQ line associated w/ the secondary interrupter. This is removed because there are other dependencies that are required for that to happen, which are not covered as part of this series, and to avoid confusion. - Due to the above, removed the need to export IMOD setting, and enable/disable interrupter APIs. Changes in v20: - Fixed up some formatting changes pointed out in the usb.rst - Added SB null check during XHCI sideband unregister in case caller passes improper argument (xhci_sideband_unregister()) Changes in v19: - Rebased to usb-next to account for some new changes in dependent drivers. Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (32): usb: host: xhci: Repurpose event handler for skipping interrupter events usb: xhci: Allow for secondary interrupter to set IMOD usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index usb: host: xhci-plat: Set XHCI max interrupters if property is present usb: dwc3: Specify maximum number of XHCI interrupters ASoC: Add SOC USB APIs for adding an USB backend ASoC: usb: Create SOC USB SND jack kcontrol ASoC: usb: Fetch ASoC card and pcm device information ASoC: doc: Add documentation for SOC USB ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qcom: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: qcom: qdsp6: Fetch USB offload mapped card and PCM device ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: qcom: Don't allow USB offload path if PCM device is in use ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: qcom: Populate PCM and USB chip information ALSA: usb-audio: qcom: Add USB offload route kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 435 ++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 36 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 50 +- drivers/usb/host/xhci-sideband.c | 419 ++++ drivers/usb/host/xhci.c | 43 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 68 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 135 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 372 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 56 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 331 +++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 101 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1937 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 5768 insertions(+), 50 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

5 86

No sound from XonarDX
by GitHub issues - opened 21 Aug '24

21 Aug '24

alsa-project/alsa-lib issue #408 was opened from 3bitbrain: No matter what wiki or advice I try, no sound at all from a known good Xonar DX card. Distro : Gentoo, fresh install. Always un-muted everything in the console alsamixer. alsa-lib version 1.2.11 [alsa.txt](https://github.com/user-attachments/files/16697602/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-lib/issues/408 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: soc-pcm: remove snd_soc_dpcm_stream_lock_irqsave_nested()
by Kuninori Morimoto 21 Aug '24

21 Aug '24

soc-pcm.c has snd_soc_dpcm_stream_lock_irqsave_nested() / snd_soc_dpcm_stream_unlock_irqrestore() helper function, but it is almost nothing help. It just makes a code complex. Let's remove it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - rebased to latest branch (for-6.12) sound/soc/soc-pcm.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index f79abcf36105..baf8b81e7110 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -49,12 +49,6 @@ static inline int _soc_pcm_ret(struct snd_soc_pcm_runtime *rtd, return ret; } -#define snd_soc_dpcm_stream_lock_irqsave_nested(rtd, stream, flags) \ - snd_pcm_stream_lock_irqsave_nested(snd_soc_dpcm_get_substream(rtd, stream), flags) - -#define snd_soc_dpcm_stream_unlock_irqrestore(rtd, stream, flags) \ - snd_pcm_stream_unlock_irqrestore(snd_soc_dpcm_get_substream(rtd, stream), flags) - #define DPCM_MAX_BE_USERS 8 static inline const char *soc_cpu_dai_name(struct snd_soc_pcm_runtime *rtd) @@ -2144,7 +2138,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be = dpcm->be; be_substream = snd_soc_dpcm_get_substream(be, stream); - snd_soc_dpcm_stream_lock_irqsave_nested(be, stream, flags); + snd_pcm_stream_lock_irqsave_nested(be_substream, flags); /* is this op for this BE ? */ if (!snd_soc_dpcm_be_can_update(fe, be, stream)) @@ -2291,7 +2285,7 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, break; } next: - snd_soc_dpcm_stream_unlock_irqrestore(be, stream, flags); + snd_pcm_stream_unlock_irqrestore(be_substream, flags); if (ret) break; } -- 2.43.0

2 1

[PATCH] ASoC: SOF: amd: remove unused variable from sof_amd_acp_desc structure
by Vijendar Mukunda 21 Aug '24

21 Aug '24

Remove unused structure member 'rev' from sof_amd_acp_desc structure. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/amd/acp.h | 1 - sound/soc/sof/amd/pci-acp63.c | 1 - sound/soc/sof/amd/pci-rmb.c | 1 - sound/soc/sof/amd/pci-rn.c | 1 - sound/soc/sof/amd/pci-vangogh.c | 1 - 5 files changed, 5 deletions(-) diff --git a/sound/soc/sof/amd/acp.h b/sound/soc/sof/amd/acp.h index 321c40cc6d50..11def07efc0f 100644 --- a/sound/soc/sof/amd/acp.h +++ b/sound/soc/sof/amd/acp.h @@ -190,7 +190,6 @@ struct acp_dsp_stream { }; struct sof_amd_acp_desc { - unsigned int rev; const char *name; unsigned int host_bridge_id; u32 pgfsm_base; diff --git a/sound/soc/sof/amd/pci-acp63.c b/sound/soc/sof/amd/pci-acp63.c index e90658ba2bd7..b54ed61b79ed 100644 --- a/sound/soc/sof/amd/pci-acp63.c +++ b/sound/soc/sof/amd/pci-acp63.c @@ -28,7 +28,6 @@ #define ACP6x_REG_END 0x125C000 static const struct sof_amd_acp_desc acp63_chip_info = { - .rev = 6, .host_bridge_id = HOST_BRIDGE_ACP63, .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_enb = ACP6X_EXTERNAL_INTR_ENB, diff --git a/sound/soc/sof/amd/pci-rmb.c b/sound/soc/sof/amd/pci-rmb.c index a366f904e6f3..c45256bf4fda 100644 --- a/sound/soc/sof/amd/pci-rmb.c +++ b/sound/soc/sof/amd/pci-rmb.c @@ -28,7 +28,6 @@ #define ACP6X_FUTURE_REG_ACLK_0 0x1854 static const struct sof_amd_acp_desc rembrandt_chip_info = { - .rev = 6, .host_bridge_id = HOST_BRIDGE_RMB, .pgfsm_base = ACP6X_PGFSM_BASE, .ext_intr_stat = ACP6X_EXT_INTR_STAT, diff --git a/sound/soc/sof/amd/pci-rn.c b/sound/soc/sof/amd/pci-rn.c index 2b7c53470ce8..386a0f1e7ee0 100644 --- a/sound/soc/sof/amd/pci-rn.c +++ b/sound/soc/sof/amd/pci-rn.c @@ -28,7 +28,6 @@ #define ACP3X_FUTURE_REG_ACLK_0 0x1860 static const struct sof_amd_acp_desc renoir_chip_info = { - .rev = 3, .host_bridge_id = HOST_BRIDGE_CZN, .pgfsm_base = ACP3X_PGFSM_BASE, .ext_intr_stat = ACP3X_EXT_INTR_STAT, diff --git a/sound/soc/sof/amd/pci-vangogh.c b/sound/soc/sof/amd/pci-vangogh.c index eba580840100..cb845f81795e 100644 --- a/sound/soc/sof/amd/pci-vangogh.c +++ b/sound/soc/sof/amd/pci-vangogh.c @@ -26,7 +26,6 @@ #define ACP5X_FUTURE_REG_ACLK_0 0x1864 static const struct sof_amd_acp_desc vangogh_chip_info = { - .rev = 5, .name = "vangogh", .host_bridge_id = HOST_BRIDGE_VGH, .pgfsm_base = ACP5X_PGFSM_BASE, -- 2.34.1

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[PATCH] ASoC: codecs: wcd934x: make read-only array minCode_param static const
by Colin Ian King 21 Aug '24

21 Aug '24

Don't populate the read-only array minCode_param on the stack at run time, instead make it static const. Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/soc/codecs/wcd934x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 291d0c80a6fc..910852eb9698 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -2643,8 +2643,8 @@ static void wcd934x_mbhc_get_result_params(struct wcd934x_codec *wcd934x, s16 c1; s32 x1, d1; int32_t denom; - int minCode_param[] = { - 3277, 1639, 820, 410, 205, 103, 52, 26 + static const int minCode_param[] = { + 3277, 1639, 820, 410, 205, 103, 52, 26 }; regmap_update_bits(wcd934x->regmap, WCD934X_ANA_MBHC_ZDET, 0x20, 0x20); -- 2.39.2

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[PATCH v1] ASoC: tas2781: Remove unnecessary line feed for tasdevice_codec_remove
by Baojun Xu 21 Aug '24

21 Aug '24

Remove unnecessary line feed for tasdevice_codec_remove. Add comma at the end the last member of the array. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- v1: --- sound/soc/codecs/tas2781-i2c.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 9a32e0504857..eb8732b1bb99 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -582,13 +582,13 @@ static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SPK("SPK", tasdevice_dapm_event), SND_SOC_DAPM_OUTPUT("OUT"), - SND_SOC_DAPM_INPUT("DMIC") + SND_SOC_DAPM_INPUT("DMIC"), }; static const struct snd_soc_dapm_route tasdevice_audio_map[] = { {"SPK", NULL, "ASI"}, {"OUT", NULL, "SPK"}, - {"ASI OUT", NULL, "DMIC"} + {"ASI OUT", NULL, "DMIC"}, }; static int tasdevice_startup(struct snd_pcm_substream *substream, @@ -730,8 +730,7 @@ static void tasdevice_deinit(void *context) tas_priv->fw_state = TASDEVICE_DSP_FW_PENDING; } -static void tasdevice_codec_remove( - struct snd_soc_component *codec) +static void tasdevice_codec_remove(struct snd_soc_component *codec) { struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(codec); -- 2.43.0

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