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

[PATCH] ASoC: qcom: q6apm-dai: drop unused 'q6apm_dai_rtd' fields
by Krzysztof Kozlowski 07 May '24

07 May '24

Remove few unused fields from 'struct q6apm_dai_rtd'. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/qcom/qdsp6/q6apm-dai.c | 10 ++-------- 1 file changed, 2 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c index 00bbd291be5c..cee23d4bda6f 100644 --- a/sound/soc/qcom/qdsp6/q6apm-dai.c +++ b/sound/soc/qcom/qdsp6/q6apm-dai.c @@ -70,14 +70,10 @@ struct q6apm_dai_rtd { unsigned int bytes_received; unsigned int copied_total; uint16_t bits_per_sample; - uint16_t source; /* Encoding source bit mask */ - uint16_t session_id; bool next_track; enum stream_state state; struct q6apm_graph *graph; spinlock_t lock; - uint32_t initial_samples_drop; - uint32_t trailing_samples_drop; bool notify_on_drain; }; @@ -720,14 +716,12 @@ static int q6apm_dai_compr_set_metadata(struct snd_soc_component *component, switch (metadata->key) { case SNDRV_COMPRESS_ENCODER_PADDING: - prtd->trailing_samples_drop = metadata->value[0]; q6apm_remove_trailing_silence(component->dev, prtd->graph, - prtd->trailing_samples_drop); + metadata->value[0]); break; case SNDRV_COMPRESS_ENCODER_DELAY: - prtd->initial_samples_drop = metadata->value[0]; q6apm_remove_initial_silence(component->dev, prtd->graph, - prtd->initial_samples_drop); + metadata->value[0]); break; default: ret = -EINVAL; -- 2.43.0

2 1

[PATCH v3] ALSA: usb-audio: Add sampling rates support for Mbox3
by Manuel Barrio Linares 06 May '24

06 May '24

This adds support for all sample rates supported by the hardware,Digidesign Mbox 3 supports: {44100, 48000, 88200, 96000} Fixes syncing clock issues that presented as pops. To test this, without this patch playing 440hz tone produces pops. Clock is now synced between playback and capture interfaces so no more latency drift issue when using pipewire pro-profile. (https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/3900) Signed-off-by: Manuel Barrio Linares <mbarriolinares(a)gmail.com> --- v2: Cleaned up code. Explicit __le32 usage. v3: Get rid of sparse warnings sound/usb/quirks-table.h | 38 +++++++++++++-------- sound/usb/quirks.c | 74 +++++++++++++++++++++++++++++++--------- 2 files changed, 81 insertions(+), 31 deletions(-) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 5d72dc8..73abc38 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3013,21 +3013,28 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 2, .altsetting = 1, .altset_idx = 1, .attributes = 0x00, - .endpoint = 0x01, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_OUT, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, + .sync_ep = USB_RECIP_INTERFACE | USB_DIR_IN, + .sync_iface = 3, + .sync_altsetting = 1, + .sync_ep_idx = 1, + .implicit_fb = 1, } }, { @@ -3035,22 +3042,25 @@ YAMAHA_DEVICE(0x7010, "UB99"), .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = &(const struct audioformat) { .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .fmt_bits = 24, .channels = 4, .iface = 3, .altsetting = 1, .altset_idx = 1, - .endpoint = 0x81, .attributes = 0x00, + .endpoint = USB_RECIP_INTERFACE | USB_DIR_IN, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, .maxpacksize = 0x009c, - .rates = SNDRV_PCM_RATE_48000, - .rate_min = 48000, - .rate_max = 48000, - .nr_rates = 1, + .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000, + .rate_min = 44100, + .rate_max = 96000, + .nr_rates = 4, .rate_table = (unsigned int[]) { - 48000 - } + 44100, 48000, 88200, 96000 + }, + .implicit_fb = 0, } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 09712e6..2f961f0 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -984,21 +984,13 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } -static void mbox3_setup_48_24_magic(struct usb_device *dev) +static void mbox3_setup_defaults(struct usb_device *dev) { /* The Mbox 3 is "little endian" */ /* max volume is: 0x0000. */ /* min volume is: 0x0080 (shown in little endian form) */ - - /* Load 48000Hz rate into buffer */ - u8 com_buff[4] = {0x80, 0xbb, 0x00, 0x00}; - - /* Set 48000Hz sample rate */ - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x0001, &com_buff, 4); //Is this really needed? - snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), - 0x01, 0x21, 0x0100, 0x8101, &com_buff, 4); + u8 com_buff[2]; /* Deactivate Tuner */ /* on = 0x01*/ @@ -1008,6 +1000,8 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 0x01, 0x21, 0x0003, 0x2001, &com_buff, 1); /* Set clock source to Internal (as opposed to S/PDIF) */ + /* Internal = 0x01*/ + /* S/PDIF = 0x02*/ com_buff[0] = 0x01; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 1, 0x21, 0x0100, 0x8001, &com_buff, 1); @@ -1113,9 +1107,11 @@ static void mbox3_setup_48_24_magic(struct usb_device *dev) 1, 0x21, 0x0107, 0x4201, &com_buff, 2); /* Toggle allowing host control */ + /* Not needed com_buff[0] = 0x02; snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + */ /* Do not dim fx returns */ com_buff[0] = 0x00; @@ -1259,26 +1255,27 @@ static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) descriptor_size = le16_to_cpu(get_cfg_desc(config)->wTotalLength); if (descriptor_size != MBOX3_DESCRIPTOR_SIZE) { - dev_err(&dev->dev, "Invalid descriptor size=%d.\n", descriptor_size); + dev_err(&dev->dev, "MBOX3: Invalid descriptor size=%d.\n", descriptor_size); return -ENODEV; } - dev_dbg(&dev->dev, "device initialised!\n"); + dev_dbg(&dev->dev, "MBOX3: device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; if (err < 0) - dev_dbg(&dev->dev, "error usb_get_descriptor: %d\n", err); + dev_dbg(&dev->dev, "MBOX3: error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); if (err < 0) - dev_dbg(&dev->dev, "error usb_reset_configuration: %d\n", err); - dev_dbg(&dev->dev, "mbox3_boot: new boot length = %d\n", + dev_dbg(&dev->dev, "MBOX3: error usb_reset_configuration: %d\n", err); + + dev_dbg(&dev->dev, "MBOX3: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); - mbox3_setup_48_24_magic(dev); - dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + mbox3_setup_defaults(dev); + dev_info(&dev->dev, "MBOX3: Initialized."); return 0; /* Successful boot */ } @@ -1734,6 +1731,46 @@ static int pioneer_djm_set_format_quirk(struct snd_usb_substream *subs, return 0; } +static void mbox3_set_format_quirk(struct snd_usb_substream *subs, + const struct audioformat *fmt) +{ + __le32 buff4 = 0; + u8 buff1 = 0x01; + u32 new_rate = subs->data_endpoint->cur_rate; + u32 current_rate; + + // Get current rate from card and check if changing it is needed + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4); + current_rate = le32_to_cpu(buff4); + dev_dbg(&subs->dev->dev, + "MBOX3: Current configured sample rate: %d", current_rate); + if (current_rate == new_rate) { + dev_dbg(&subs->dev->dev, + "MBOX3: No change needed (current rate:%d == new rate:%d)", + current_rate, new_rate); + return; + } + + // Set new rate + dev_info(&subs->dev->dev, + "MBOX3: Changing sample rate to: %d", new_rate); + buff4 = cpu_to_le32(new_rate); + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &buff4, 4); + + // Set clock source to Internal + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21, 0x0100, 0x8001, &buff1, 1); + + // Check whether the change was successful + buff4 = 0; + snd_usb_ctl_msg(subs->dev, usb_sndctrlpipe(subs->dev, 0), + 0x01, 0x21 | USB_DIR_IN, 0x0100, 0x8101, &buff4, 4); + if (new_rate != le32_to_cpu(buff4)) + dev_warn(&subs->dev->dev, "MBOX3: Couldn't set the sample rate"); +} + void snd_usb_set_format_quirk(struct snd_usb_substream *subs, const struct audioformat *fmt) { @@ -1755,6 +1792,9 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x08e4, 0x0163): /* Pioneer DJM-850 */ pioneer_djm_set_format_quirk(subs, 0x0086); break; + case USB_ID(0x0dba, 0x5000): + mbox3_set_format_quirk(subs, fmt); /* Digidesign Mbox 3 */ + break; } } -- 2.45.0

3 4

[PATCH v3 0/3] ALSA: hda/tas2781: Add tas2781 driver for SPI.
by Baojun Xu 06 May '24

06 May '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> Baojun Xu (3): ALSA: hda/tas2781: Modification for add tas2781 driver for SPI ALSA: hda/tas2781: Main code of tas2781 driver for SPI ALSA: hda/tas2781: Firmware load for tas2781 driver for SPI drivers/acpi/scan.c | 1 + .../platform/x86/serial-multi-instantiate.c | 10 + sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 165 ++ sound/pci/hda/tas2781_hda_spi.c | 1305 ++++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2272 +++++++++++++++++ 8 files changed, 3783 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c -- 2.40.1

3 8

[PATCH v4 0/3] ALSA: hda/tas2781: Add tas2781 driver for SPI.
by Baojun Xu 06 May '24

06 May '24

This patch was used to add TAS2781 devices on SPI support in sound/pci/hda. It use ACPI node descript about parameters of TAS2781 on SPI, it like: Scope (_SB.PC00.SPI0) { Device (GSPK) { Name (_HID, "TXNW2781") // _HID: Hardware ID Method (_CRS, 0, NotSerialized) { Name (RBUF, ResourceTemplate () { SpiSerialBusV2 (...) SpiSerialBusV2 (...) } } } } And in platform/x86/serial-multi-instantiate.c, those spi devices will be added into system as a single SPI device, so TAS2781 SPI driver will probe twice for every single SPI device. And driver will also parser mono DSP firmware binary and RCA binary for itself. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> Baojun Xu (3): ALSA: hda/tas2781: Add tas2781 hda driver based on SPI ALSA: hda/tas2781: Tas2781 hda driver for SPI ALSA: hda/tas2781: Firmware load for tas2781 hda driver based on SPI drivers/acpi/scan.c | 2 + .../platform/x86/serial-multi-instantiate.c | 13 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/patch_realtek.c | 13 + sound/pci/hda/tas2781-spi.h | 149 ++ sound/pci/hda/tas2781_hda_spi.c | 1240 +++++++++ sound/pci/hda/tas2781_spi_fwlib.c | 2252 +++++++++++++++++ 8 files changed, 3685 insertions(+) create mode 100644 sound/pci/hda/tas2781-spi.h create mode 100644 sound/pci/hda/tas2781_hda_spi.c create mode 100644 sound/pci/hda/tas2781_spi_fwlib.c -- 2.40.1

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[PATCH v20 00/41] Introduce QC USB SND audio offloading support
by Wesley Cheng 03 May '24

03 May '24

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 Requesting to see if we can get some Acked-By tags, and merge on usb-next. Changelog -------------------------------------------- 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 (39): usb: host: xhci: Export enable and disable interrupter APIs usb: host: xhci: Repurpose event handler for skipping interrupter events xhci: export XHCI IMOD setting helper for interrupters usb: host: xhci-sideband: Expose a sideband interrupter enable API 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: qcom: Populate SoC components string ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 611 ++++++ 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 | 37 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 50 +- drivers/usb/host/xhci-sideband.c | 444 ++++ drivers/usb/host/xhci.c | 48 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 70 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 182 ++ 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 | 408 ++++ sound/soc/qcom/sm8250.c | 15 +- sound/soc/qcom/usb_offload_utils.c | 51 + sound/soc/qcom/usb_offload_utils.h | 26 + sound/soc/soc-usb.c | 660 ++++++ 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 | 1914 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 6311 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/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

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usb:gadget:f_uac2: RFC: allowing multiple altsetttings for channel/samplesize combinations
by Pavel Hofman 02 May '24

02 May '24

Hi, I am considering implementation of multiple altsettings to f_uac2, so that multiple combinations of channels and samplesizes can be offered to the host. Configuration: -------------- * each altsetting for each direction should define * channel mask * samplesize * hs_bint bInterval * c_sync type (for capture only) Perhaps the easiest config would be allowing lists for the existing parameters (like the multiple samplerates were implemented). All the list params would have to have the same number of items - initial check. First values in the list would apply to altsetting 1, second to altsetting 2 etc. Or the altsetting could be some structured configfs param - please is there any recommended standard for structured configfs params? Should the config also adjust the list of allowed samplerates for each altsetting? Technically it makes sense as higher number of channels can decrease the max samplerate, e.g. for via a TDM interface. If so, it would need either the structured configuration or some "list of lists" format. Implementation: --------------- Parameters could be turned to arrays of fixed predefined sizes, like the p/s_srates. E.g. 5 max. altsettings in each direction would consume only 4 * (5-1) + 3* (5-1) = 28 extra ints (excluding the samplerates config). Currently all descriptor structs are statically pre-allocated as there are only two hard-coded altsettings. IMO the descriptors specific for each altsetting could be allocated dynamically in a loop over all none-zero alsettings. Please may I ask UAC2 gadget "stakeholders" for comments, suggestions, recommendations, so that my eventual initial version was in some generally acceptable direction? Thanks a lot, Pavel.

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[PATCH v5 00/18] ASoC: Cleanup MediaTek soundcard machine drivers
by AngeloGioacchino Del Regno 02 May '24

02 May '24

Changes in v5: - Cleaned up MT8186 soundcard migration commit which erroneously had leftovers from development - Changed the mtk_pcm_constraints_data structure to hold pointers to snd_pcm_hw_constraint_list, as to really reuse the const data - Tested again on all of the listed MTK platforms. Changes in v4: - Rebased over next-20240409 - Dropped the first 4 patches from v3 as were already picked by Mark - Fixed W=1 build issue Changes in v3: - Added audio-routing names in enum in all yaml files - Added mention of disallowing old and new properties together in commit message of bindings patches - Fixed validation errors with sound-card-common.yaml inclusion due to missing model property in examples - Removed `else` enforcing headset-codec/speaker-codecs requirement if xxx-dai-link not present to avoid future commit noise as the deprecated statement will disallow deprecated properties as required Changes in v2: - Bindings: Changed link-name/codec/clk-provider to remove `items` and leave just the enum - Moved .*-dai-link pattern additionalProperties after `type: object` - Added ref to sound-card-common.yaml - Fixed dai-link-xxx -> xxx-dai-link typo in example comment This series performs a cleanup of most of the MediaTek AFE drivers and soundcard machine drivers, reducing code duplication and setting a base to be consistent with their devicetree bindings, as many of those are using different properties and nodes for no good reason. Summarizing: - Commonizes functions and ops where possible - Adds a common probe mechanism, increasing maintainability of soundcard drivers for older MediaTek SoCs - Migrates all drivers to support the new bindings - Obviously retains compatibility with old device trees - Reduces machine-specific parameters hardcoding in drivers - Can now set machine-specific params in device tree - Uses the `audio-routing` and `dai-link` nodes like some other non-MediaTek SoC sound drivers - Imposes consistency between MediaTek ASoC machine soundcard drivers bindings - Reduces code size and greatly reduces the amount of code that will be required for newer drivers (retaining compatibility with the old bindings was costly in terms of code size, otherwise this series would've removed ~1000 more lines, or something along that line). This series was (manually) tested on MT8173, MT8192, MT8195 and MT8186 Chromebooks. AngeloGioacchino Del Regno (18): ASoC: mediatek: Add common machine soundcard driver probe mechanism ASoC: mediatek: common: Constify struct mtk_sof_priv ASoC: mediatek: mt8188: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8195: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8192: Migrate to mtk_soundcard_common_probe ASoC: mediatek: mt8186: Migrate to mtk_soundcard_common_probe ASoC: mediatek: Add common snd_soc_ops .startup() callback ASoC: mediatek: mt8195: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8192: Migrate to the common mtk_soundcard_startup ASoC: mediatek: mt8186-rt1019: Migrate to the common mtk_soundcard_startup ASoC: mediatek: Add common mtk_afe_component_probe callback ASoC: mediatek: Use common mtk_afe_pcm_platform with common probe cb ASoC: mediatek: mt8186: Unify mt8186-mt6366 machine drivers ASoC: dt-bindings: mt8195: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8192: Document audio-routing and dai-link subnode ASoC: dt-bindings: mt8186: Document audio-routing and dai-link subnode arm64: dts: mediatek: mt8195-cherry: Specify sound DAI links and routing arm64: dts: mediatek: mt8186-corsola: Specify sound DAI links and routing .../sound/mt8186-mt6366-da7219-max98357.yaml | 131 +- .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 120 +- .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 139 +- .../bindings/sound/mt8195-mt6359.yaml | 134 ++ .../boot/dts/mediatek/mt8186-corsola.dtsi | 42 +- .../boot/dts/mediatek/mt8195-cherry.dtsi | 45 + sound/soc/mediatek/Kconfig | 24 +- .../mediatek/common/mtk-afe-platform-driver.c | 18 + .../soc/mediatek/common/mtk-dsp-sof-common.c | 15 +- .../soc/mediatek/common/mtk-dsp-sof-common.h | 1 - sound/soc/mediatek/common/mtk-soc-card.h | 7 +- .../mediatek/common/mtk-soundcard-driver.c | 199 +++ .../mediatek/common/mtk-soundcard-driver.h | 42 + sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 14 +- sound/soc/mediatek/mt7986/mt7986-afe-pcm.c | 14 +- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 14 +- sound/soc/mediatek/mt8186/Makefile | 3 +- .../mt8186/mt8186-mt6366-da7219-max98357.c | 1189 ----------------- ...t6366-rt1019-rt5682s.c => mt8186-mt6366.c} | 578 ++++---- sound/soc/mediatek/mt8188/mt8188-afe-pcm.c | 21 +- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 203 +-- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 25 +- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 301 ++--- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 21 +- sound/soc/mediatek/mt8195/mt8195-mt6359.c | 487 +++---- 25 files changed, 1597 insertions(+), 2190 deletions(-) delete mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c rename sound/soc/mediatek/mt8186/{mt8186-mt6366-rt1019-rt5682s.c => mt8186-mt6366.c} (72%) -- 2.44.0

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[PATCH 0/5] ASoc: SOF: misc updates for 6.10
by Pierre-Louis Bossart 01 May '24

01 May '24

Fixes when fw_lib_prefix is not set, updated error messages, improved dmesg logs to SoundWire configurations not supported by ACPI tables/topology and support for IEC61937 passthrough. Peter Ujfalusi (2): ASoC: SOF: debug: Handle cases when fw_lib_prefix is not set, NULL ASoC: SOF: ipc4-topology: Advertise passthrough capable PCMs (using ChainDMA) Pierre-Louis Bossart (2): ASoC: SOF: Intel: hda: simplify and reduce indentation for hda_sdw_machine_select() ASoC: SOF: Intel: hda: list SoundWire peripherals on mismatch Yong Zhi (1): ASoC: SOF: topology: remove incorrect widget id in error message sound/soc/sof/debug.c | 23 ++++- sound/soc/sof/intel/hda.c | 154 +++++++++++++++++++--------------- sound/soc/sof/ipc4-topology.c | 50 +++++++++++ sound/soc/sof/topology.c | 7 +- 4 files changed, 160 insertions(+), 74 deletions(-) -- 2.40.1

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[PATCH 0/4] ASoC: meson: tdm fixes
by Jerome Brunet 01 May '24

01 May '24

This patchset fixes 2 problems on TDM which both find a solution by properly implementing the .trigger() callback for the TDM backend. ATM, enabling the TDM formatters is done by the .prepare() callback because handling the formatter is slow due to necessary calls to CCF. The first problem affects the TDMIN. Because .prepare() is called on DPCM backend first, the formatter are started before the FIFOs and this may cause a random channel shifts if the TDMIN use multiple lanes with more than 2 slots per lanes. Using trigger() allows to set the FE/BE order, solving the problem. There has already been an attempt to fix this 3y ago [1] and reverted [2] It triggered a 'sleep in irq' error on the period IRQ. The solution is to just use the bottom half of threaded IRQ. This is patch #1. Patch #2 and #3 remain mostly the same as 3y ago. For TDMOUT, the problem is on pause. ATM pause only stops the FIFO and the TDMOUT just starves. When it does, it will actually repeat the last sample continuously. Depending on the platform, if there is no high-pass filter on the analog path, this may translate to a constant position of the speaker membrane. There is no audible glitch but it may damage the speaker coil. Properly stopping the TDMOUT in pause solves the problem. There is behaviour change associated with that fix. Clocks used to be continuous on pause because of the problem above. They will now be gated on pause by default, as they should. The last change introduce the proper support for continuous clocks, if needed. [1]: https://lore.kernel.org/linux-amlogic/20211020114217.133153-1-jbrunet@bayli… [2]: https://lore.kernel.org/linux-amlogic/20220421155725.2589089-1-narmstrong@b… Jerome Brunet (4): ASoC: meson: axg-fifo: use threaded irq to check periods ASoC: meson: axg-card: make links nonatomic ASoC: meson: axg-tdm-interface: manage formatters in trigger ASoC: meson: axg-tdm: add continuous clock support sound/soc/meson/axg-card.c | 1 + sound/soc/meson/axg-fifo.c | 29 +++++++++++++-------- sound/soc/meson/axg-tdm-formatter.c | 40 +++++++++++++++++++++++++++++ sound/soc/meson/axg-tdm-interface.c | 38 +++++++++++++++++++-------- sound/soc/meson/axg-tdm.h | 5 ++++ 5 files changed, 93 insertions(+), 20 deletions(-) -- 2.43.0

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[PATCH 00/23] ASoC: Intel: updates for 6.10 - part6
by Pierre-Louis Bossart 01 May '24

01 May '24

Last batch of cleanups from Brent Lu, with Chromebooks now supported with fewer modular machine drivers. Bard Liao (1): ASoC: Intel: sof_rt5682: add missing MAX98357A config Brent Lu (22): ASoC: Intel: nau8825/rt5682: move speaker widget to common modules ASoC: Intel: sof_maxim_common: support MAX98390 on cml boards ASoC: Intel: sof_da7219: support MAX98390 ASoC: Intel: sof_da7219: add cml_da7219_def for cml boards ASoC: Intel: sof_da7219: support MAX98390 on cml boards ASoC: Intel: bxt_da7219_max98357a: remove MAX98390 support ASoC: Intel: sof_da7219: support MAX98357A ASoC: Intel: sof_da7219: support MAX98357A on cml boards ASoC: Intel: bxt_da7219_max98357a: remove cml support ASoC: Intel: sof_da7219: add glk_da7219_def for glk boards ASoC: Intel: sof_da7219: support MAX98357A on glk boards ASoC: Intel: bxt_da7219_max98357a: remove glk support ASoC: Intel: sof_rt5682: add glk_rt5682_def for glk boards ASoC: Intel: sof_rt5682: support MAX98357A on glk boards ASoC: Intel: glk_rt5682_max98357a: delete driver ASoC: Intel: sof-rt5682: add mclk_en to sof_rt5682_private ASoC: Intel: sof-rt5682: remove SOF_RT5682_MCLK_BYTCHT_EN ASoC: Intel: sof_rt5682: add icl_rt5682_def for icl boards ASoC: Intel: sof-rt5682: add driver_data to sof_rt5682 board ASoC: Intel: sof-rt5682: setup pll_id only when needed ASoC: Intel: sof-rt5682: get bclk frequency from topology ASoC: Intel: sof-rt5682: support bclk as PLL source on rt5682s sound/soc/intel/boards/Kconfig | 41 +- sound/soc/intel/boards/Makefile | 4 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 179 +---- sound/soc/intel/boards/glk_rt5682_max98357a.c | 693 ------------------ sound/soc/intel/boards/sof_board_helpers.h | 2 + sound/soc/intel/boards/sof_da7219.c | 110 ++- sound/soc/intel/boards/sof_maxim_common.c | 108 ++- sound/soc/intel/boards/sof_nau8825.c | 4 - sound/soc/intel/boards/sof_realtek_common.c | 93 ++- sound/soc/intel/boards/sof_rt5682.c | 233 +++--- .../intel/common/soc-acpi-intel-cml-match.c | 4 +- .../intel/common/soc-acpi-intel-glk-match.c | 4 +- .../intel/common/soc-acpi-intel-icl-match.c | 2 +- 13 files changed, 461 insertions(+), 1016 deletions(-) delete mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c -- 2.40.1

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