Alsa-devel April 2022

alsa-devel@alsa-project.org

156 participants
384 discussions

Regression playing 24bit/96kHz audio on USB audio interface between 5.10.94 and 5.15.21
by Ruud van Asseldonk 02 Jan '23

02 Jan '23

Hi, I have a program that configures a Behringer UMC404HD 192k USB audio interface for playback as follows (pseudocode): snd_pcm_hw_params_set_channels(pcm, hwp, 2); snd_pcm_hw_params_set_format(pcm, hwp, SND_PCM_FORMAT_S24_3LE); snd_pcm_hw_params_set_access(pcm, hwp, SND_PCM_ACCESS_MMAP_INTERLEAVED); snd_pcm_hw_params_set_rate(pcm, hwp, val=96000, dir=0); snd_pcm_hw_params_set_period_size_near(pcm, hwp, val=513, dir=0); // If I confirm at this point with snd_pcm_hw_params_get_period_size, // I get back 513. snd_pcm_hw_params_set_buffer_size_near(pcm, hwp, val=2052); snd_pcm_hw_params(pcm, hwp); // If I confirm with snd_pcm_hw_params_get_buffer_size, // I get back 2052. On Linux 5.10.94 (specifically Arch Linux ARM package linux-rpi-5.10.94-1), this succeeds, and I can play back audio just fine. On 5.15.21 (specifically linux-rpi-5.15.21-3), all the calls succeed apart from the last one, which returns EINVAL. (My actual program is written in Rust and uses a Rust wrapper library for libasound, https://docs.rs/alsa/0.6.0/alsa/, but the calls should be the ones listed.) I confirmed with snd_pcm_hw_params_test_format that the S24_3LE format is supported. On both kernels it is successful. I confirmed with snd_pcm_hw_params_test_rate that 96 kHz is supported. On both kernels it is successful. There are differences though. When I query these: snd_pcm_hw_params_get_period_size_min snd_pcm_hw_params_get_period_size_max snd_pcm_hw_params_get_buffer_size_min snd_pcm_hw_params_get_buffer_size_max Then on 5.10.94 (the good version), I get: period size min/max: 12/32768, buffer size min/max: 24/65536 But on 5.15.21 (the bad version), I get: period size min/max: 13/71332, buffer size min/max: 26/142663 Also, after I set the period size to 513, if I then query the buffer size min/max, on 5.10.94 I get: buffer size min/max: 1026/65536 But on 5.15.21 I get: buffer size min/max: 1027/142663 In lsmod, snd_usb_audio shows up, and from lsusb -v, the UMC404HD appears to be a standard USB audio interface. The lsusb output is identical on both kernels. Please let me know if there is anything I can do to help further diagnose the issue, or if I should report this elsewhere. Kind regards, Ruud van Asseldonk

5 10

[alsa-devel] [PATCH 0/6] ASoC: mediatek: mt8183: support DP audio
by Tzung-Bi Shih 24 Nov '22

24 Nov '22

(This series depends on https://lore.kernel.org/patchwork/patch/1126819/ which has not applied to the maintainer's tree yet.) This series makes mt8183-mt6358-ts3a227-max98357 support DP (display port) audio output. ASoC: hdmi-codec: add PCM trigger operator - add a PCM trigger callback if codec driver is interested in PCM events. drm: bridge: it6505: bridge to hdmi-codec - add audio feature support of it6505. - bridge to hdmi-codec so that ASoC can use the audio feature. - defer the "enabling" after receiving PCM START or RESUME event. ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec - bind to hdmi-codec in mt8183-mt6358-ts3a227-max98357. drm: bridge: it6505: report connector status ASoC: mediatek: mt8183: support HDMI jack reporting - use the mechanism from the following patch https://mailman.alsa-project.org/pipermail/alsa-devel/2019-September/155302… to report jack status (from DRM to ASoC). Allen Chen (1): drm: bridge: it6505: bridge to hdmi-codec Tzung-Bi Shih (5): ASoC: hdmi-codec: add PCM trigger operator ASoC: dt-bindings: mt8183: add a property "mediatek,hdmi-codec" ASoC: mediatek: mt8183: use hdmi-codec drm: bridge: it6505: report connector status ASoC: mediatek: mt8183: support HDMI jack reporting .../sound/mt8183-mt6358-ts3a227-max98357.txt | 2 + drivers/gpu/drm/bridge/ite-it6505.c | 191 +++++++++++++++++- include/sound/hdmi-codec.h | 13 ++ sound/soc/codecs/hdmi-codec.c | 31 +++ sound/soc/mediatek/Kconfig | 3 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 36 +++- 6 files changed, 266 insertions(+), 10 deletions(-) -- 2.23.0.866.gb869b98d4c-goog

3 9

[PATCH v6 00/10] Add support for CS35L41 in HDA systems
by Lucas Tanure 31 Oct '22

31 Oct '22

Add support for laptops that have CS35L41 connected to an HDA codec by I2S and direct I2C connection to the CPU. Laptops that use CS35L41 and are SPI will be added in the future, after the support for it is resolved at i2c-multi-instantiate driver. i2c-multi-instantiate thread: https://lkml.org/lkml/2021/12/10/557 Hardware: - Some laptops have two CS35L41 amplifiers, connected to Realtek ALC287 by an I2S bus and by and direct I2C to the CPU. - The ALC287 codec is connected to the CPU by an HDA bus. - The CS35L41 has a DSP which will require firmware to be loaded. Architecture: - Creation of a library of shared functions for CS35L41 ASoC and HDA - HDA codec driver (RealTek) and CS35L41 HDA driver are combined by using component binding, where it uses device names to find the components and bind to the master - The HDA CS35L41 driver applies pre-defined registers sequences for each action in playback for HDA Sound card Changes from V5: - Fix build issues reported by Intel Test Bot - Check devm_kasprintf return - Add ACPI dependency to HDA drivers V5: https://lkml.org/lkml/2021/12/16/430 Changes from V4: - Save index received from ACPI for latter use in DSP Support - Move CS35L41_DSP1_CCM_CORE_CTRL to regmap_write so doesn't affect DSP hibernation feature V4: https://lkml.org/lkml/2021/12/14/487 Changes from V3: - SPI bus driver starter added - Use separate modules approach instead of liking library - Add CSC3551 ACPI _HID for more I2C laptops - Removed regulators from HDA driver - Add note about Non-conforming _HID V3: https://lkml.org/lkml/2021/11/23/723 Changes from V2: - Not an RFC - Create a new HDA driver for CS35L41 instead of using the ASoC one - Use component binding and device names to find the CS35L41 driver - Create a shared library for ASoC and HDA CS35L41 drivers v2: https://lkml.org/lkml/2021/10/8/344 Lucas Tanure (9): ASoC: cs35l41: Convert tables to shared source code ASoC: cs35l41: Move cs35l41_otp_unpack to shared code ASoC: cs35l41: Move power initializations to reg_sequence ASoC: cs35l41: Create shared function for errata patches ASoC: cs35l41: Create shared function for setting channels ASoC: cs35l41: Create shared function for boost configuration hda: cs35l41: Add support for CS35L41 in HDA systems ACPI / scan: Create platform device for CLSA0100 and CSC3551 ACPI nodes ALSA: hda/realtek: Add support for Legion 7 16ACHg6 laptop Stefan Binding (1): ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops MAINTAINERS | 2 + drivers/acpi/scan.c | 3 + drivers/platform/x86/i2c-multi-instantiate.c | 11 + include/sound/cs35l41.h | 739 ++++++++++++++++++ sound/pci/hda/Kconfig | 29 + sound/pci/hda/Makefile | 10 + sound/pci/hda/cs35l41_hda.c | 527 +++++++++++++ sound/pci/hda/cs35l41_hda.h | 69 ++ sound/pci/hda/cs35l41_hda_i2c.c | 66 ++ sound/pci/hda/cs35l41_hda_spi.c | 63 ++ sound/pci/hda/hda_component.h | 20 + sound/pci/hda/patch_realtek.c | 146 ++++ sound/soc/codecs/Kconfig | 11 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/cs35l41-i2c.c | 1 - .../{cs35l41-tables.c => cs35l41-lib.c} | 355 ++++++++- sound/soc/codecs/cs35l41-spi.c | 1 - sound/soc/codecs/cs35l41.c | 360 +-------- sound/soc/codecs/cs35l41.h | 734 ----------------- 19 files changed, 2055 insertions(+), 1096 deletions(-) create mode 100644 sound/pci/hda/cs35l41_hda.c create mode 100644 sound/pci/hda/cs35l41_hda.h create mode 100644 sound/pci/hda/cs35l41_hda_i2c.c create mode 100644 sound/pci/hda/cs35l41_hda_spi.c create mode 100644 sound/pci/hda/hda_component.h rename sound/soc/codecs/{cs35l41-tables.c => cs35l41-lib.c} (71%) -- 2.34.1

11 28

[PATCH 1/2] ALSA: hda: Add Intel DG2 PCI ID and HDMI codec vid
by Kai Vehmanen 14 Oct '22

14 Oct '22

Add HD Audio PCI ID and HDMI codec vendor ID for Intel DG2. Reviewed-by: Uma Shankar <uma.shankar(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 12 +++++++++++- sound/pci/hda/patch_hdmi.c | 1 + 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 221afacbc7fd..b98d7975adea 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -335,7 +335,10 @@ enum { ((pci)->device == 0x0c0c) || \ ((pci)->device == 0x0d0c) || \ ((pci)->device == 0x160c) || \ - ((pci)->device == 0x490d)) + ((pci)->device == 0x490d) || \ + ((pci)->device == 0x4f90) || \ + ((pci)->device == 0x4f91) || \ + ((pci)->device == 0x4f92)) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) @@ -2477,6 +2480,13 @@ static const struct pci_device_id azx_ids[] = { /* DG1 */ { PCI_DEVICE(0x8086, 0x490d), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* DG2 */ + { PCI_DEVICE(0x8086, 0x4f90), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f91), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x4f92), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Alderlake-S */ { PCI_DEVICE(0x8086, 0x7ad0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 65d2c5539919..98633d2684de 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4382,6 +4382,7 @@ HDA_CODEC_ENTRY(0x80862814, "DG1 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862815, "Alderlake HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x80862816, "Rocketlake HDMI", patch_i915_tgl_hdmi), +HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_tgl_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), base-commit: d7aca8d8f548570f650d9b1dfedf6902f56f2bce -- 2.34.1

3 3

[PATCH] sound: soc/codecs: tlv320adc3xxx: needs GPIOLIB
by Randy Dunlap 31 Aug '22

31 Aug '22

Fix build errors when CONFIG_GPIOLIB is not enabled: ../sound/soc/codecs/tlv320adc3xxx.c: In function 'adc3xxx_i2c_probe': ../sound/soc/codecs/tlv320adc3xxx.c:1352:28: error: implicit declaration of function 'devm_gpiod_get'; did you mean 'devm_gpio_free'? [-Werror=implicit-function-declaration] 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); ../sound/soc/codecs/tlv320adc3xxx.c:1352:57: error: 'GPIOD_OUT_LOW' undeclared (first use in this function); did you mean 'GPIOF_INIT_LOW'? 1352 | adc3xxx->rst_pin = devm_gpiod_get(dev, "reset", GPIOD_OUT_LOW); CC lib/dynamic_debug.o ../sound/soc/codecs/tlv320adc3xxx.c:1400:9: error: implicit declaration of function 'gpiod_set_value_cansleep'; did you mean 'gpio_set_value_cansleep'? [-Werror=implicit-function-declaration] 1400 | gpiod_set_value_cansleep(adc3xxx->rst_pin, 1); Fixes: e9a3b57efd28 ("ASoC: codec: tlv320adc3xxx: New codec driver") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Ricard Wanderlof <ricardw(a)axis.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1567,6 +1567,7 @@ config SND_SOC_TFA989X config SND_SOC_TLV320ADC3XXX tristate "Texas Instruments TLV320ADC3001/3101 audio ADC" depends on I2C + depends on GPIOLIB help Enable support for Texas Instruments TLV320ADC3001 and TLV320ADC3101 ADCs.

1 1

Request for setup of new repositories
by Takashi Sakamoto 27 Aug '22

27 Aug '22

Hi Jaroslav, Iwai-san, Thanks for your maintenance for alsa-project organization in github.com. Currently I'd like to add new three repositories under the organization as a part of my work for ALSA firewire stack. I've been maintaining libhinawa since 2014 and recently realized that current design is not necessarily convenient since it includes two functions; operation to Linux FireWire cdev, and operation of ALSA HwDep cdev. Currently I'm working for new library to split the latter operation. Then I'd like you to setup below repositories: * 'libhitaki' * 'libhitaki-doc' * 'hitaki-rs' The library itself and its Rust binding are utilized by 'snd-firewire-ctl-services'[2], thus it's preferable to register them under 'GObject Introspection' Team. Thanks for your assist for my work. [1] https://github.com/alsa-project/libhinawa [2] https://github.com/alsa-project/snd-firewire-ctl-services Regards Takashi Sakamoto

2 20

[PATCH] Add quirk to enable Avid Mbox 3 support
by connerknoxpublic＠gmail.com 22 Aug '22

22 Aug '22

From: Conner Knox <connerknoxpublic(a)gmail.com> Hi all, I believe these are the right emails to send this patch to, but if I've missed something please let me know. This patch enables the Avid Mbox 3 audio interface to be used through ALSA. It locks the device at a 48khz sample rate. Then it mutes and raises the appropriate channels/inputs to allow for expected functionality. Thanks for your time. --- sound/usb/quirks-table.h | 76 ++++++++++ sound/usb/quirks.c | 311 +++++++++++++++++++++++++++++++++++++++ 2 files changed, 387 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 0ea39565e623..5f43863e5a5f 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2978,6 +2978,82 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +/* DIGIDESIGN MBOX 3 */ +{ + USB_DEVICE(0x0dba, 0x5000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 3", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 2, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0x00, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 3, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, + .iface = 3, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x81, + .attributes = 0x00, + .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, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 4, + .type = QUIRK_MIDI_FIXED_ENDPOINT, + .data = & (const struct snd_usb_midi_endpoint_info) { + .out_cables = 0x0001, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, { /* Tascam US122 MKII - playback-only support */ USB_DEVICE_VENDOR_SPEC(0x0644, 0x8021), diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..e7caa7eeb955 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1020,6 +1020,313 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } +static void mbox3_setup_48_24_magic(struct usb_device *dev) +{ + u8 com_buff[4]; + + /* Load 48000Hz rate into buffer */ + com_buff[0] = 0x80; + com_buff[1] = 0xbb; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + + /* Set 48000Hz sample rate */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x0001, &com_buff, 0x0004); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x01, 0x21, 0x0100, 0x8101, &com_buff, 0x0004); + + /* Set clock source to Internal (as opposed to S/PDIF) */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x01; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x8001, &com_buff, 1); + + /* Mute the hardware loopbacks to start the device in a known state. + * The Mbox 3 is little endian and + * the data written here seems a bit odd. + * 0x8000 (shown in big endian form) is muted + * and volume increases to 0xffff it seems. + * I've yet to observe 0xffff being sent. + * Instead, full volume seems to be 0x0000. + * Per my understanding of 16 bit integers, this is seems strange, but maybe I'm missing something + */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue input 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0110, 0x4001, &com_buff, 2); + /* Analogue input 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0111, 0x4001, &com_buff, 2); + /* Analogue input 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0114, 0x4001, &com_buff, 2); + /* Analogue input 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0115, 0x4001, &com_buff, 2); + /* Analogue input 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0118, 0x4001, &com_buff, 2); + /* Analogue input 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0119, 0x4001, &com_buff, 2); + /* Analogue input 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011c, 0x4001, &com_buff, 2); + /* Analogue input 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011d, 0x4001, &com_buff, 2); + + /* Set software sends to output */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue software return 1 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4001, &com_buff, 2); + /* Analogue software return 1 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4001, &com_buff, 2); + /* Analogue software return 2 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4001, &com_buff, 2); + /* Analogue software return 2 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4001, &com_buff, 2); + /* Analogue software return 3 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0108, 0x4001, &com_buff, 2); + /* Analogue software return 3 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0109, 0x4001, &com_buff, 2); + /* Analogue software return 4 left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010c, 0x4001, &com_buff, 2); + /* Analogue software return 4 right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010d, 0x4001, &com_buff, 2); + + /* Return to muting sends */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* Analogue fx return left channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0120, 0x4001, &com_buff, 2); + /* Analogue fx return right channel: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0121, 0x4001, &com_buff, 2); + + /* Analogue software input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0100, 0x4201, &com_buff, 2); + /* Analogue software input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0101, 0x4201, &com_buff, 2); + /* Analogue software input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4201, &com_buff, 2); + /* Analogue software input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4201, &com_buff, 2); + /* Analogue input 1 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0104, 0x4201, &com_buff, 2); + /* Analogue input 2 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0105, 0x4201, &com_buff, 2); + /* Analogue input 3 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4201, &com_buff, 2); + /* Analogue input 4 fx send: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4201, &com_buff, 2); + + /* Toggle allowing host control */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x02; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0000, 0x2001, &com_buff, 1); + + /* Do not dim fx returns */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0002, 0x2001, &com_buff, 1); + + /* Do not set fx returns to mono */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 3, 0x21, 0x0001, 0x2001, &com_buff, 1); + + /* Mute the S/PDIF hardware loopback + * same odd volume logic here as above + */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x80; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* S/PDIF hardware input 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0112, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0113, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0116, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0117, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011a, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011b, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011e, 0x4001, &com_buff, 2); + /* S/PDIF hardware input 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x011f, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0102, 0x4001, &com_buff, 2); + /* S/PDIF software return 1 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0103, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0106, 0x4001, &com_buff, 2); + /* S/PDIF software return 2 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0107, 0x4001, &com_buff, 2); + /* S/PDIF software return 3 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010a, 0x4001, &com_buff, 2); + /* S/PDIF software return 3 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010b, 0x4001, &com_buff, 2); + /* S/PDIF software return 4 left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010e, 0x4001, &com_buff, 2); + /* S/PDIF software return 4 right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x010f, 0x4001, &com_buff, 2); + /* S/PDIF fx returns left channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0122, 0x4001, &com_buff, 2); + /* S/PDIF fx returns right channel */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0123, 0x4001, &com_buff, 2); + + /* Set the dropdown "Effect" to the first option */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0200, 0x4301, &com_buff, 1); + + /* Set the effect duration to 0 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0400, 0x4301, &com_buff, 2); + + /* Set the effect volume and feedback to 0 */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + /* feedback: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0500, 0x4301, &com_buff, 1); + /* volume: */ + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 1, 0x21, 0x0300, 0x4301, &com_buff, 1); + + /* Set soft button hold duration */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x05; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0005, 0x2001, &com_buff, 1); + + /* Use dim LEDs for button of state */ + memset(com_buff, 0x00, 4); + com_buff[0] = 0x00; + com_buff[1] = 0x00; + com_buff[2] = 0x00; + com_buff[3] = 0x00; + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 3, 0x21, 0x0004, 0x2001, &com_buff, 1); + + return; +} + +#define MBOX3_DESCRIPTOR_SIZE 464 + +static int snd_usb_mbox3_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + int descriptor_size; + + 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); + return -ENODEV; + } + + dev_dbg(&dev->dev, "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); + + 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", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + mbox3_setup_48_24_magic(dev); + dev_info(&dev->dev, "Digidesign Mbox 3: 24bit 48kHz"); + + return 0; /* Successful boot */ +} #define MICROBOOK_BUF_SIZE 128 @@ -1324,6 +1631,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x0dba, 0x3000): /* Digidesign Mbox 2 */ return snd_usb_mbox2_boot_quirk(dev); + case USB_ID(0x0dba, 0x5000): + /* Digidesign Mbox 3 */ + return snd_usb_mbox3_boot_quirk(dev); + case USB_ID(0x1235, 0x0010): /* Focusrite Novation Saffire 6 USB */ case USB_ID(0x1235, 0x0018): /* Focusrite Novation Twitch */ -- 2.35.1

4 8

[PATCH][next] ASoC: q6asm: use kcalloc() instead of kzalloc()
by Gustavo A. R. Silva 28 Jul '22

28 Jul '22

Use 2-factor multiplication argument form kcalloc() instead of kzalloc(). Link: https://github.com/KSPP/linux/issues/162 Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 9251d8548965..195780f75d05 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -513,7 +513,7 @@ int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac, return 0; } - buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC); + buf = kcalloc(periods, sizeof(*buf), GFP_ATOMIC); if (!buf) { spin_unlock_irqrestore(&ac->lock, flags); return -ENOMEM; -- 2.27.0

3 2

[PATCH v3 0/2] Fix apq8016 compat string
by Bryan O'Donoghue 27 Jun '22

27 Jun '22

V3: - Marks qcom,lpass-cpu-apq8016 as deprecated instead of removing - Bjorn V2: - Adds Reviewed-by: - Srini - Adds Fixes - Srini V1: Reusing the apq8016 on msm8939 I found running checkpatch that the compat string for the LPASS was throwing a warning. This is easily fixed by alinging the YAML, DTS and driver to the documented compat string - compatible = "qcom,lpass-cpu-apq8016"; + compatible = "qcom,apq8016-lpass-cpu"; Bryan O'Donoghue (2): ASoC: qcom: lpass: Fix apq8016 compat string to match yaml arm64: dts: qcom: Fix apq8016 compat string to match yaml arch/arm64/boot/dts/qcom/msm8916.dtsi | 2 +- sound/soc/qcom/lpass-apq8016.c | 1 + sound/soc/qcom/lpass-cpu.c | 5 +++++ 3 files changed, 7 insertions(+), 1 deletion(-) -- 2.35.1

3 5

[PATCH 1/4] ASoC: fsl-asoc-card: add WM8904 support
by Alifer Moraes 21 Jun '22

21 Jun '22

From: Pierluigi Passaro <pierluigi.p(a)variscite.com> The codec WM8904 can use internal FLL as PLL source. Whenever the PLL source is not an external MCLK, this source must be setup during hw_params callback otherwise the BCLK could be wrongly evaluated. The SND_SOC_BIAS_PREPARE event is raised after the hw_params callback, so there is no need to set again PLL and SYSCLK and actually there's no need at all the set_bias_level function. Also, when esai is used, a dedicated snd_soc_dai_set_tdm_slot call is required. Signed-off-by: Pierluigi Passaro <pierluigi.p(a)variscite.com> Signed-off by: Alifer Moraes <alifer.m(a)variscite.com> --- sound/soc/fsl/fsl-asoc-card.c | 47 +++++++++++++++++++++++++++++++++++ 1 file changed, 47 insertions(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5ee945505281..817dbc1ec635 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -23,6 +23,7 @@ #include "imx-audmux.h" #include "../codecs/sgtl5000.h" +#include "../codecs/wm8904.h" #include "../codecs/wm8962.h" #include "../codecs/wm8960.h" #include "../codecs/wm8994.h" @@ -257,6 +258,38 @@ static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) } } + if (of_device_is_compatible(dev->of_node, "fsl,imx-audio-wm8904")) { + struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0); + unsigned int pll_out; + + ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0, 0, 2, + params_physical_width(params)); + if (ret) { + dev_err(dev, "failed to set TDM slot for cpu dai\n"); + return ret; + } + + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, + codec_priv->pll_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + } + return 0; } @@ -651,6 +684,19 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBP_CFP; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8904")) { + codec_dai_name = "wm8904-hifi"; + priv->card.set_bias_level = NULL; + priv->codec_priv.mclk_id = WM8904_CLK_FLL; + priv->codec_priv.fll_id = WM8904_CLK_FLL; + priv->codec_priv.pll_id = WM8904_FLL_MCLK; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + if (strstr(cpu_np->name, "esai")) { + priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; + priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; + priv->cpu_priv.sysclk_dir[RX] = SND_SOC_CLOCK_OUT; + } } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; priv->dai_fmt = SND_SOC_DAIFMT_AC97; @@ -900,6 +946,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-tlv320aic32x4", }, { .compatible = "fsl,imx-audio-tlv320aic31xx", }, { .compatible = "fsl,imx-audio-sgtl5000", }, + { .compatible = "fsl,imx-audio-wm8904", }, { .compatible = "fsl,imx-audio-wm8962", }, { .compatible = "fsl,imx-audio-wm8960", }, { .compatible = "fsl,imx-audio-mqs", }, -- 2.25.1

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Alsa-devel April 2022