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

[PATCH] ASoC: rt1017: Remove unused function
by Jiapeng Chong 28 Jul '23

28 Jul '23

The function are defined in the rt1017-sdca-sdw.h file, but not called elsewhere, so delete the unused functions. sound/soc/codecs/rt1017-sdca-sdw.h:183:33: warning: ‘rt1017_sdca_mbq_defaults’ defined but not used. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Closes: https://bugzilla.openanolis.cn/show_bug.cgi?id=5998 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/codecs/rt1017-sdca-sdw.h | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/codecs/rt1017-sdca-sdw.h b/sound/soc/codecs/rt1017-sdca-sdw.h index 5968b9a7d124..4932b5dbe3c0 100644 --- a/sound/soc/codecs/rt1017-sdca-sdw.h +++ b/sound/soc/codecs/rt1017-sdca-sdw.h @@ -180,9 +180,4 @@ static const struct reg_default rt1017_sdca_reg_defaults[] = { RT1017_SDCA_CTL_REQ_POWER_STATE, 0), 0x03 }, }; -static const struct reg_default rt1017_sdca_mbq_defaults[] = { - { SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1017_SDCA_ENT_FU, - RT1017_SDCA_CTL_FU_VOLUME, 0x01), 0x00 }, -}; - #endif /* __RT1017_SDW_H__ */ -- 2.20.1.7.g153144c

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[RFC v2 00/11] iio: add iio backend device type
by Olivier Moysan 27 Jul '23

27 Jul '23

This v2 is an addon to initial RFC: https://lore.kernel.org/lkml/20230623140944.2613002-1-olivier.moysan@foss.s… Despite the "IIO backend" naming has to be changed (as pointed out by Jonathan previously), it has been kept here, for time being. The appropriated naming still has to be discussed later on. In the previous RFC the "IIO backend" concept was proposed through a set of template APIs. This v2 implements a functionnal exemple based on STM32 DFSDM, to bring scaling support to this peripheral. Olivier Moysan (11): iio: introduce iio backend device of: property: add device link support for io-backends dt-bindings: iio: stm32-dfsdm-adc: add scaling support dt-bindings: iio: adc: add scaling support to sd modulator iio: adc: stm32-dfsdm: manage dfsdm as a channel provider iio: adc: stm32-dfsdm: adopt generic channel bindings iio: adc: stm32-dfsdm: add scaling support to dfsdm iio: adc: sd modulator: add scale and offset support ARM: dts: stm32: adopt new dfsdm bindings on stm32mp151 ARM: dts: stm32: add dfsdm pins muxing on stm32mp15 ARM: dts: stm32: add dfsdm iio support on stm32mp157c-ev .../iio/adc/sigma-delta-modulator.yaml | 9 +- .../bindings/iio/adc/st,stm32-dfsdm-adc.yaml | 189 ++++++------------ arch/arm/boot/dts/st/stm32mp15-pinctrl.dtsi | 39 ++++ arch/arm/boot/dts/st/stm32mp151.dtsi | 18 +- arch/arm/boot/dts/st/stm32mp157c-ev1.dts | 68 +++++++ drivers/iio/Makefile | 1 + drivers/iio/adc/sd_adc_modulator.c | 106 ++++++++-- drivers/iio/adc/stm32-dfsdm-adc.c | 187 +++++++++++------ drivers/iio/industrialio-backend.c | 107 ++++++++++ drivers/of/property.c | 2 + include/linux/iio/backend.h | 56 ++++++ 11 files changed, 561 insertions(+), 221 deletions(-) create mode 100644 drivers/iio/industrialio-backend.c create mode 100644 include/linux/iio/backend.h -- 2.25.1

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[GIT PULL] ASoC fixes for v6.5-rc3
by Mark Brown 27 Jul '23

27 Jul '23

The following changes since commit 6eaae198076080886b9e7d57f4ae06fa782f90ef: Linux 6.5-rc3 (2023-07-23 15:24:10 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.5-rc3 for you to fetch changes up to f85739c0b2b0d98a32f5ca4fcc5501d2b76df4f6: ASoC: atmel: Fix the 8K sample parameter in I2SC master (2023-07-25 12:10:00 +0100) ---------------------------------------------------------------- ASoC: Fixes for v6.5 A collection of device specific fixes, none particularly remarkable. There's a set of repetitive fixes for the RealTek drivers fixing an issue with suspend that was replicated in multiple drivers. ---------------------------------------------------------------- Dmytro Maluka (2): ASoC: da7219: Flush pending AAD IRQ when suspending ASoC: da7219: Check for failure reading AAD IRQ events Edgar (1): ASoc: codecs: ES8316: Fix DMIC config Edson Juliano Drosdeck (1): ASoC: nau8821: Add DMI quirk mechanism for active-high jack-detect Guiting Shen (1): ASoC: atmel: Fix the 8K sample parameter in I2SC master Mark Brown (3): ASoC: Merge up fixes from mainline ASoC: da7219: Patches related to a spurious AAD IRQ ASoC: wm8904: Fill the cache for WM8904_ADC_TEST_0 register Matus Gajdos (1): ASoC: fsl_spdif: Silence output on stop Shuming Fan (5): ASoC: rt5682-sdw: fix for JD event handling in ClockStop Mode0 ASoC: rt712-sdca: fix for JD event handling in ClockStop Mode0 ASoC: rt722-sdca: fix for JD event handling in ClockStop Mode0 ASoC: rt711: fix for JD event handling in ClockStop Mode0 ASoC: rt711-sdca: fix for JD event handling in ClockStop Mode0 sound/soc/atmel/atmel-i2s.c | 5 ++++- sound/soc/codecs/da7219-aad.c | 12 +++++++++--- sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/nau8821.c | 41 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5682-sdw.c | 9 ++++++++- sound/soc/codecs/rt711-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/rt711-sdw.c | 9 ++++++++- sound/soc/codecs/rt712-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/rt722-sdca-sdw.c | 10 +++++++++- sound/soc/codecs/wm8904.c | 3 +++ sound/soc/fsl/fsl_spdif.c | 2 ++ 11 files changed, 103 insertions(+), 10 deletions(-)

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[PATCH] ALSA: hda/cs35l56: Do some clean up on probe error
by Dan Carpenter 27 Jul '23

27 Jul '23

Smatch complains that this return should be a goto: sound/pci/hda/cs35l56_hda.c:910 cs35l56_hda_common_probe() warn: missing unwind goto? The goto error disables cansleep so that seems reasonable. Fixes: 73cfbfa9caea ("ALSA: hda/cs35l56: Add driver for Cirrus Logic CS35L56 amplifier") Signed-off-by: Dan Carpenter <dan.carpenter(a)linaro.org> --- sound/pci/hda/cs35l56_hda.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/cs35l56_hda.c b/sound/pci/hda/cs35l56_hda.c index 71e95e64f8a4..4c3279f61b94 100644 --- a/sound/pci/hda/cs35l56_hda.c +++ b/sound/pci/hda/cs35l56_hda.c @@ -907,7 +907,7 @@ int cs35l56_hda_common_probe(struct cs35l56_hda *cs35l56, int id) ret = cs35l56_set_patch(&cs35l56->base); if (ret) - return ret; + goto err; regcache_mark_dirty(cs35l56->base.regmap); regcache_sync(cs35l56->base.regmap); -- 2.39.2

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[PATCH] ALSA: hda/realtek: Support ASUS G713PV laptop
by Pavel Asyutchenko 27 Jul '23

27 Jul '23

This laptop has CS35L41 amp connected via I2C. With this patch speakers begin to work if the missing _DSD properties are added to ACPI tables. Signed-off-by: Pavel Asyutchenko <svenpavel(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7c0f474ea3b3..dff92679ae72 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9680,6 +9680,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JI", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JYR/JZR", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), SND_PCI_QUIRK(0x1043, 0x1ccd, "ASUS X555UB", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x1d1f, "ASUS ROG Strix G17 2023 (G713PV)", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402", ALC245_FIXUP_CS35L41_SPI_2), -- 2.39.2

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[PATCH v2 0/2] Intel: sof_rt5682: remove quirk flag
by Brent Lu 27 Jul '23

27 Jul '23

We add a helper function to detect amplifier number according to device instance in ACPI table so the SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag and a dmi quirk for 4-amplifier configuration could be safely removed. Also refactor the max_98390_hw_params() function to use an array to handle the TDM parameter. Amplifier number detection and TDM parameter are tested on two Chromebooks. One with 2 MAX98390 and one with 4 MAX98390 amplifier. V2 Changes: - fix some typos in cover letter - use for_each_acpi_dev_match() macro to simplify the codec - use 'unsign int' for codec number variable Brent Lu (2): ASoC: Intel: maxim-common: get codec number from ACPI ASoC: Intel: sof_rt5682: remove SOF_MAX98390_TWEETER_SPEAKER_PRESENT flag sound/soc/intel/boards/sof_maxim_common.c | 171 +++++++++++++--------- sound/soc/intel/boards/sof_maxim_common.h | 21 ++- sound/soc/intel/boards/sof_rt5682.c | 37 +---- 3 files changed, 112 insertions(+), 117 deletions(-) -- 2.34.1

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[PATCH 1/1] ASoC: fsl: fsl_qmc_audio: Fix snd_pcm_format_t values handling
by Herve Codina 26 Jul '23

26 Jul '23

Running sparse on fsl_qmc_audio (make C=1) raises the following warnings: fsl_qmc_audio.c:387:26: warning: restricted snd_pcm_format_t degrades to integer fsl_qmc_audio.c:389:59: warning: incorrect type in argument 1 (different base types) fsl_qmc_audio.c:389:59: expected restricted snd_pcm_format_t [usertype] format fsl_qmc_audio.c:389:59: got unsigned int [assigned] i fsl_qmc_audio.c:564:26: warning: restricted snd_pcm_format_t degrades to integer fsl_qmc_audio.c:569:50: warning: incorrect type in argument 1 (different base types) fsl_qmc_audio.c:569:50: expected restricted snd_pcm_format_t [usertype] format fsl_qmc_audio.c:569:50: got int [assigned] i fsl_qmc_audio.c:573:62: warning: incorrect type in argument 1 (different base types) fsl_qmc_audio.c:573:62: expected restricted snd_pcm_format_t [usertype] format fsl_qmc_audio.c:573:62: got int [assigned] i These warnings are due to snd_pcm_format_t values handling done in the driver. Some macros and functions exist to handle safely these values. Use dedicated macros and functions to remove these warnings. Fixes: 075c7125b11c ("ASoC: fsl: Add support for QMC audio") Signed-off-by: Herve Codina <herve.codina(a)bootlin.com> --- sound/soc/fsl/fsl_qmc_audio.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/fsl_qmc_audio.c b/sound/soc/fsl/fsl_qmc_audio.c index 7cbb8e4758cc..56d6b0b039a2 100644 --- a/sound/soc/fsl/fsl_qmc_audio.c +++ b/sound/soc/fsl/fsl_qmc_audio.c @@ -372,8 +372,8 @@ static int qmc_dai_hw_rule_format_by_channels(struct qmc_dai *qmc_dai, struct snd_mask *f_old = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); unsigned int channels = params_channels(params); unsigned int slot_width; + snd_pcm_format_t format; struct snd_mask f_new; - unsigned int i; if (!channels || channels > nb_ts) { dev_err(qmc_dai->dev, "channels %u not supported\n", @@ -384,10 +384,10 @@ static int qmc_dai_hw_rule_format_by_channels(struct qmc_dai *qmc_dai, slot_width = (nb_ts / channels) * 8; snd_mask_none(&f_new); - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - if (snd_mask_test(f_old, i)) { - if (snd_pcm_format_physical_width(i) <= slot_width) - snd_mask_set(&f_new, i); + pcm_for_each_format(format) { + if (snd_mask_test_format(f_old, format)) { + if (snd_pcm_format_physical_width(format) <= slot_width) + snd_mask_set_format(&f_new, format); } } @@ -551,26 +551,26 @@ static const struct snd_soc_dai_ops qmc_dai_ops = { static u64 qmc_audio_formats(u8 nb_ts) { - u64 formats; - unsigned int chan_width; unsigned int format_width; - int i; + unsigned int chan_width; + snd_pcm_format_t format; + u64 formats_mask; if (!nb_ts) return 0; - formats = 0; + formats_mask = 0; chan_width = nb_ts * 8; - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + pcm_for_each_format(format) { /* * Support format other than little-endian (ie big-endian or * without endianness such as 8bit formats) */ - if (snd_pcm_format_little_endian(i) == 1) + if (snd_pcm_format_little_endian(format) == 1) continue; /* Support physical width multiple of 8bit */ - format_width = snd_pcm_format_physical_width(i); + format_width = snd_pcm_format_physical_width(format); if (format_width == 0 || format_width % 8) continue; @@ -581,9 +581,9 @@ static u64 qmc_audio_formats(u8 nb_ts) if (format_width > chan_width || chan_width % format_width) continue; - formats |= (1ULL << i); + formats_mask |= pcm_format_to_bits(format); } - return formats; + return formats_mask; } static int qmc_audio_dai_parse(struct qmc_audio *qmc_audio, struct device_node *np, -- 2.41.0

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[PATCH v2] ASoC: rt1316: fix key tone missing
by shumingf＠realtek.com 26 Jul '23

26 Jul '23

From: Shuming Fan <shumingf(a)realtek.com> This patch adds a control that there are three options to control the digital volume output. The user could select "immediately" to make volume updates immediately and avoid key tone missing issues. In default, the driver selects that the volume update when a zero-crossing with a soft ramp. Signed-off-by: Shuming Fan <shumingf(a)realtek.com> Reported-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/codecs/rt1316-sdw.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/soc/codecs/rt1316-sdw.c b/sound/soc/codecs/rt1316-sdw.c index 721821d9e9af..10a53c8d4874 100644 --- a/sound/soc/codecs/rt1316-sdw.c +++ b/sound/soc/codecs/rt1316-sdw.c @@ -424,6 +424,15 @@ static SOC_ENUM_SINGLE_DECL(rt1316_rx_data_ch_enum, SDW_SDCA_CTL(FUNC_NUM_SMART_AMP, RT1316_SDCA_ENT_UDMPU21, RT1316_SDCA_CTL_UDMPU_CLUSTER, 0), 0, rt1316_rx_data_ch_select); +static const char * const rt1316_dac_output_vol_select[] = { + "immediately", + "zero crossing", + "zero crossing with soft ramp", +}; + +static SOC_ENUM_SINGLE_DECL(rt1316_dac_vol_ctl_enum, + 0xc010, 6, rt1316_dac_output_vol_select); + static const struct snd_kcontrol_new rt1316_snd_controls[] = { /* I2S Data Channel Selection */ @@ -442,6 +451,9 @@ static const struct snd_kcontrol_new rt1316_snd_controls[] = { /* IV mixer Control */ SOC_DOUBLE("Isense Mixer Switch", 0xc605, 2, 0, 1, 1), SOC_DOUBLE("Vsense Mixer Switch", 0xc605, 3, 1, 1, 1), + + /* DAC Output Volume Control */ + SOC_ENUM("DAC Output Vol Control", rt1316_dac_vol_ctl_enum), }; static const struct snd_kcontrol_new rt1316_sto_dac = -- 2.34.1

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[PATCH] ASoC: intel: avs: refactor strncpy usage in topology
by justinstitt＠google.com 26 Jul '23

26 Jul '23

`strncpy` is deprecated for use on NUL-terminated destination strings [1]. A suitable replacement is `strscpy` [2]. There are some hopes that someday the `strncpy` api could be ripped out due to the vast number of suitable replacements (strscpy, strscpy_pad, strtomem, strtomem_pad, strlcpy) [1]. [1]: www.kernel.org/doc/html/latest/process/deprecated.html#strncpy-on-nul-termi… [2]: manpages.debian.org/testing/linux-manual-4.8/strscpy.9.en.html --- Link: https://github.com/KSPP/linux/issues/90 Signed-off-by: Justin Stitt <justinstitt(a)google.com> --- sound/soc/intel/avs/topology.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/avs/topology.c b/sound/soc/intel/avs/topology.c index cdb4ec500261..45d0eb2a8e71 100644 --- a/sound/soc/intel/avs/topology.c +++ b/sound/soc/intel/avs/topology.c @@ -1388,12 +1388,12 @@ static int avs_route_load(struct snd_soc_component *comp, int index, port = __ffs(mach->mach_params.i2s_link_mask); snprintf(buf, len, route->source, port); - strncpy((char *)route->source, buf, len); + strscpy((char *)route->source, buf, len); snprintf(buf, len, route->sink, port); - strncpy((char *)route->sink, buf, len); + strscpy((char *)route->sink, buf, len); if (route->control) { snprintf(buf, len, route->control, port); - strncpy((char *)route->control, buf, len); + strscpy((char *)route->control, buf, len); } } --- base-commit: 0b4a9fdc9317440a71d4d4c264a5650bf4a90f3c change-id: 20230725-sound-soc-intel-avs-remove-deprecated-strncpy-2bc41a5a5f81 Best regards, -- Justin Stitt <justinstitt(a)google.com>

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[PATCH] ALSA: usb-audio: Update for native DSD support quirks
by Jussi Laako 26 Jul '23

26 Jul '23

Maintenance patch for native DSD support. Remove incorrect T+A device quirks. Move set of device quirks to vendor quirks. Add set of missing device and vendor quirks. Signed-off-by: Jussi Laako <jussi(a)sonarnerd.net> --- sound/usb/quirks.c | 34 ++++++++++++++++++++++++++++------ 1 file changed, 28 insertions(+), 6 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index f9e200309b9e..598659d761cc 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1876,8 +1876,10 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* XMOS based USB DACs */ switch (chip->usb_id) { - case USB_ID(0x1511, 0x0037): /* AURALiC VEGA */ - case USB_ID(0x21ed, 0xd75a): /* Accuphase DAC-60 option card */ + case USB_ID(0x139f, 0x5504): /* Nagra DAC */ + case USB_ID(0x20b1, 0x3089): /* Mola-Mola DAC */ + case USB_ID(0x2522, 0x0007): /* LH Labs Geek Out 1V5 */ + case USB_ID(0x2522, 0x0009): /* LH Labs Geek Pulse X Inifinity 2V0 */ case USB_ID(0x2522, 0x0012): /* LH Labs VI DAC Infinity */ case USB_ID(0x2772, 0x0230): /* Pro-Ject Pre Box S2 Digital */ if (fp->altsetting == 2) @@ -1887,14 +1889,18 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, case USB_ID(0x0d8c, 0x0316): /* Hegel HD12 DSD */ case USB_ID(0x10cb, 0x0103): /* The Bit Opus #3; with fp->dsd_raw */ case USB_ID(0x16d0, 0x06b2): /* NuPrime DAC-10 */ - case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ + case USB_ID(0x16d0, 0x06b4): /* NuPrime Audio HD-AVP/AVA */ case USB_ID(0x16d0, 0x0733): /* Furutech ADL Stratos */ + case USB_ID(0x16d0, 0x09d8): /* NuPrime IDA-8 */ case USB_ID(0x16d0, 0x09db): /* NuPrime Audio DAC-9 */ + case USB_ID(0x16d0, 0x09dd): /* Encore mDSD */ case USB_ID(0x1db5, 0x0003): /* Bryston BDA3 */ + case USB_ID(0x20a0, 0x4143): /* WaveIO USB Audio 2.0 */ case USB_ID(0x22e1, 0xca01): /* HDTA Serenade DSD */ case USB_ID(0x249c, 0x9326): /* M2Tech Young MkIII */ case USB_ID(0x2616, 0x0106): /* PS Audio NuWave DAC */ case USB_ID(0x2622, 0x0041): /* Audiolab M-DAC+ */ + case USB_ID(0x278b, 0x5100): /* Rotel RC-1590 */ case USB_ID(0x27f7, 0x3002): /* W4S DAC-2v2SE */ case USB_ID(0x29a2, 0x0086): /* Mutec MC3+ USB */ case USB_ID(0x6b42, 0x0042): /* MSB Technology */ @@ -1904,9 +1910,6 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, /* Amanero Combo384 USB based DACs with native DSD support */ case USB_ID(0x16d0, 0x071a): /* Amanero - Combo384 */ - case USB_ID(0x2ab6, 0x0004): /* T+A DAC8DSD-V2.0, MP1000E-V2.0, MP2000R-V2.0, MP2500R-V2.0, MP3100HV-V2.0 */ - case USB_ID(0x2ab6, 0x0005): /* T+A USB HD Audio 1 */ - case USB_ID(0x2ab6, 0x0006): /* T+A USB HD Audio 2 */ if (fp->altsetting == 2) { switch (le16_to_cpu(chip->dev->descriptor.bcdDevice)) { case 0x199: @@ -2051,6 +2054,9 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x0644, 0x805f, /* TEAC Model 12 */ QUIRK_FLAG_FORCE_IFACE_RESET), + DEVICE_FLG(0x0644, 0x806b, /* TEAC UD-701 */ + QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY | + QUIRK_FLAG_IFACE_DELAY), DEVICE_FLG(0x06f8, 0xb000, /* Hercules DJ Console (Windows Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ @@ -2089,6 +2095,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), DEVICE_FLG(0x154e, 0x3006, /* Marantz SA-14S1 */ QUIRK_FLAG_ITF_USB_DSD_DAC | QUIRK_FLAG_CTL_MSG_DELAY), + DEVICE_FLG(0x154e, 0x300b, /* Marantz SA-KI RUBY / SA-12 */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x154e, 0x500e, /* Denon DN-X1600 */ QUIRK_FLAG_IGNORE_CLOCK_SOURCE), DEVICE_FLG(0x1686, 0x00dd, /* Zoom R16/24 */ @@ -2133,6 +2141,10 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_SHARE_MEDIA_DEVICE | QUIRK_FLAG_ALIGN_TRANSFER), DEVICE_FLG(0x21b4, 0x0081, /* AudioQuest DragonFly */ QUIRK_FLAG_GET_SAMPLE_RATE), + DEVICE_FLG(0x21b4, 0x0230, /* Ayre QB-9 Twenty */ + QUIRK_FLAG_DSD_RAW), + DEVICE_FLG(0x21b4, 0x0232, /* Ayre QX-5 Twenty */ + QUIRK_FLAG_DSD_RAW), DEVICE_FLG(0x2522, 0x0007, /* LH Labs Geek Out HD Audio 1V5 */ QUIRK_FLAG_SET_IFACE_FIRST), DEVICE_FLG(0x2708, 0x0002, /* Audient iD14 */ @@ -2175,12 +2187,18 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_VALIDATE_RATES), VENDOR_FLG(0x1235, /* Focusrite Novation */ QUIRK_FLAG_VALIDATE_RATES), + VENDOR_FLG(0x1511, /* AURALiC */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x152a, /* Thesycon devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x18d1, /* iBasso devices */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x1de7, /* Phoenix Audio */ QUIRK_FLAG_GET_SAMPLE_RATE), VENDOR_FLG(0x20b1, /* XMOS based devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x21ed, /* Accuphase Laboratory */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x22d9, /* Oppo */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x23ba, /* Playback Design */ @@ -2196,10 +2214,14 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x2ab6, /* T+A devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x2d87, /* Cayin device */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3336, /* HEM devices */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3353, /* Khadas devices */ QUIRK_FLAG_DSD_RAW), + VENDOR_FLG(0x35f4, /* MSB Technology */ + QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0x3842, /* EVGA */ QUIRK_FLAG_DSD_RAW), VENDOR_FLG(0xc502, /* HiBy devices */ -- 2.34.1

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