February 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH] ALSA: hda/realtek: Improve support for Dell Precision 3260
by Jaroslav Kysela 03 Mar '23

03 Mar '23

The headset jack works better with model=alc283-dac-wcaps. Without this option, the headset insertion (separate physical jack) may not be handled correctly (re-insertion is required). It seems that it follows the "Intel Reference Board" defaults. Reported-by: steven_wu2(a)dell.com Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- 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 e103bb3693c0..7afbb708b705 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9260,6 +9260,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0ac9, "Dell Precision 3260", ALC295_FIXUP_CHROME_BOOK), SND_PCI_QUIRK(0x1028, 0x0b19, "Dell XPS 15 9520", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b1a, "Dell Precision 5570", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0b37, "Dell Inspiron 16 Plus 7620 2-in-1", ALC295_FIXUP_DELL_INSPIRON_TOP_SPEAKERS), -- 2.39.0

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[PATCH V2] ucm2: Rockchip: rk817: Add ALSA UCM support
by Chris Morgan 02 Mar '23

02 Mar '23

From: Chris Morgan <macromorgan(a)hotmail.com> Add ALSA-UCM support for the Rockchip RK817 audio codec. This codec is typically configured either with an internal or external amplifier as reflected by the longname. This configuration has been tested on the Anbernic RG353P (rk817_ext) and the Odroid Go Advance (rk817_int). Changes from V1: - Use a constant to control if the output mux is set to HP or SPK. Signed-off-by: Chris Morgan <macromorgan(a)hotmail.com> --- ucm2/Rockchip/rk817-sound/HiFi.conf | 68 ++++++++++++++++++++++ ucm2/Rockchip/rk817-sound/rk817-sound.conf | 6 ++ ucm2/conf.d/simple-card/rk817_ext.conf | 1 + ucm2/conf.d/simple-card/rk817_int.conf | 1 + 4 files changed, 76 insertions(+) create mode 100644 ucm2/Rockchip/rk817-sound/HiFi.conf create mode 100644 ucm2/Rockchip/rk817-sound/rk817-sound.conf create mode 120000 ucm2/conf.d/simple-card/rk817_ext.conf create mode 120000 ucm2/conf.d/simple-card/rk817_int.conf diff --git a/ucm2/Rockchip/rk817-sound/HiFi.conf b/ucm2/Rockchip/rk817-sound/HiFi.conf new file mode 100644 index 0000000..8d90567 --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/HiFi.conf @@ -0,0 +1,68 @@ +Define.pbk_mux "SPK" + +If.1 { + Condition { + Type ControlExists + Control "name='Internal Speakers Switch'" + } + + True { + Define.pbk_mux "HP" + SectionDevice."Speaker".EnableSequence [ + cset "name='Internal Speakers Switch' on" + ] + + SectionDevice."Speaker".DisableSequence [ + cset "name='Internal Speakers Switch' off" + ] + } +} + +SectionDevice."Speaker" { + Comment "Internal Speaker" + + EnableSequence [ + cset "name='Playback Mux' ${var:pbk_mux}" + ] + + Value { + PlaybackMixerElem "Master Playback Volume" + PlaybackPriority 100 + PlaybackPCM "hw:${CardId}" + } + + ConflictingDevice [ + "Headphones" + ] +} + +SectionDevice."Mic" { + Comment "Microphone" + + Value { + CapturePriority 100 + CapturePCM "hw:${CardId}" + CaptureMixerElem "Mic Capture Gain" + CaptureMasterElem "Master Capture Volume" + } +} + +SectionDevice."Headphones" { + Comment "Headphones" + + EnableSequence [ + cset "name='Playback Mux' HP" + ] + + Value { + PlaybackMixerElem "Master Playback Volume" + PlaybackPriority 200 + PlaybackPCM "hw:${CardId}" + JackControl "Headphones Jack" + JackHWMute "Speaker" + } + + ConflictingDevice [ + "Speaker" + ] +} diff --git a/ucm2/Rockchip/rk817-sound/rk817-sound.conf b/ucm2/Rockchip/rk817-sound/rk817-sound.conf new file mode 100644 index 0000000..90ec794 --- /dev/null +++ b/ucm2/Rockchip/rk817-sound/rk817-sound.conf @@ -0,0 +1,6 @@ +Syntax 4 + +SectionUseCase."HiFi" { + File "/Rockchip/rk817-sound/HiFi.conf" + Comment "Play HiFi quality music" +} diff --git a/ucm2/conf.d/simple-card/rk817_ext.conf b/ucm2/conf.d/simple-card/rk817_ext.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_ext.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file diff --git a/ucm2/conf.d/simple-card/rk817_int.conf b/ucm2/conf.d/simple-card/rk817_int.conf new file mode 120000 index 0000000..6fa8db7 --- /dev/null +++ b/ucm2/conf.d/simple-card/rk817_int.conf @@ -0,0 +1 @@ +../../Rockchip/rk817-sound/rk817-sound.conf \ No newline at end of file -- 2.34.1

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[PATCH v3 0/3] ASoC: mchp-pdmc: fix poc noises when starting capture
by Claudiu Beznea 28 Feb '23

28 Feb '23

To start capture on Microchip PDMC the enable bits for each supported microphone need to be set. After this bit is set the PDMC starts to receive data from microphones and it considers this data as valid data. Thus if microphones are not ready the PDMC captures anyway data from its lines. This data is interpreted by the human ear as poc noises. To avoid this the following software workaround need to be applied when starting capture: 1/ enable PDMC channel 2/ wait 150ms 3/ execute 16 dummy reads from RHR 4/ clear interrupts 5/ enable interrupts 6/ enable DMA channel For this workaround to work step 6 need to be executed at the end. For step 6 was added patch 1/3 from this series. With this, component DAI driver sets its struct snd_soc_component_driver::start_dma_last = 1 and proper action is taken based on this flag when starting DAI trigger vs DMA. Thank you, Claudiu Beznea Changes in v3: - update the commit message of patch 2/3 to be more descriptive - in patch 3/3 initialize dd->startup_delay_us before calling of_property_read_u32() and don't check its return value; property is optional and the the default value is 150ms Changes in v2: - patch 1/3 from v1 is now "ASoC: soc-pcm: add option to start DMA after DAI" - pass start_dma_last from component DAI driver object (struct snd_soc_component_driver::start_dma_last); adapt patch 3/3 after this; - in patch 1/3 s/Do we need to start dma first/Do we need to start dma last in comment from soc_pcm_trigger() - collect review tag from Krzysztof Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup Claudiu Beznea (3): ASoC: soc-pcm: add option to start DMA after DAI ASoC: dt-bindings: sama7g5-pdmc: add microchip,startup-delay-us binding ASoC: mchp-pdmc: fix poc noise at capture startup .../sound/microchip,sama7g5-pdmc.yaml | 6 +++ include/sound/soc-component.h | 2 + sound/soc/atmel/mchp-pdmc.c | 53 +++++++++++++++++-- sound/soc/soc-pcm.c | 27 ++++++++-- 4 files changed, 78 insertions(+), 10 deletions(-) -- 2.34.1

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[PATCH] ASoC: zl38060 add gpiolib dependency
by Arnd Bergmann 28 Feb '23

28 Feb '23

From: Arnd Bergmann <arnd(a)arndb.de> Without gpiolib, this driver fails to link: arm-linux-gnueabi-ld: sound/soc/codecs/zl38060.o: in function `chip_gpio_get': zl38060.c:(.text+0x30): undefined reference to `gpiochip_get_data' arm-linux-gnueabi-ld: sound/soc/codecs/zl38060.o: in function `zl38_spi_probe': zl38060.c:(.text+0xa18): undefined reference to `devm_gpiochip_add_data_with_key' This appears to have been in the driver since the start, but is hard to hit in randconfig testing since gpiolib is almost always selected by something else. Fixes: 52e8a94baf90 ("ASoC: Add initial ZL38060 driver") Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bd72c426a93d..07747565c3b5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -2103,6 +2103,7 @@ config SND_SOC_WSA883X config SND_SOC_ZL38060 tristate "Microsemi ZL38060 Connected Home Audio Processor" depends on SPI_MASTER + depends on GPIOLIB select REGMAP help Support for ZL38060 Connected Home Audio Processor from Microsemi, -- 2.39.2

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[PATCH] ASoC: sam9g20ek: Disable capture unless building with microphone input
by Mark Brown 28 Feb '23

28 Feb '23

Without modification the AT91SAM9G20-EK has no capture support, none of the inputs of the CODEC are wired to anything to useful and there are no paths supporting loopback. Since the audio is clocked from the CODEC and the DAPM inputs are marked as unusable this means that capture will fail to transfer any data as the ADC path can't be powered up. Flag this in the device description so apps don't see unusable capture support, guarded with the existing optional define for mic input. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/atmel/sam9g20_wm8731.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 1430642c8433..785b9d01d8af 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -98,6 +98,9 @@ static struct snd_soc_dai_link at91sam9g20ek_dai = { .init = at91sam9g20ek_wm8731_init, .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBP_CFP, +#ifndef ENABLE_MIC_INPUT + .playback_only = true, +#endif SND_SOC_DAILINK_REG(pcm), }; --- base-commit: cac85e4616b1cf4a90844b952b49b9cbc4562530 change-id: 20230225-asoc-sam9g20ek-f4e775205d8c Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH 0/2] ASoC: mt8183: Fixes from an initial glance at a kselftest run
by Mark Brown 28 Feb '23

28 Feb '23

This is a collection of fixes I came up after glancing through an initial test run with the snappily named Kukui Jacuzzi SKU16 Chromebook on KernelCI. There are more issues flagged, this is just what I fixed thus far. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (2): ASoC: mt8183: Remove spammy logging from I2S DAI driver ASoC: mt8183: Fix event generation for I2S DAI operations sound/soc/mediatek/mt8183/mt8183-dai-i2s.c | 21 +++------------------ 1 file changed, 3 insertions(+), 18 deletions(-) --- base-commit: d2980d8d826554fa6981d621e569a453787472f8 change-id: 20230224-asoc-mt8183-quick-fixes-ccb7c567c755 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH 0/3] ASoC: mt6358: Fixes from an initial glance at a kselftest run
by Mark Brown 28 Feb '23

28 Feb '23

This is a collection of fixes I came up after glancing through an initial test run with the snappily named Kukui Jacuzzi SKU16 Chromebook on KernelCI. There are more issues flagged, this is just what I fixed thus far. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (3): ASoC: mt6358: Fix event generation for wake on voice stage 2 switch ASoC: mt6358: Validate Wake on Voice 2 writes ASoC: mt6358: Remove undefined HPx Mux enumeration values sound/soc/codecs/mt6358.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) --- base-commit: d2980d8d826554fa6981d621e569a453787472f8 change-id: 20230224-asoc-mt6358-quick-fixes-71b4ea6b8f13 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[PATCH 0/4] ASoC: mt8192: Fixes from initial glance at kselftest run
by Mark Brown 28 Feb '23

28 Feb '23

This is a collection of fixes I came up after glancing through an initial test run with the Spherion Chromebook on KernelCI. There are more issues flagged, this is just what I fixed thus far - the volume controls on the MT6359 have issues for example, and a lot of controls aren't marked as Switches like they should be. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Mark Brown (4): ASoC: mt8192: Remove spammy log messages ASoC: mt8192: Fix event generation for controls ASoC: mt8192: Report an error if when an invalid sidetone gain is written ASoC: mt8192: Fix range for sidetone positive gain sound/soc/mediatek/mt8192/mt8192-dai-adda.c | 58 ++++++++++------------------- 1 file changed, 19 insertions(+), 39 deletions(-) --- base-commit: b361d5d2464a88184f6e17a6462719ba79180b1a change-id: 20230223-asoc-mt8192-quick-fixes-8f3e0a187226 Best regards, -- Mark Brown <broonie(a)kernel.org>

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[RFC PATCH v2 00/22] Introduce QC USB SND audio offloading support
by Wesley Cheng 28 Feb '23

28 Feb '23

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. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. 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 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... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Mathias Nyman (4): xhci: fix event ring segment table related masks and variables in header xhci: remove xhci_test_trb_in_td_math early development check xhci: Refactor interrupter code for initial multi interrupter support. xhci: Add support to allocate several interrupters Wesley Cheng (18): usb: xhci: Add XHCI APIs to support USB offloading usb: host: xhci-mem: Cleanup pending secondary event ring events ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: Add USB backend ASoC driver for Q6 sound: usb: card: Introduce USB SND platform op callbacks sound: usb: Export USB SND APIs for modules dt-bindings: usb: dwc3: Add snps,num-hc-interrupters definition usb: dwc3: Add DT parameter to specify maximum number of interrupters sound: usb: Introduce QC USB SND offloading support sound: usb: card: Check for support for requested audio format sound: soc: soc-usb: Add PCM format check API for USB backend sound: soc: qcom: qusb6: Ensure PCM format is supported by USB audio device sound: usb: Prevent starting of audio stream if in use ASoC: dt-bindings: Add Q6USB backend bindings ASoC: dt-bindings: Update example for enabling USB offload on SM8250 .../bindings/sound/qcom,q6usb-dais.yaml | 55 + .../bindings/sound/qcom,sm8250.yaml | 13 + .../devicetree/bindings/usb/snps,dwc3.yaml | 12 + drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-mem.c | 471 +++-- drivers/usb/host/xhci-ring.c | 68 +- drivers/usb/host/xhci.c | 199 +- drivers/usb/host/xhci.h | 81 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 94 + include/sound/pcm_params.h | 4 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 36 + sound/core/oss/pcm_oss.c | 58 - sound/core/pcm_lib.c | 65 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 48 + sound/soc/qcom/qdsp6/q6afe.c | 185 +- sound/soc/qcom/qdsp6/q6afe.h | 46 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 8 + sound/soc/qcom/qdsp6/q6usb.c | 234 +++ sound/soc/soc-usb.c | 215 ++ sound/usb/Kconfig | 14 + sound/usb/Makefile | 2 +- sound/usb/card.c | 58 + sound/usb/card.h | 29 + sound/usb/endpoint.c | 2 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 28 +- sound/usb/pcm.h | 12 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1789 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 41 files changed, 4546 insertions(+), 410 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb-dais.yaml create mode 100644 include/linux/usb/xhci-intr.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/soc-usb.c 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 V4 0/8] Add SoundWire support for AMD platforms
by Vijendar Mukunda 28 Feb '23

28 Feb '23

ACP IP(v6.x) block has two SoundWire manager instance support. This patchset adds support for AMD SoundWire manager driver. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> changes since v3: - add usleep_range() in command/response implementation. - add usleep_range() in clock stop sequence. - modify usleep_range() values throughout the code. - remove unncessary debug statement from code. - update comment in probe() call. - fix timeout condition checks in the code. - drop "ret" variable in amd_disable_sdw_manager(). Changes since v2: - Remove useless variable initializations. - Add helper function to interpret peripheral status. - Move runtime pm sequence to probe_work workqueue. - Use string "SoundWire" instead of "soundwire" in code. - Update comments in interrupt handler and probe sequence. - Rename "sdw_lock" as "acp_sdw_lock". - Remove __func__ from dev_dbg statements. Changes since v1: - Drop asoc tree based patches. will send asoc patches as a separate series. - Fixed double space errors. - Use dev instead of pci->dev. - Use SoundWire manager terminology. - Remove amd_sdw_compute_slave_ports() function and use exported sdw_compute_slave_ports() function. - Remove unused variable "num_ports" from amd_manager structure. - Drop startup and shutdown dai callbacks. - Drop reset_page_addr callback. - Use relative address offset to program SoundWire manager registers throughout the code. - Separate wake enable interrupt handling from slave status handling logic. - Use acp_mmio to program ACP common registers. - Use dai_runtime_array implementation in dai_ops. - Refactor port_ops callbacks. - Add comments in port_ops callbacks. - Add retry count logic in irq thread to address faulty case. - Add helper function to interpret command response. - Add generic bandwidth allocation dependency in Kconfig options. - Add comments for AMD SoundWire power modes. - Add missing timeout check in amd_init_sdw_manager callback. - Declare frameshape parameters in probe call. - Handle error case in clock stop sequence. - Add comments in pm_prepare and pm_ops callbacks. Vijendar Mukunda (8): soundwire: export sdw_compute_slave_ports() function soundwire: amd: Add support for AMD Manager driver soundwire: amd: register SoundWire manager dai ops soundwire: amd: enable build for AMD SoundWire manager driver soundwire: amd: add SoundWire manager interrupt handling soundwire: amd: add runtime pm ops for AMD SoundWire manager driver soundwire: amd: handle SoundWire wake enable interrupt soundwire: amd: add pm_prepare callback and pm ops support drivers/soundwire/Kconfig | 10 + drivers/soundwire/Makefile | 4 + drivers/soundwire/amd_manager.c | 1321 +++++++++++++++++ drivers/soundwire/amd_manager.h | 275 ++++ drivers/soundwire/bus.h | 9 + .../soundwire/generic_bandwidth_allocation.c | 12 +- include/linux/soundwire/sdw_amd.h | 109 ++ 7 files changed, 1731 insertions(+), 9 deletions(-) create mode 100644 drivers/soundwire/amd_manager.c create mode 100644 drivers/soundwire/amd_manager.h create mode 100644 include/linux/soundwire/sdw_amd.h -- 2.34.1

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