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[PATCH v5 0/7] ASoC: codecs: wcd937x: add wcd937x audio codec support
by Mohammad Rafi Shaik 11 Jun '24

11 Jun '24

This patchset adds support for Qualcomm WCD9370/WCD9375 codec. Qualcomm WCD9370/WCD9375 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 3 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 6 DMICs and MBHC. For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on QCM6490 IDP device. Changes since v4: - Removed volatile/read-only registers from defaults list - Added wcd939x_volatile_register() with only volatile registers - Added a wcd939x_readable_register() with read-only and read-write registers, so cache does it's job - Fixed Spurious events for mixer controls and validated with mixer selftest tool - Used TLV instead of enum for ear_pa_gain mixer control - Used enum constraints instead of OneOf in dt-binding patch - Added vdd-px supply property as non optional in dt-binding patch - Reworked and done driver cleanup Changes since v3: - Fixed dt binding check errors. - Added constraints on values in v4-0001 binding patch as suggested by Krzysztof - Change the patch sequence soundwire driver first then codec driver - Added missing .remove soundwire driver function - Reworked and done driver cleanup Changes since v2: - Used common qcom,wcd93xx-common.yaml. removed duplicate properties. - Merged bindings patches "v2-0001" and "v2-0003" in single patch for easy review. - Fixed dt binding check errors. - Added missing "qcom,wcd9375-codec" in v3-0001 dt binding patch. - Added constraints on values in v3-0001 binding patch as suggested by Krzysztof - Fix the typo mistake in v2 cover letter Changes since v1: - Split the patch per driver for easier review as suggested by Krzysztof - Used devm_gpiod_get api to get reset gpio as suggested by Krzysztof Prasad Kumpatla (7): ASoC: dt-bindings: document wcd937x Audio Codec ASoC: codecs: wcd937x-sdw: add SoundWire driver ASoC: codecs: wcd937x: add wcd937x codec driver ASoC: codecs: wcd937x: add basic controls ASoC: codecs: wcd937x: add playback dapm widgets ASoC: codecs: wcd937x: add capture dapm widgets ASoC: codecs: wcd937x: add audio routing and Kconfig .../bindings/sound/qcom,wcd937x-sdw.yaml | 91 + .../bindings/sound/qcom,wcd937x.yaml | 82 + sound/soc/codecs/Kconfig | 20 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/wcd937x-sdw.c | 1139 +++++++ sound/soc/codecs/wcd937x.c | 3029 +++++++++++++++++ sound/soc/codecs/wcd937x.h | 653 ++++ 7 files changed, 5021 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml create mode 100644 sound/soc/codecs/wcd937x-sdw.c create mode 100644 sound/soc/codecs/wcd937x.c create mode 100644 sound/soc/codecs/wcd937x.h base-commit: 124cfbcd6d185d4f50be02d5f5afe61578916773 -- 2.25.1

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[PATCH] ASoC: samsung: midas_wm1811: Fix error code in probe()
by Dan Carpenter 11 Jun '24

11 Jun '24

This accidentally returns success instead of -EINVAL. Fixes: c91d0c2e198d ("ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/soc/samsung/midas_wm1811.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/midas_wm1811.c b/sound/soc/samsung/midas_wm1811.c index 3a269c7de169..bbfe5fef59af 100644 --- a/sound/soc/samsung/midas_wm1811.c +++ b/sound/soc/samsung/midas_wm1811.c @@ -623,7 +623,7 @@ static int midas_probe(struct platform_device *pdev) if (channel_type != IIO_VOLTAGE) { dev_err(dev, "ADC channel is not voltage\n"); - return ret; + return -EINVAL; } priv->gpio_headset_key = devm_gpiod_get(dev, "headset-key", -- 2.43.0

3 2

[PATCH v2 0/2] Fixes for McASP and dmaengine_pcm
by Jai Luthra 11 Jun '24

11 Jun '24

This series fixes two patches: 1. Fix the dmaengine API usage by calling dmaengine_synchronize() after dmaengine_terminate_async() when xrun events occur in application 2. Use the McASP AFIFO property from DT to refine the period size, instead of hardcoding minimum to 64 samples Signed-off-by: Jai Luthra <j-luthra(a)ti.com> --- Changes in v2: - Fix compiler warning for prepare callback by marking it static - Pass numevt directly to hw_rule_min_periodsize() - Link to v1: https://lore.kernel.org/r/20240604-asoc_next-v1-0-e895c88e744d@ti.com --- Jai Luthra (2): ALSA: dmaengine: Synchronize dma channel in prepare() ASoC: ti: davinci-mcasp: Set min period size using FIFO config include/sound/dmaengine_pcm.h | 1 + sound/core/pcm_dmaengine.c | 10 ++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 8 ++++++++ sound/soc/ti/davinci-mcasp.c | 9 +++++++-- 4 files changed, 26 insertions(+), 2 deletions(-) --- base-commit: d97496ca23a2d4ee80b7302849404859d9058bcd change-id: 20240604-asoc_next-c063fcc190c6 Best regards, -- Jai Luthra <j-luthra(a)ti.com>

4 7

[PATCH] bytcr_rt5640 : inverse jack detect for Archos 101 cesium
by Thomas GENTY 11 Jun '24

11 Jun '24

When headphones are plugged in, they appear absent; when they are removed, they appear present. Add a specific entry in bytcr_rt5640 for this device Signed-off-by: Thomas GENTY <tomlohave(a)gmail.com> Reviewed-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index b41a1147f1c3..a64d1989e28a 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -610,6 +610,17 @@ static const struct dmi_system_id byt_rt5640_quirk_table[] = { BYT_RT5640_SSP0_AIF1 | BYT_RT5640_MCLK_EN), }, + { + .matches = { + DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), + DMI_EXACT_MATCH(DMI_PRODUCT_NAME, "ARCHOS 101 CESIUM"), + }, + .driver_data = (void *)(BYTCR_INPUT_DEFAULTS | + BYT_RT5640_JD_NOT_INV | + BYT_RT5640_DIFF_MIC | + BYT_RT5640_SSP0_AIF1 | + BYT_RT5640_MCLK_EN), + }, { .matches = { DMI_EXACT_MATCH(DMI_SYS_VENDOR, "ARCHOS"), -- 2.45.2

3 3

[PATCH 0/3] ACPI/ALSA/soundwire: add acpi_get_local_u64_address() helper
by Pierre-Louis Bossart 11 Jun '24

11 Jun '24

The acpi_get_local_address() helper assumes a 32-bit ADR is used. If we want to use this helper for SoundWire/SDCA ASoC codecs, we need an extension where the native 64-bits are used. This patchset suggests a new helper, acpi_get_local_address() may be renamed if desired in a folow-up patch. The path of least resistance would be to merge this patchset in the ASoC tree, since I have additional changes for ASoC/SDCA (SoundWire Device Class) that depend on the new helper. Pierre-Louis Bossart (3): ACPI: utils: introduce acpi_get_local_u64_address() soundwire: slave: simplify code with acpi_get_local_u64_address() ALSA: hda: intel-sdw-acpi: use acpi_get_local_u64_address() drivers/acpi/utils.c | 22 ++++++++++++++++------ drivers/soundwire/slave.c | 13 ++++--------- include/linux/acpi.h | 1 + sound/hda/intel-sdw-acpi.c | 6 +++--- 4 files changed, 24 insertions(+), 18 deletions(-) -- 2.43.0

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[RESEND v5 0/7] ASoC: codecs: wcd937x: add wcd937x audio codec support
by Mohammad Rafi Shaik 11 Jun '24

11 Jun '24

This patchset adds support for Qualcomm WCD9370/WCD9375 codec. Qualcomm WCD9370/WCD9375 Codec is a standalone Hi-Fi audio codec IC connected over SoundWire. This device has two SoundWire devices, RX and TX respectively supporting 3 x ADCs, ClassH, Ear, Aux PA, 2xHPH, 6 DMICs and MBHC. For codec driver to be functional it would need both tx and rx Soundwire devices to be up and this is taken care by using device component framework and device-links are used to ensure proper pm dependencies. Ex tx does not enter suspend before rx or codec is suspended. This patchset along with other SoundWire patches on the list have been tested on QCM6490 IDP device. Changes since v4: - Removed volatile/read-only registers from defaults list - Added wcd939x_volatile_register() with only volatile registers - Added a wcd939x_readable_register() with read-only and read-write registers, so cache does it's job - Fixed Spurious events for mixer controls and validated with mixer selftest tool - Used TLV instead of enum for ear_pa_gain mixer control - Used enum constraints instead of OneOf in dt-binding patch - Added vdd-px supply property as non optional in dt-binding patch - Reworked and done driver cleanup Changes since v3: - Fixed dt binding check errors. - Added constraints on values in v4-0001 binding patch as suggested by Krzysztof - Change the patch sequence soundwire driver first then codec driver - Added missing .remove soundwire driver function - Reworked and done driver cleanup Changes since v2: - Used common qcom,wcd93xx-common.yaml. removed duplicate properties. - Merged bindings patches "v2-0001" and "v2-0003" in single patch for easy review. - Fixed dt binding check errors. - Added missing "qcom,wcd9375-codec" in v3-0001 dt binding patch. - Added constraints on values in v3-0001 binding patch as suggested by Krzysztof - Fix the typo mistake in v2 cover letter Changes since v1: - Split the patch per driver for easier review as suggested by Krzysztof - Used devm_gpiod_get api to get reset gpio as suggested by Krzysztof Prasad Kumpatla (7): ASoC: dt-bindings: document wcd937x Audio Codec ASoC: codecs: wcd937x-sdw: add SoundWire driver ASoC: codecs: wcd937x: add wcd937x codec driver ASoC: codecs: wcd937x: add basic controls ASoC: codecs: wcd937x: add playback dapm widgets ASoC: codecs: wcd937x: add capture dapm widgets ASoC: codecs: wcd937x: add audio routing and Kconfig .../bindings/sound/qcom,wcd937x-sdw.yaml | 91 + .../bindings/sound/qcom,wcd937x.yaml | 82 + sound/soc/codecs/Kconfig | 20 + sound/soc/codecs/Makefile | 7 + sound/soc/codecs/wcd937x-sdw.c | 1139 +++++++ sound/soc/codecs/wcd937x.c | 3029 +++++++++++++++++ sound/soc/codecs/wcd937x.h | 653 ++++ 7 files changed, 5021 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x-sdw.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,wcd937x.yaml create mode 100644 sound/soc/codecs/wcd937x-sdw.c create mode 100644 sound/soc/codecs/wcd937x.c create mode 100644 sound/soc/codecs/wcd937x.h base-commit: 3689b0ef08b70e4e03b82ebd37730a03a672853a -- 2.25.1

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[PATCH v22 00/38] Introduce QC USB SND audio offloading support
by Wesley Cheng 11 Jun '24

11 Jun '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... 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 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. 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. 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 (36): 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 ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 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 usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support 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 ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Introduce SND kcontrols to track USB offloading state ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request ASoC: usb: Rediscover USB SND devices on USB port add ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 603 ++++++ 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 | 418 ++++ 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 | 188 ++ 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 | 417 ++++ sound/soc/qcom/sm8250.c | 23 +- sound/soc/qcom/usb_offload_utils.c | 55 + sound/soc/qcom/usb_offload_utils.h | 29 + sound/soc/soc-usb.c | 684 ++++++ 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 | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1915 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 6327 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

2 42

[PATCH 0/5] ASoC: simple-card / audio-graph:
by Kuninori Morimoto 11 Jun '24

11 Jun '24

Hi Mark, Rob, Krzysztof Cc Kochetkov This patch-set adds link-trigger-order to Simple-Card / Audio-Graph-Card. Kuninori Morimoto (5): ASoC: audio-graph-port: add link-trigger-order ASoC: simple-card-utils: add link-trigger-order support ASoC: simple-audio-card: add link-trigger-order support ASoC: audio-graph-card: add link-trigger-order support ASoC: audio-graph-card2: add link-trigger-order support .../bindings/sound/audio-graph-port.yaml | 9 +++ include/dt-bindings/sound/audio-graph.h | 21 ++++++ include/sound/simple_card_utils.h | 4 ++ sound/soc/generic/audio-graph-card.c | 13 ++++ sound/soc/generic/audio-graph-card2.c | 13 ++++ sound/soc/generic/simple-card-utils.c | 71 +++++++++++++++++++ sound/soc/generic/simple-card.c | 10 +++ 7 files changed, 141 insertions(+) create mode 100644 include/dt-bindings/sound/audio-graph.h -- 2.43.0

2 8

[PATCH 1/2] sof-soundwire: Add missing match for cs42l43 speakers
by Charles Keepax 10 Jun '24

10 Jun '24

The cs42l43 reports as cs42l43-spk in the speaker case, the regex in sof-soundwire needs updated to recognise that. Fixes: 035d9206cffd ("sof-soundwire: Add basic support for cs42l43's speaker") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- ucm2/sof-soundwire/sof-soundwire.conf | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ucm2/sof-soundwire/sof-soundwire.conf b/ucm2/sof-soundwire/sof-soundwire.conf index 31492e3..2f61d11 100644 --- a/ucm2/sof-soundwire/sof-soundwire.conf +++ b/ucm2/sof-soundwire/sof-soundwire.conf @@ -21,7 +21,7 @@ Define { DefineRegex { SpeakerCodec { - Regex " spk:([a-z0-9]+(-sdca)?)" + Regex " spk:([a-z0-9]+((-sdca)|(-spk))?)" String "${CardComponents}" } SpeakerChannels { -- 2.39.2

1 1

[PATCH] ASoC: fsl: Add i2s and pcm drivers for LPC32xx CPUs
by Piotr Wojtaszczyk 10 Jun '24

10 Jun '24

This driver was ported from an old version in linux 2.6.27 and adjusted for the new ASoC framework and DMA API. Signed-off-by: Piotr Wojtaszczyk <piotr.wojtaszczyk(a)timesys.com> --- .../bindings/sound/nxp,lpc3220-i2s.yaml | 50 +++ arch/arm/boot/dts/lpc32xx.dtsi | 4 + arch/arm/mach-lpc32xx/phy3250.c | 60 +++ sound/soc/fsl/Kconfig | 7 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/lpc3xxx-i2s.c | 411 ++++++++++++++++++ sound/soc/fsl/lpc3xxx-i2s.h | 94 ++++ sound/soc/fsl/lpc3xxx-pcm.c | 75 ++++ 8 files changed, 703 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml 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 diff --git a/Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml b/Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml new file mode 100644 index 000000000000..e41330b6775c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nxp,lpc3220-i2s.yaml @@ -0,0 +1,50 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/nxp,lpc3220-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: NXP LPC32XX I2S Controller + +description: + The block adds I2S and PCM drivers for LPC32XX + +properties: + compatible: + enum: + - nxp,lpc3220-i2s + + reg: + maxItems: 1 + + clocks: + items: + - description: input clock of the peripheral. + + clock-names: + items: + - const: i2s_clk + + interrupts: + maxItems: 1 + +required: + - compatible + - reg + - clocks + - clock-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/clock/lpc32xx-clock.h> + + i2s0: i2s@20094000 { + compatible = "nxp,lpc3220-i2s"; + reg = <0x20094000 0x1000>; + clocks = <&clk LPC32XX_CLK_I2S0>; + clock-names = "i2s_clk"; + }; + +... diff --git a/arch/arm/boot/dts/lpc32xx.dtsi b/arch/arm/boot/dts/lpc32xx.dtsi index c87066d6c995..dc5738f2b42d 100644 --- a/arch/arm/boot/dts/lpc32xx.dtsi +++ b/arch/arm/boot/dts/lpc32xx.dtsi @@ -221,6 +221,8 @@ spi2: spi@20090000 { i2s0: i2s@20094000 { compatible = "nxp,lpc3220-i2s"; reg = <0x20094000 0x1000>; + clocks = <&clk LPC32XX_CLK_I2S0>; + clock-names = "i2s_clk"; status = "disabled"; }; @@ -237,6 +239,8 @@ sd: sd@20098000 { i2s1: i2s@2009c000 { compatible = "nxp,lpc3220-i2s"; reg = <0x2009c000 0x1000>; + clocks = <&clk LPC32XX_CLK_I2S1>; + clock-names = "i2s_clk"; status = "disabled"; }; diff --git a/arch/arm/mach-lpc32xx/phy3250.c b/arch/arm/mach-lpc32xx/phy3250.c index 66701bf43248..b866b9a75558 100644 --- a/arch/arm/mach-lpc32xx/phy3250.c +++ b/arch/arm/mach-lpc32xx/phy3250.c @@ -9,6 +9,7 @@ */ #include <linux/amba/pl08x.h> +#include <linux/amba/pl022.h> #include <linux/mtd/lpc32xx_mlc.h> #include <linux/mtd/lpc32xx_slc.h> #include <linux/of_platform.h> @@ -29,6 +30,46 @@ static struct pl08x_channel_data pl08x_slave_channels[] = { .max_signal = 12, .periph_buses = PL08X_AHB1, }, + { + .bus_id = "i2s-tx", + .min_signal = 13, + .max_signal = 13, + .periph_buses = PL08X_AHB1, + }, + { + .bus_id = "i2s-rx", + .min_signal = 0, + .max_signal = 0, + .periph_buses = PL08X_AHB1, + }, + { + .bus_id = "ssp0-tx", + .min_signal = 15, + .max_signal = 15, + .muxval = 1, + .periph_buses = PL08X_AHB1, + }, + { + .bus_id = "ssp0-rx", + .min_signal = 14, + .max_signal = 14, + .muxval = 1, + .periph_buses = PL08X_AHB1, + }, + { + .bus_id = "ssp1-tx", + .min_signal = 11, + .max_signal = 11, + .muxval = 1, + .periph_buses = PL08X_AHB1, + }, + { + .bus_id = "ssp1-rx", + .min_signal = 3, + .max_signal = 3, + .muxval = 1, + .periph_buses = PL08X_AHB1, + }, }; static int pl08x_get_signal(const struct pl08x_channel_data *cd) @@ -60,12 +101,31 @@ static struct lpc32xx_mlc_platform_data lpc32xx_mlc_data = { .dma_filter = pl08x_filter_id, }; +static struct pl022_ssp_controller lpc32xx_ssp_data[] = { + { + .bus_id = 0, + .enable_dma = 0, + .dma_filter = pl08x_filter_id, + .dma_tx_param = "ssp0-tx", + .dma_rx_param = "ssp0-rx", + }, + { + .bus_id = 1, + .enable_dma = 0, + .dma_filter = pl08x_filter_id, + .dma_tx_param = "ssp1-tx", + .dma_rx_param = "ssp1-rx", + } +}; + static const struct of_dev_auxdata lpc32xx_auxdata_lookup[] __initconst = { OF_DEV_AUXDATA("arm,pl080", 0x31000000, "pl08xdmac", &pl08x_pd), OF_DEV_AUXDATA("nxp,lpc3220-slc", 0x20020000, "20020000.flash", &lpc32xx_slc_data), OF_DEV_AUXDATA("nxp,lpc3220-mlc", 0x200a8000, "200a8000.flash", &lpc32xx_mlc_data), + OF_DEV_AUXDATA("arm,pl022", 0x20084000, NULL, &lpc32xx_ssp_data[0]), + OF_DEV_AUXDATA("arm,pl022", 0x2008c000, NULL, &lpc32xx_ssp_data[1]), { } }; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 258306d8ce51..5f25aa7d632c 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -132,6 +132,13 @@ config SND_SOC_FSL_RPMSG This option is only useful for out-of-tree drivers since in-tree drivers select it automatically. +config SND_SOC_FSL_LPC3XXX + tristate "SoC Audio for NXP LPC32XX CPUs" + depends on ARCH_LPC32XX && SND_SOC + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M if you want to add support for the LPC3XXX I2S interface. + config SND_SOC_IMX_PCM_DMA tristate select SND_SOC_GENERIC_DMAENGINE_PCM diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 9f743fcc0e02..988dc98cdfc6 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -29,6 +29,7 @@ snd-soc-fsl-easrc-objs := fsl_easrc.o snd-soc-fsl-xcvr-objs := fsl_xcvr.o snd-soc-fsl-aud2htx-objs := fsl_aud2htx.o snd-soc-fsl-rpmsg-objs := fsl_rpmsg.o +snd-soc-fsl-lpc3xxx-objs := lpc3xxx-pcm.o lpc3xxx-i2s.o obj-$(CONFIG_SND_SOC_FSL_AUDMIX) += snd-soc-fsl-audmix.o obj-$(CONFIG_SND_SOC_FSL_ASOC_CARD) += snd-soc-fsl-asoc-card.o @@ -45,6 +46,7 @@ obj-$(CONFIG_SND_SOC_POWERPC_DMA) += snd-soc-fsl-dma.o obj-$(CONFIG_SND_SOC_FSL_XCVR) += snd-soc-fsl-xcvr.o obj-$(CONFIG_SND_SOC_FSL_AUD2HTX) += snd-soc-fsl-aud2htx.o obj-$(CONFIG_SND_SOC_FSL_RPMSG) += snd-soc-fsl-rpmsg.o +obj-$(CONFIG_SND_SOC_FSL_LPC3XXX) += snd-soc-fsl-lpc3xxx.o # MPC5200 Platform Support obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o diff --git a/sound/soc/fsl/lpc3xxx-i2s.c b/sound/soc/fsl/lpc3xxx-i2s.c new file mode 100644 index 000000000000..853284eb14f1 --- /dev/null +++ b/sound/soc/fsl/lpc3xxx-i2s.c @@ -0,0 +1,411 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * Author: Kevin Wells <kevin.wells(a)nxp.com> + * + * Copyright (C) 2008 NXP Semiconductors + * Copyright 2023 Timesys Corporation <piotr.wojtaszczyk(a)timesys.com> + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/interrupt.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> +#include <linux/io.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> + +#include "lpc3xxx-i2s.h" + +#define I2S_PLAYBACK_FLAG 0x1 +#define I2S_CAPTURE_FLAG 0x2 + +#define LPC3XXX_I2S_RATES ( \ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + +#define LPC3XXX_I2S_FORMATS ( \ + SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static u32 absd32(u32 v1, u32 v2) +{ + if (v1 > v2) { + return v1 - v2; + } + return v2 - v1; +} + +static void __lpc3xxx_find_clkdiv(u32 *clkx, u32 *clky, int freq, + int xbytes, u32 clkrate) +{ + u32 i2srate; + u32 idxx, idyy; + u32 savedbitclkrate, diff, trate, baseclk; + + /* Adjust rate for sample size (bits) and 2 channels and offset for + * divider in clock output + */ + i2srate = (freq / 100) * 2 * (8 * xbytes); + i2srate = i2srate << 1; + clkrate = clkrate / 100; + baseclk = clkrate; + *clkx = 1; + *clky = 1; + + /* Find the best divider */ + *clkx = *clky = 0; + savedbitclkrate = 0; + diff = ~0; + for (idxx = 1; idxx < 0xFF; idxx++) { + for (idyy = 1; idyy < 0xFF; idyy++) { + trate = (baseclk * idxx) / idyy; + if (absd32(trate, i2srate) < diff) { + diff = absd32(trate, i2srate); + savedbitclkrate = trate; + *clkx = idxx; + *clky = idyy; + } + } + } +} + +static int lpc3xxx_i2s_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) +{ + struct lpc3xxx_i2s_info *i2s_info_p = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = i2s_info_p->dev; + u32 flag; + int ret = 0; + + mutex_lock(&i2s_info_p->lock); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + flag = I2S_PLAYBACK_FLAG; + } else { + flag = I2S_CAPTURE_FLAG; + } + + if (flag & i2s_info_p->streams_in_use) { + dev_warn(dev, "I2S channel is busy\n"); + ret = -EBUSY; + goto lpc32xx_unlock; + } + + if (i2s_info_p->streams_in_use == 0) { + ret = clk_prepare_enable(i2s_info_p->clk); + if (ret) { + dev_err(dev, "Can't enable clock, err=%d\n", ret); + goto lpc32xx_unlock; + } + } + + i2s_info_p->streams_in_use |= flag; + +lpc32xx_unlock: + mutex_unlock(&i2s_info_p->lock); + return ret; +} + +static void lpc3xxx_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) +{ + struct lpc3xxx_i2s_info *i2s_info_p = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regs = i2s_info_p->regs; + const u32 stop_bits = (I2S_RESET | I2S_STOP); + u32 flag; + + mutex_lock(&i2s_info_p->lock); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + flag = I2S_PLAYBACK_FLAG; + regmap_write(regs, I2S_TX_RATE, 0); + regmap_update_bits(regs, I2S_DAO, stop_bits, stop_bits); + } else { + flag = I2S_CAPTURE_FLAG; + regmap_write(regs, I2S_RX_RATE, 0); + regmap_update_bits(regs, I2S_DAI, stop_bits, stop_bits); + } + i2s_info_p->streams_in_use &= ~flag; + + if (i2s_info_p->streams_in_use == 0) { + clk_disable_unprepare(i2s_info_p->clk); + } + + mutex_unlock(&i2s_info_p->lock); +} + +static int lpc3xxx_i2s_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct lpc3xxx_i2s_info *i2s_info_p = snd_soc_dai_get_drvdata(cpu_dai); + + /* Will use in HW params later */ + i2s_info_p->freq = freq; + + return 0; +} + +static int lpc3xxx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct lpc3xxx_i2s_info *i2s_info_p = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = i2s_info_p->dev; + + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_I2S) { + dev_warn(dev, "unsupported bus format %d\n", fmt); + return -EINVAL; + } + return 0; +} + +static int lpc3xxx_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *cpu_dai) +{ + struct lpc3xxx_i2s_info *i2s_info_p = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = i2s_info_p->dev; + struct regmap *regs = i2s_info_p->regs; + int xfersize; + u32 tmp, clkx, clky; + + tmp = I2S_RESET | I2S_STOP; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + tmp |= I2S_WW8 | I2S_WS_HP(I2S_WW8_HP); + xfersize = 1; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + tmp |= I2S_WW16 | I2S_WS_HP(I2S_WW16_HP); + xfersize = 2; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + tmp |= I2S_WW32 | I2S_WS_HP(I2S_WW32_HP); + xfersize = 4; + break; + + default: + dev_warn(dev, "Unsupported audio data format %d\n", + params_format(params)); + return -EINVAL; + } + + if (params_channels(params) == 1) { + tmp |= I2S_MONO; + } + + __lpc3xxx_find_clkdiv(&clkx, &clky, i2s_info_p->freq, xfersize, i2s_info_p->clkrate); + + dev_dbg(dev, "Stream : %s\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? "playback" : "capture"); + dev_dbg(dev, "Desired clock rate : %d\n", i2s_info_p->freq); + dev_dbg(dev, "Base clock rate : %d\n", i2s_info_p->clkrate); + dev_dbg(dev, "Transfer size (bytes) : %d\n", xfersize); + dev_dbg(dev, "Clock divider (x) : %d\n", clkx); + dev_dbg(dev, "Clock divider (y) : %d\n", clky); + dev_dbg(dev, "Channels : %d\n", params_channels(params)); + dev_dbg(dev, "Data format : %s\n", "I2S"); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_write(regs, I2S_DMA1, I2S_DMA1_TX_EN | I2S_DMA0_TX_DEPTH(4)); + regmap_write(regs, I2S_TX_RATE, (clkx << 8) | clky); + regmap_write(regs, I2S_DAO, tmp); + } else { + regmap_write(regs, I2S_DMA0, I2S_DMA0_RX_EN | I2S_DMA1_RX_DEPTH(4)); + regmap_write(regs, I2S_RX_RATE, (clkx << 8) | clky); + regmap_write(regs, I2S_DAI, tmp); + } + + return 0; +} + +static int lpc3xxx_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) +{ + return 0; +} + +static int lpc3xxx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *cpu_dai) +{ + struct lpc3xxx_i2s_info *i2s_info_p = snd_soc_dai_get_drvdata(cpu_dai); + struct regmap *regs = i2s_info_p->regs; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(regs, I2S_DAO, I2S_STOP, I2S_STOP); + } else { + regmap_update_bits(regs, I2S_DAI, I2S_STOP, I2S_STOP); + } + break; + + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(regs, I2S_DAO, (I2S_RESET | I2S_STOP), 0); + } else { + regmap_update_bits(regs, I2S_DAI, (I2S_RESET | I2S_STOP), 0); + } + break; + default: + ret = -EINVAL; + } + + return ret; +} + +const struct snd_soc_dai_ops lpc3xxx_i2s_dai_ops = { + .startup = lpc3xxx_i2s_startup, + .shutdown = lpc3xxx_i2s_shutdown, + .prepare = lpc3xxx_i2s_prepare, + .trigger = lpc3xxx_i2s_trigger, + .hw_params = lpc3xxx_i2s_hw_params, + .set_sysclk = lpc3xxx_i2s_set_dai_sysclk, + .set_fmt = lpc3xxx_i2s_set_dai_fmt, +}; + +static int lpc3xxx_i2s_dai_probe(struct snd_soc_dai *dai) +{ + struct lpc3xxx_i2s_info *i2s_info_p = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, &i2s_info_p->playback_dma_config, + &i2s_info_p->capture_dma_config); + return 0; +} + +struct snd_soc_dai_driver lpc3xxx_i2s_dai_driver = { + .probe = lpc3xxx_i2s_dai_probe, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = LPC3XXX_I2S_RATES, + .formats = LPC3XXX_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = LPC3XXX_I2S_RATES, + .formats = LPC3XXX_I2S_FORMATS, + }, + .ops = &lpc3xxx_i2s_dai_ops, + .symmetric_rate = 1, + .symmetric_channels = 1, + .symmetric_sample_bits = 1, +}; + +static const struct snd_soc_component_driver lpc32xx_i2s_component = { + .name = "lpc32xx-i2s", +}; + +static const struct regmap_config lpc32xx_i2s_regconfig = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = I2S_RX_RATE, +}; + +static int lpc32xx_i2s_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct lpc3xxx_i2s_info *i2s_info_p; + struct resource *res; + void __iomem *iomem; + int ret; + + i2s_info_p = devm_kzalloc(dev, sizeof(*i2s_info_p), GFP_KERNEL); + if (!i2s_info_p) + return -ENOMEM; + + platform_set_drvdata(pdev, i2s_info_p); + i2s_info_p->dev = dev; + + iomem = devm_platform_get_and_ioremap_resource(pdev, 0, &res); + if (IS_ERR(iomem)) { + dev_err(dev, "Can't map registers\n"); + return PTR_ERR(iomem); + } + + i2s_info_p->regs = devm_regmap_init_mmio(dev, iomem, &lpc32xx_i2s_regconfig); + if (IS_ERR(i2s_info_p->regs)) { + ret = PTR_ERR(i2s_info_p->regs); + dev_err(dev, "failed to init register map: %d\n", ret); + return ret; + } + + i2s_info_p->clk = devm_clk_get(dev, "i2s_clk"); + if (IS_ERR(i2s_info_p->clk)) { + dev_err(dev, "Can't get clock\n"); + return PTR_ERR(i2s_info_p->clk); + } + + i2s_info_p->clkrate = clk_get_rate(i2s_info_p->clk); + if (i2s_info_p->clkrate == 0) { + dev_err(dev, "Invalid returned clock rate\n"); + goto err_clk_disable; + } + + mutex_init(&i2s_info_p->lock); + + ret = devm_snd_soc_register_component(dev, &lpc32xx_i2s_component, + &lpc3xxx_i2s_dai_driver, 1); + if (ret) { + dev_err(dev, "Can't register cpu_dai component\n"); + goto err_clk_disable; + } + + i2s_info_p->playback_dma_config.addr = (dma_addr_t)(res->start + I2S_TX_FIFO); + i2s_info_p->playback_dma_config.maxburst = 4; + i2s_info_p->playback_dma_config.filter_data = "i2s-tx"; + i2s_info_p->capture_dma_config.addr = (dma_addr_t)(res->start + I2S_RX_FIFO); + i2s_info_p->capture_dma_config.maxburst = 4; + i2s_info_p->capture_dma_config.filter_data = "i2s-rx"; + + ret = lpc3xxx_pcm_register(pdev); + if (ret) { + dev_err(dev, "Can't register pcm component\n"); + goto err_clk_disable; + } + + return 0; + +err_clk_disable: + return ret; +} + +static int lpc32xx_i2s_remove(struct platform_device *pdev) +{ + return 0; +} + +static const struct of_device_id lpc32xx_i2s_match[] = { + { .compatible = "nxp,lpc3220-i2s" }, + {}, +}; +MODULE_DEVICE_TABLE(of, lpc32xx_i2s_match); + +static struct platform_driver lpc32xx_i2s_driver = { + .probe = lpc32xx_i2s_probe, + .remove = lpc32xx_i2s_remove, + .driver = { + .name = "lpc3xxx-i2s", + .of_match_table = of_match_ptr(lpc32xx_i2s_match), + }, +}; + +module_platform_driver(lpc32xx_i2s_driver); + +MODULE_AUTHOR("Kevin Wells <kevin.wells(a)nxp.com>"); +MODULE_AUTHOR("Piotr Wojtaszczyk <piotr.wojtaszczyk(a)timesys.com>"); +MODULE_DESCRIPTION("ASoC LPC3XXX I2S interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/fsl/lpc3xxx-i2s.h b/sound/soc/fsl/lpc3xxx-i2s.h new file mode 100644 index 000000000000..f88ab74cfe41 --- /dev/null +++ b/sound/soc/fsl/lpc3xxx-i2s.h @@ -0,0 +1,94 @@ +/* SPDX-License-Identifier: GPL-2.0-or-later */ +/* + * Author: Kevin Wells <kevin.wells(a)nxp.com> + * + * Copyright (C) 2008 NXP Semiconductors + * Copyright 2023 Timesys Corporation <piotr.wojtaszczyk(a)timesys.com> + */ + +#ifndef __SOUND_SOC_LPC3XXX_I2S_H +#define __SOUND_SOC_LPC3XXX_I2S_H + +#include <linux/types.h> +#include <linux/regmap.h> + +struct lpc3xxx_i2s_info { + struct device *dev; + struct clk *clk; + struct mutex lock; + struct regmap *regs; + u32 streams_in_use; + u32 clkrate; + int freq; + struct snd_dmaengine_dai_dma_data playback_dma_config; + struct snd_dmaengine_dai_dma_data capture_dma_config; +}; + +int lpc3xxx_pcm_register(struct platform_device *pdev); + +#define _SBF(f, v) ((v) << (f)) +#define _BIT(n) _SBF(n, 1) + +/* I2S controller register offsets */ +#define I2S_DAO 0x00 +#define I2S_DAI 0x04 +#define I2S_TX_FIFO 0x08 +#define I2S_RX_FIFO 0x0C +#define I2S_STAT 0x10 +#define I2S_DMA0 0x14 +#define I2S_DMA1 0x18 +#define I2S_IRQ 0x1C +#define I2S_TX_RATE 0x20 +#define I2S_RX_RATE 0x24 + +/* i2s_daO i2s_dai register definitions */ +#define I2S_WW8 _SBF(0, 0) /* Word width is 8bit*/ +#define I2S_WW16 _SBF(0, 1) /* Word width is 16bit*/ +#define I2S_WW32 _SBF(0, 3) /* Word width is 32bit*/ +#define I2S_MONO _BIT(2) /* Mono */ +#define I2S_STOP _BIT(3) /* Stop, diables the access to FIFO, mutes the channel */ +#define I2S_RESET _BIT(4) /* Reset the channel */ +#define I2S_WS_SEL _BIT(5) /* Channel Master(0) or slave(1) mode select*/ +#define I2S_WS_HP(s) _SBF(6, s) /* Word select half period - 1 */ + +#define I2S_MUTE _BIT(15) /* Mute the channel, Transmit channel only */ + +#define I2S_WW32_HP 0x1f /* Word select half period for 32bit word width */ +#define I2S_WW16_HP 0x0f /* Word select half period for 16bit word width */ +#define I2S_WW8_HP 0x7 /* Word select half period for 8bit word width */ + +#define WSMASK_HP 0X7FC /* Mask for WS half period bits */ + +/* i2s_tx_fifo register definitions */ +#define I2S_FIFO_TX_WRITE(d) (d) + +/* i2s_rx_fifo register definitions */ +#define I2S_FIFO_RX_WRITE(d) (d) + +/* i2s_stat register definitions */ +#define I2S_IRQ_STAT _BIT(0) +#define I2S_DMA0_REQ _BIT(1) +#define I2S_DMA1_REQ _BIT(2) + +#define I2S_RX_STATE_MASK 0x0000ff00 +#define I2S_TX_STATE_MASK 0x00ff0000 + +/* i2s_dma0 Configuration register definitions */ +#define I2S_DMA0_RX_EN _BIT(0) /* Enable RX DMA1*/ +#define I2S_DMA0_TX_EN _BIT(1) /* Enable TX DMA1*/ +#define I2S_DMA0_RX_DEPTH(s) _SBF(8, s) /* Set the level for DMA1 RX Request */ +#define I2S_DMA0_TX_DEPTH(s) _SBF(16, s) /* Set the level for DMA1 TX Request */ + +/* i2s_dma1 Configuration register definitions */ +#define I2S_DMA1_RX_EN _BIT(0) /* Enable RX DMA1*/ +#define I2S_DMA1_TX_EN _BIT(1) /* Enable TX DMA1*/ +#define I2S_DMA1_RX_DEPTH(s) _SBF(8, s) /* Set the level for DMA1 RX Request */ +#define I2S_DMA1_TX_DEPTH(s) _SBF(16, s) /* Set the level for DMA1 TX Request */ + +/* i2s_irq register definitions */ +#define I2S_RX_IRQ_EN _BIT(0) /* Enable RX IRQ*/ +#define I2S_TX_IRQ_EN _BIT(1) /* Enable TX IRQ*/ +#define I2S_IRQ_RX_DEPTH(s) _SBF(8, s) /* valid values ar 0 to 7 */ +#define I2S_IRQ_TX_DEPTH(s) _SBF(16, s) /* valid values ar 0 to 7 */ + +#endif diff --git a/sound/soc/fsl/lpc3xxx-pcm.c b/sound/soc/fsl/lpc3xxx-pcm.c new file mode 100644 index 000000000000..2dbc60d06452 --- /dev/null +++ b/sound/soc/fsl/lpc3xxx-pcm.c @@ -0,0 +1,75 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +/* + * Author: Kevin Wells <kevin.wells(a)nxp.com> + * + * Copyright (C) 2008 NXP Semiconductors + * Copyright 2023 Timesys Corporation <piotr.wojtaszczyk(a)timesys.com> + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/slab.h> +#include <linux/dma-mapping.h> +#include <linux/amba/pl08x.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/dmaengine_pcm.h> +#include <sound/soc.h> + +#include "lpc3xxx-i2s.h" + +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_U24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE | \ + SNDRV_PCM_FMTBIT_U32_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) + + +static const struct snd_pcm_hardware lpc3xxx_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME), + .formats = STUB_FORMATS, + .period_bytes_min = 128, + .period_bytes_max = 2048, + .periods_min = 2, + .periods_max = 1024, + .buffer_bytes_max = 128 * 1024 +}; + +static const struct snd_dmaengine_pcm_config lpc3xxx_dmaengine_pcm_config = { + .pcm_hardware = &lpc3xxx_pcm_hardware, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, + .compat_filter_fn = pl08x_filter_id, + .prealloc_buffer_size = 128 * 1024, +}; + +const struct snd_soc_component_driver lpc3xxx_soc_platform_driver = { + .name = "lpc32xx-pcm", +}; + +int lpc3xxx_pcm_register(struct platform_device *pdev) +{ + int ret; + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, &lpc3xxx_dmaengine_pcm_config, + SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); + if (ret) { + dev_err(&pdev->dev, "failed to register dmaengine: %d\n", ret); + return ret; + } + + return devm_snd_soc_register_component(&pdev->dev, &lpc3xxx_soc_platform_driver, + NULL, 0); +} +EXPORT_SYMBOL(lpc3xxx_pcm_register); -- 2.25.1

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