December 2021 - Alsa-devel - mailman.alsa-project.org

[RFC patch 0/5] Support BCLK input clock in tlv320aic31xx
by Ariel D'Alessandro 03 Dec '21

03 Dec '21

The tlv320aic31xx codec allows using BCLK as the input clock for PLL, deriving all the frequencies through a set of divisors. In this case, codec sysclk is determined by the hwparams sample rate/format. So its frequency must be updated from the codec itself when these are changed. This patchset modifies the tlv320aic31xx driver to update its sysclk if BCLK is used as the input clock. This allows to be used by the generic fsl-asoc-card, without having to add a specific driver. Ariel D'Alessandro (5): ASoC: tlv320aic31xx: Fix typo in BCLK clock name ASoC: tlv320aic31xx: Add support for pll_r coefficient ASoC: tlv320aic31xx: Add divs for bclk as clk_in ASoC: tlv320aic31xx: Handle BCLK set as PLL input configuration ASoC: fsl-asoc-card: Support fsl,imx-audio-tlv320aic31xx codec sound/soc/codecs/tlv320aic31xx.c | 105 ++++++++++++++++++++----------- sound/soc/codecs/tlv320aic31xx.h | 2 +- sound/soc/fsl/fsl-asoc-card.c | 12 ++++ 3 files changed, 83 insertions(+), 36 deletions(-) -- 2.30.2

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[PATCH v7 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 03 Dec '21

03 Dec '21

From: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add regmap config support for codec dma driver ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio ASoC: qcom: SC7280: Update config for building codec dma drivers .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 70 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 +++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 441 +++++++++++++++ sound/soc/qcom/lpass.h | 162 ++++++ 9 files changed, 1864 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH v7 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 03 Dec '21

03 Dec '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Changes Since V6: -- Remove redundant headers. -- Move max ports macro to lpass.h header. -- Arrange structure alignment. -- Fix indentation errors. -- Update module license. Changes Since V5: -- Add required properties to dt-bindings Changes Since V4: -- Add COMPILE_TEST flag in sc7280 configuration. -- Remove redundant startup and shutdown callbacks of snd_soc_ops. -- Fix typo errors. Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 171 ++++++++++++ sound/soc/qcom/Kconfig | 14 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass.h | 1 + sound/soc/qcom/sc7280.c | 290 +++++++++++++++++++++ 5 files changed, 478 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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Re: [linux-next:master 3956/5128] sound/soc/sof/intel/hda-codec.c:132:35: error: use of undeclared identifier 'CODEC_PROBE_RETRIES'
by Pierre-Louis Bossart 03 Dec '21

03 Dec '21

[adding alsa-devel] On 12/3/21 5:25 AM, kernel test robot wrote: > tree: https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git master > head: 7afeac307a9561e3a93682c1e7eb22f918aa1187 > commit: 046aede2f847676f93a2ea4f48b77909c51dba40 [3956/5128] ASoC: SOF: Intel: Retry codec probing if it fails > config: i386-randconfig-r033-20211202 (https://download.01.org/0day-ci/archive/20211203/202112031943.Twg19fWT-lkp@…) > compiler: clang version 14.0.0 (https://github.com/llvm/llvm-project 4b553297ef3ee4dc2119d5429adf3072e90fac38) > reproduce (this is a W=1 build): > wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross > chmod +x ~/bin/make.cross > # https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git/commit/… > git remote add linux-next https://git.kernel.org/pub/scm/linux/kernel/git/next/linux-next.git > git fetch --no-tags linux-next master > git checkout 046aede2f847676f93a2ea4f48b77909c51dba40 > # save the config file to linux build tree > mkdir build_dir > COMPILER_INSTALL_PATH=$HOME/0day COMPILER=clang make.cross W=1 O=build_dir ARCH=i386 SHELL=/bin/bash > > If you fix the issue, kindly add following tag as appropriate > Reported-by: kernel test robot <lkp(a)intel.com> > > All errors (new ones prefixed by >>): > >>> sound/soc/sof/intel/hda-codec.c:132:35: error: use of undeclared identifier 'CODEC_PROBE_RETRIES' > } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); > ^ > 1 error generated. > > > vim +/CODEC_PROBE_RETRIES +132 sound/soc/sof/intel/hda-codec.c > > 110 > 111 /* probe individual codec */ > 112 static int hda_codec_probe(struct snd_sof_dev *sdev, int address, > 113 bool hda_codec_use_common_hdmi) > 114 { > 115 #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) > 116 struct hdac_hda_priv *hda_priv; > 117 struct hda_codec *codec; > 118 int type = HDA_DEV_LEGACY; > 119 #endif > 120 struct hda_bus *hbus = sof_to_hbus(sdev); > 121 struct hdac_device *hdev; > 122 u32 hda_cmd = (address << 28) | (AC_NODE_ROOT << 20) | > 123 (AC_VERB_PARAMETERS << 8) | AC_PAR_VENDOR_ID; > 124 u32 resp = -1; > 125 int ret, retry = 0; > 126 > 127 do { > 128 mutex_lock(&hbus->core.cmd_mutex); > 129 snd_hdac_bus_send_cmd(&hbus->core, hda_cmd); > 130 snd_hdac_bus_get_response(&hbus->core, address, &resp); > 131 mutex_unlock(&hbus->core.cmd_mutex); > > 132 } while (resp == -1 && retry++ < CODEC_PROBE_RETRIES); Indeed, something's not right here. CODEC_PROBE_RETRIES is defined conditionally #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_AUDIO_CODEC) #define IDISP_VID_INTEL 0x80860000 #define CODEC_PROBE_RETRIES 3 but it's used unconditionally. We could define this constant unconditionally as a quick fix, but this compilation problem might expose a larger problem. Kai, I wonder if this is code from lines 120 to 139 that we didn't update when we moved to support HDMI with the generic HDaudio parts? I don't see why we could even try to send a command on the bus is there's no audio codec support? hda_codec_use_common_hdmi should be the default assumption now, I don't think we support the old solution, do we? > 133 > 134 if (resp == -1) > 135 return -EIO; > 136 dev_dbg(sdev->dev, "HDA codec #%d probed OK: response: %x\n", > 137 address, resp); > 138 > > --- > 0-DAY CI Kernel Test Service, Intel Corporation > https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org >

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[PATCH v2] ALSA: hda/realtek: Fix quirk for TongFang PHxTxX1
by Werner Sembach 03 Dec '21

03 Dec '21

This fixes the SND_PCI_QUIRK(...) of the TongFang PHxTxX1 barebone. This fixes the issue of sound not working after s3 suspend. When waking up from s3 suspend the Coef 0x10 is set to 0x0220 instead of 0x0020. Setting the value manually makes the sound work again. This patch does this automatically. While being on it, I also fixed the comment formatting of the quirk and shortened variable and function names. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Fixes: dd6dd6e3c791 ("ALSA: hda/realtek: Add quirk for TongFang PHxTxX1") Cc: <stable(a)vger.kernel.org> --- sound/pci/hda/patch_realtek.c | 40 +++++++++++++++++++---------------- 1 file changed, 22 insertions(+), 18 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9ce7457533c96..d361a1260d5a1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6503,22 +6503,26 @@ static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec, /* for alc285_fixup_ideapad_s740_coef() */ #include "ideapad_s740_helper.c" -static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *codec, - const struct hda_fixup *fix, - int action) +static const struct coef_fw alc256_fixup_set_coef_defaults_coefs[] = { + WRITE_COEF(0x10, 0x0020), WRITE_COEF(0x24, 0x0000), + WRITE_COEF(0x26, 0x0000), WRITE_COEF(0x29, 0x3000), + WRITE_COEF(0x37, 0xfe05), WRITE_COEF(0x45, 0x5089), + {} +}; + +static void alc256_fixup_set_coef_defaults(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) { /* - * A certain other OS sets these coeffs to different values. On at least one TongFang - * barebone these settings might survive even a cold reboot. So to restore a clean slate the - * values are explicitly reset to default here. Without this, the external microphone is - * always in a plugged-in state, while the internal microphone is always in an unplugged - * state, breaking the ability to use the internal microphone. - */ - alc_write_coef_idx(codec, 0x24, 0x0000); - alc_write_coef_idx(codec, 0x26, 0x0000); - alc_write_coef_idx(codec, 0x29, 0x3000); - alc_write_coef_idx(codec, 0x37, 0xfe05); - alc_write_coef_idx(codec, 0x45, 0x5089); + * A certain other OS sets these coeffs to different values. On at least + * one TongFang barebone these settings might survive even a cold + * reboot. So to restore a clean slate the values are explicitly reset + * to default here. Without this, the external microphone is always in a + * plugged-in state, while the internal microphone is always in an + * unplugged state, breaking the ability to use the internal microphone. + */ + alc_process_coef_fw(codec, alc256_fixup_set_coef_defaults_coefs); } static const struct coef_fw alc233_fixup_no_audio_jack_coefs[] = { @@ -6759,7 +6763,7 @@ enum { ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE, ALC287_FIXUP_YOGA7_14ITL_SPEAKERS, ALC287_FIXUP_13S_GEN2_SPEAKERS, - ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS, + ALC256_FIXUP_SET_COEF_DEFAULTS, ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE, ALC233_FIXUP_NO_AUDIO_JACK, }; @@ -8465,9 +8469,9 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE, }, - [ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS] = { + [ALC256_FIXUP_SET_COEF_DEFAULTS] = { .type = HDA_FIXUP_FUNC, - .v.func = alc256_fixup_tongfang_reset_persistent_settings, + .v.func = alc256_fixup_set_coef_defaults, }, [ALC245_FIXUP_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, @@ -8929,7 +8933,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), - SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS), + SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), -- 2.25.1

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[PATCH v6 0/2] Machine driver to support LPASS SC7280 sound card registration
by Srinivasa Rao Mandadapu 03 Dec '21

03 Dec '21

This patch set is to add support for SC7280 sound card registration and to add dt-bindings documentation file. Srinivasa Rao Mandadapu (2): ASoC: google: dt-bindings: Add sc7280-herobrine machine bindings ASoC: qcom: SC7280: Add machine driver Changes Since V5: -- Add required properties to dt-bindings Changes Since V4: -- Add COMPILE_TEST flag in sc7280 configuration. -- Remove redundant startup and shutdown callbacks of snd_soc_ops. -- Fix typo errors. Changes Since V3: -- Change audio jack playpause key value. Changes Since V2: -- updated required field in bindings -- updated Return type check with proper enum in sc7280.c -- Updated Header name and Typos in sc7280.c Changes Since V1: -- Indentation changes and typo. .../bindings/sound/google,sc7280-herobrine.yaml | 171 ++++++++++++ sound/soc/qcom/Kconfig | 12 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/sc7280.c | 299 +++++++++++++++++++++ 4 files changed, 484 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,sc7280-herobrine.yaml create mode 100644 sound/soc/qcom/sc7280.c -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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Practical questions about mainlining a downstream codec driver
by Vincent Knecht 03 Dec '21

03 Dec '21

Hi ! I'm looking at submitting a patch to add support for (i2c) ak4375 codec (a headphones amplifier). It works in close-to-mainline kernel forks used in postmarketOS for msm8916-alcatel-idol347 and msm8939-alcatel-idol3 phones (the latter is not in mainline yet). Currently the driver is mostly a cleanup and update of the downstream one to use modern framework (eg. regmap, snd_soc_component_* functions, etc.) The question is : how much should it be trimmed down ? For example : IC supports PLL and MCLK modes, and both testing devices use PLL mode. - should the driver only support the mode that I could test ? - should the UCM control(s) related to untested mode/features also be trimmed down ? or can they be kept, with an error/warning that selected value is unsupported ? Another question : While both devices are quite close, downstream drivers have little differences. For example, the idol347 one has an additional initialization register sequence. What are the options ? - allow setting this optional sequence from devicetree ? with register/values in driver and a boolean devicetree property to enable it ? or with an optional devicetree array property - put it in driver, and associate it with a specific additional compatible ? - ... ? Bonus question : Sadly, there's no detailed datasheet for ak4375 at https://www.akm.com/eu/en/products/audio/audio-dac/ and only a brief one is floating on the Internets, eg. https://datasheetspdf.com/pdf-file/1400317/AKM/AK4375A/1 I sent multiple inquiries through akm.com, even mailed directly to persons with akm.com addresses I could find in alsa mailing list so as to obtain a complete datasheet, with no success. Can anyone help with this ? Work on this driver is there : https://github.com/msm8916-mainline/linux/pull/203 Thank you

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init procedure is executed during a store and restore process when the final snd_ctl_elem is not readable
by GitHub issues - edited 03 Dec '21

03 Dec '21

alsa-project/alsa-utils issue #122 was edited from yixingwang0517: When i execute the following commands sequencitally, alsactl store -f 1.txt alsactl restore -f 1.txt my playing volume was initialized. After a debugging process, I found the problem: During the "store" process, a control element is saved to the configuration file only when this element is readable.While during the "restore" process, after all the control element from the configuration file is processed, an "additional control element check" which causes a "init procedure" if the result is true is processed. ![image](https://user-images.githubusercontent.com/19835921/144555692-6b3cee21-73fc-4ade-a1f4-cc8f99679a9d.png) Thus, if total control element count is 5 with the 5th element is not readable, then there will be only 4 control element stored in the configuration file, ranges from "control.1" to "control.4". Then the "additional control checking" process produces "true" and the variable "err" is evaluated as "-EAGAIN". Issue URL : https://github.com/alsa-project/alsa-utils/issues/122 Repository URL: https://github.com/alsa-project/alsa-utils

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init procedure is executed during a store and restore process when the final snd_ctl_elem is not readable
by GitHub issues - opened 03 Dec '21

03 Dec '21

alsa-project/alsa-utils issue #122 was opened from yixingwang0517: When i execute the following commands sequencitally, alsa store -f 1.txt alsa restore -f 1.txt my playing volume was initialized. After a debugging process, I found the problem: During the "store" process, a control element is saved to the configuration file only when this element is readable.While during the "restore" process, after all the control element from the configuration file is processed, an "additional control element check" which causes a "init procedure" if the result is true is processed. ![image](https://user-images.githubusercontent.com/19835921/144555692-6b3cee21-73fc-4ade-a1f4-cc8f99679a9d.png) Thus, if total control element count is 5 with the 5th element is not readable, then there will be only 4 control element stored in the configuration file, ranges from "control.1" to "control.4". Then the "additional control checking" process produces "true" and the variable "err" is evaluated as "-EAGAIN". Issue URL : https://github.com/alsa-project/alsa-utils/issues/122 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH] Sound: Add support for the Pioneer DJM 750MK2 Mixer/Soundcard
by William Overton 02 Dec '21

02 Dec '21

The kernel already has support for very similar Pioneer djm products and this work is based on that. Added device to quirks-table.h and added control info to mixer_quirks.c. Tested on my hardware and all working. Signed-off-by: William Overton <willovertonuk(a)gmail.com> --- sound/usb/mixer_quirks.c | 34 +++++++++++++++++++++++ sound/usb/quirks-table.h | 58 ++++++++++++++++++++++++++++++++++++++++ 2 files changed, 92 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 46082dc57be0..d489c1de3bae 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -2795,6 +2795,7 @@ static int snd_bbfpro_controls_create(struct usb_mixer_interface *mixer) #define SND_DJM_750_IDX 0x1 #define SND_DJM_850_IDX 0x2 #define SND_DJM_900NXS2_IDX 0x3 +#define SND_DJM_750MK2_IDX 0x4 #define SND_DJM_CTL(_name, suffix, _default_value, _windex) { \ @@ -2984,10 +2985,40 @@ static const struct snd_djm_ctl snd_djm_ctls_900nxs2[] = { SND_DJM_CTL("Ch5 Input", 900nxs2_cap5, 3, SND_DJM_WINDEX_CAP) }; +// DJM-750MK2 +static const u16 snd_djm_opts_750mk2_cap1[] = { + 0x0100, 0x0102, 0x0103, 0x0106, 0x0107, 0x0108, 0x0109, 0x010a }; +static const u16 snd_djm_opts_750mk2_cap2[] = { + 0x0200, 0x0202, 0x0203, 0x0206, 0x0207, 0x0208, 0x0209, 0x020a }; +static const u16 snd_djm_opts_750mk2_cap3[] = { + 0x0300, 0x0302, 0x0303, 0x0306, 0x0307, 0x0308, 0x0309, 0x030a }; +static const u16 snd_djm_opts_750mk2_cap4[] = { + 0x0400, 0x0402, 0x0403, 0x0406, 0x0407, 0x0408, 0x0409, 0x040a }; +static const u16 snd_djm_opts_750mk2_cap5[] = { + 0x0507, 0x0508, 0x0509, 0x050a, 0x0511, 0x0512, 0x0513, 0x0514 }; + +static const u16 snd_djm_opts_750mk2_pb1[] = { 0x0100, 0x0101, 0x0104 }; +static const u16 snd_djm_opts_750mk2_pb2[] = { 0x0200, 0x0201, 0x0204 }; +static const u16 snd_djm_opts_750mk2_pb3[] = { 0x0300, 0x0301, 0x0304 }; + + +static const struct snd_djm_ctl snd_djm_ctls_750mk2[] = { + SND_DJM_CTL("Capture Level", cap_level, 0, SND_DJM_WINDEX_CAPLVL), + SND_DJM_CTL("Ch1 Input", 750mk2_cap1, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch2 Input", 750mk2_cap2, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch3 Input", 750mk2_cap3, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch4 Input", 750mk2_cap4, 2, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch5 Input", 750mk2_cap5, 3, SND_DJM_WINDEX_CAP), + SND_DJM_CTL("Ch1 Output", 750mk2_pb1, 0, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch2 Output", 750mk2_pb2, 1, SND_DJM_WINDEX_PB), + SND_DJM_CTL("Ch3 Output", 750mk2_pb3, 2, SND_DJM_WINDEX_PB) +}; + static const struct snd_djm_device snd_djm_devices[] = { SND_DJM_DEVICE(250mk2), SND_DJM_DEVICE(750), + SND_DJM_DEVICE(750mk2), SND_DJM_DEVICE(850), SND_DJM_DEVICE(900nxs2) }; @@ -3235,6 +3266,9 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x08e4, 0x017f): /* Pioneer DJ DJM-750 */ err = snd_djm_controls_create(mixer, SND_DJM_750_IDX); break; + case USB_ID(0x2b73, 0x001b): /* Pioneer DJ DJM-750MK2 */ + err = snd_djm_controls_create(mixer, SND_DJM_750MK2_IDX); + break; case USB_ID(0x08e4, 0x0163): /* Pioneer DJ DJM-850 */ err = snd_djm_controls_create(mixer, SND_DJM_850_IDX); break; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index e03043f7dad3..bc0116273e94 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3850,6 +3850,64 @@ YAMAHA_DEVICE(0x7010, "UB99"), } } }, +{ + /* + * Pioneer DJ DJM-750MK2 + * 10 channels playback & 12 channels capture @ 48kHz S24LE + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x001b), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 10, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .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 = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 12, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_idx = 1, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 48000 } + } + }, + { + .ifnum = -1 + } + } + } +}, { /* * Pioneer DJ DJM-850 -- 2.30.2

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