December 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH 00/14] ASoC: Intel/SOF: extend run-time driver selection to ACPI devices
by Pierre-Louis Bossart 26 Apr '21

26 Apr '21

The module snd-intel-dspcfg, suggested by Jaroslav last year, currently provide the means to select a PCI driver at run-time, based on quirks, recommendations or user selection via a kernel parameter. This capability removed a lot of confusions in distributions and removed the need for recompilations to select legacy HDaudio, SST or SOF drivers. This patchset extends the concept to ACPI devices. This was driven by the desire to at some point deprecate the Atom/SST driver for Baytrail and Cherrytrail, which is no longer maintained by Intel. By having the SOF driver enabled by distributions for Baytrail/Cherrytrail, we can enable more end-user tests and make the transition easier for distributions (likely in 2021 at this point). This patchset provides the same solution for Broadwell, mainly to have a single build for all Intel platforms. SOF on Broadwell remains an option not recommended for distributions, as long as the 'catpt' driver is maintained there is no burning desire to make SOF the default on the three Broadwell-based platforms with the DSP enabled. Pierre-Louis Bossart (14): ASoC: Intel: broadwell: add missing pm_ops ASoC: Intel: bdw-rt5677: add missing pm_ops ALSA: hda: intel-dsp-config: add helper for ACPI DSP driver selection ASoC: soc-acpi: add helper to identify parent driver. ASoC: Intel: boards: byt/cht: set card and driver name at run time ASoC: Intel: byt/cht: set pm ops dynamically ASoC: SOF: acpi: add dynamic selection of DSP driver ASoC: Intel: Atom: add dynamic selection of DSP driver ASoC: SOF: Intel: allow for coexistence between SOF and Atom/SST drivers ALSA: hda: intel-dsp-config: add Broadwell ACPI DSP driver selection ASoC: Intel: broadwell: set card and driver name dynamically ASoC: Intel: catpt: add dynamic selection of DSP driver ASoC: SOF: Intel: allow for coexistence between SOF and catpt drivers ALSA: hda: intel-dsp-config: ignore dsp_driver parameter for PCI legacy devices include/sound/intel-dsp-config.h | 7 ++ include/sound/soc-acpi.h | 6 + sound/hda/intel-dsp-config.c | 111 +++++++++++++++++++ sound/soc/intel/Kconfig | 2 + sound/soc/intel/atom/sst/sst_acpi.c | 8 ++ sound/soc/intel/boards/bdw-rt5650.c | 17 ++- sound/soc/intel/boards/bdw-rt5677.c | 18 ++- sound/soc/intel/boards/broadwell.c | 20 ++-- sound/soc/intel/boards/bytcht_cx2072x.c | 27 +++-- sound/soc/intel/boards/bytcht_da7213.c | 27 +++-- sound/soc/intel/boards/bytcht_es8316.c | 29 +++-- sound/soc/intel/boards/bytcr_rt5640.c | 30 +++-- sound/soc/intel/boards/bytcr_rt5651.c | 27 +++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 29 +++-- sound/soc/intel/boards/cht_bsw_nau8824.c | 29 +++-- sound/soc/intel/boards/cht_bsw_rt5645.c | 38 ++++--- sound/soc/intel/boards/cht_bsw_rt5672.c | 29 +++-- sound/soc/intel/catpt/device.c | 12 ++ sound/soc/sof/intel/Kconfig | 33 +++--- sound/soc/sof/sof-acpi-dev.c | 14 ++- 20 files changed, 392 insertions(+), 121 deletions(-) -- 2.25.1

6 54

[PATCH] ASoC: fsl: imx-pcm-dma: Don't request dma channel in probe
by Shengjiu Wang 22 Apr '21

22 Apr '21

There are two requirements that we need to move the request of dma channel from probe to open. - When dma device binds with power-domains, the power will be enabled when we request dma channel. If the request of dma channel happen on probe, then the power-domains will be always enabled after kernel boot up, which is not good for power saving, so we need to move the request of dma channel to .open(); - With FE-BE case, if the dma channel is requested in probe, then there will be below issue, which is caused by that the dma channel will be requested duplicately [ 638.906268] sysfs: cannot create duplicate filename '/devices/soc0/soc/2000000.bus/2000000.spba-bus/2024000.esai/dma:tx' [ 638.919061] CPU: 1 PID: 673 Comm: aplay Not tainted 5.7.0-rc1-12956-gfc64b2585593 #287 [ 638.927113] Hardware name: Freescale i.MX6 Quad/DualLite (Device Tree) [ 638.933690] [<c0110dd8>] (unwind_backtrace) from [<c010b8ec>] (show_stack+0x10/0x14) [ 638.941464] [<c010b8ec>] (show_stack) from [<c0557fc0>] (dump_stack+0xe4/0x118) [ 638.948808] [<c0557fc0>] (dump_stack) from [<c032aeb4>] (sysfs_warn_dup+0x50/0x64) [ 638.956406] [<c032aeb4>] (sysfs_warn_dup) from [<c032b1a8>] (sysfs_do_create_link_sd+0xc8/0xd4) [ 638.965134] [<c032b1a8>] (sysfs_do_create_link_sd) from [<c05dc668>] (dma_request_chan+0xb0/0x210) [ 638.974120] [<c05dc668>] (dma_request_chan) from [<c05dc7d0>] (dma_request_slave_channel+0x8/0x14) [ 638.983111] [<c05dc7d0>] (dma_request_slave_channel) from [<c09d5548>] (fsl_asrc_dma_hw_params+0x1e0/0x438) [ 638.992881] [<c09d5548>] (fsl_asrc_dma_hw_params) from [<c09c1654>] (soc_pcm_hw_params+0x4a0/0x6a8) [ 639.001952] [<c09c1654>] (soc_pcm_hw_params) from [<c09c39d4>] (dpcm_fe_dai_hw_params+0x70/0xe4) [ 639.010765] [<c09c39d4>] (dpcm_fe_dai_hw_params) from [<c099b274>] (snd_pcm_hw_params+0x158/0x418) [ 639.019750] [<c099b274>] (snd_pcm_hw_params) from [<c099c5a0>] (snd_pcm_ioctl+0x734/0x183c) [ 639.028129] [<c099c5a0>] (snd_pcm_ioctl) from [<c029ff94>] (ksys_ioctl+0x2ac/0xb98) [ 639.035812] [<c029ff94>] (ksys_ioctl) from [<c0100080>] (ret_fast_syscall+0x0/0x28) [ 639.043490] Exception stack(0xec529fa8 to 0xec529ff0) [ 639.048565] 9fa0: bee84650 01321870 00000004 c25c4111 bee84650 0002000f [ 639.056766] 9fc0: bee84650 01321870 01321820 00000036 00001f40 00000000 0002c2f8 00000003 [ 639.064964] 9fe0: b6f483fc bee8451c b6ee2655 b6e1dcf8 [ 639.070339] fsl-esai-dai 2024000.esai: Cannot create DMA dma:tx symlink Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/imx-pcm-dma.c | 173 +++++++++++++++++++++++++++++++++--- 1 file changed, 159 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 04a9bc749016..dae53b384df4 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -11,6 +11,7 @@ #include <linux/dmaengine.h> #include <linux/types.h> #include <linux/module.h> +#include <linux/dma-mapping.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,24 +30,168 @@ static bool filter(struct dma_chan *chan, void *param) return true; } -static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { - .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = filter, -}; +static int imx_pcm_hw_params(struct snd_soc_component *component, + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_dmaengine_dai_dma_data *dma_data; + struct dma_slave_config config; + struct dma_chan *chan; + int ret = 0; -int imx_pcm_dma_init(struct platform_device *pdev, size_t size) + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + chan = snd_dmaengine_pcm_get_chan(substream); + if (!chan) + return -EINVAL; + + ret = snd_hwparams_to_dma_slave_config(substream, params, &config); + if (ret) + return ret; + + dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); + if (!dma_data) + return -EINVAL; + + snd_dmaengine_pcm_set_config_from_dai_data(substream, + dma_data, + &config); + return dmaengine_slave_config(chan, &config); +} + +static int imx_pcm_hw_free(struct snd_soc_component *component, + struct snd_pcm_substream *substream) { - struct snd_dmaengine_pcm_config *config; + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static snd_pcm_uframes_t imx_pcm_pointer(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + return snd_dmaengine_pcm_pointer(substream); +} + +static int imx_pcm_trigger(struct snd_soc_component *component, + struct snd_pcm_substream *substream, int cmd) +{ + return snd_dmaengine_pcm_trigger(substream, cmd); +} + +static int imx_pcm_open(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct snd_dmaengine_dai_dma_data *dma_data; + struct device *dev = component->dev; + struct snd_pcm_hardware hw; + struct dma_chan *chan; + int ret; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(dev, "failed to set pcm hw params periods\n"); + return ret; + } + + dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); + if (!dma_data) + return -EINVAL; + + chan = dma_request_slave_channel(cpu_dai->dev, tx ? "tx" : "rx"); + if (!chan) { + /* Try to request channel using compat_filter_fn */ + chan = snd_dmaengine_pcm_request_channel(filter, + dma_data->filter_data); + if (!chan) + return -ENXIO; + } - config = devm_kzalloc(&pdev->dev, - sizeof(struct snd_dmaengine_pcm_config), GFP_KERNEL); - if (!config) - return -ENOMEM; - *config = imx_dmaengine_pcm_config; + ret = snd_dmaengine_pcm_open(substream, chan); + if (ret) + goto pcm_open_fail; - return devm_snd_dmaengine_pcm_register(&pdev->dev, - config, - SND_DMAENGINE_PCM_FLAG_COMPAT); + memset(&hw, 0, sizeof(hw)); + hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED; + hw.periods_min = 2; + hw.periods_max = UINT_MAX; + hw.period_bytes_min = 256; + hw.period_bytes_max = dma_get_max_seg_size(chan->device->dev); + hw.buffer_bytes_max = IMX_DEFAULT_DMABUF_SIZE; + hw.fifo_size = dma_data->fifo_size; + + /* Refine the hw according to caps of DMA. */ + ret = snd_dmaengine_pcm_refine_runtime_hwparams(substream, + dma_data, + &hw, + chan); + if (ret < 0) + goto refine_runtime_hwparams_fail; + + snd_soc_set_runtime_hwparams(substream, &hw); + + /* Support allocate memory from IRAM */ + ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_IRAM, + chan->device->dev, + hw.buffer_bytes_max, + &substream->dma_buffer); + if (ret < 0) + goto alloc_pagas_fail; + + return 0; + +alloc_pagas_fail: +refine_runtime_hwparams_fail: + snd_dmaengine_pcm_close(substream); +pcm_open_fail: + dma_release_channel(chan); + + return ret; +} + +static int imx_pcm_close(struct snd_soc_component *component, + struct snd_pcm_substream *substream) +{ + if (substream) { + snd_dma_free_pages(&substream->dma_buffer); + substream->dma_buffer.area = NULL; + substream->dma_buffer.addr = 0; + } + + return snd_dmaengine_pcm_close_release_chan(substream); +} + +static int imx_pcm_new(struct snd_soc_component *component, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + + return dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); +} + +static const struct snd_soc_component_driver imx_pcm_component = { + .name = "imx-pcm-dma", + .pcm_construct = imx_pcm_new, + .open = imx_pcm_open, + .close = imx_pcm_close, + .hw_params = imx_pcm_hw_params, + .hw_free = imx_pcm_hw_free, + .trigger = imx_pcm_trigger, + .pointer = imx_pcm_pointer, +}; + +int imx_pcm_dma_init(struct platform_device *pdev, size_t size) +{ + return devm_snd_soc_register_component(&pdev->dev, + &imx_pcm_component, NULL, 0); } EXPORT_SYMBOL_GPL(imx_pcm_dma_init); -- 2.21.0

5 17

[PATCH v6 0/5] Enable rk3066a HDMI sound
by Johan Jonker 22 Mar '21

22 Mar '21

Update the rk3066a HDMI documents with a #sound-dai-cells property. Include the code for sound in the HDMI driver. Add a simple-sound-card compatible node to rk3066a.dtsi, because I2S0 and HDMI TX are connected internally. And as last enable rk3066a HDMI sound in the rk3066a-mk808.dts file. make ARCH=arm CROSS_COMPILE=/usr/bin/arm-linux-gnueabi- -j4 cp ./arch/arm/boot/zImage ../zImage-dtb cat ./arch/arm/boot/dts/rk3066a-mk808.dtb >> ../zImage-dtb ../tools/rkcrc -k ../zImage-dtb ../mk808.img sudo ../tools/rkflashtool w 0x4000 0x8000 < ../mk808.img sudo ../tools/rkflashtool b Changed v6: remove patches that are applied to linux-next add platform_device_unregister() restyle Changed v5: removed unused variable fill frame structure Johan Jonker (4): dt-bindings: display: add #sound-dai-cells property to rockchip rk3066 hdmi ARM: dts: rockchip: rk3066a: add #sound-dai-cells to hdmi node ARM: dts: rockchip: add hdmi-sound node to rk3066a.dtsi ARM: dts: rockchip: enable hdmi_sound and i2s0 for rk3066a-mk808 Zheng Yang (1): drm: rockchip: add sound support to rk3066 hdmi driver .../display/rockchip/rockchip,rk3066-hdmi.yaml | 4 + arch/arm/boot/dts/rk3066a-mk808.dts | 8 + arch/arm/boot/dts/rk3066a.dtsi | 17 ++ drivers/gpu/drm/rockchip/Kconfig | 2 + drivers/gpu/drm/rockchip/rk3066_hdmi.c | 277 ++++++++++++++++++++- 5 files changed, 307 insertions(+), 1 deletion(-) -- 2.11.0

2 7

ASoc: soc_core.c stream direction from snd_soc_dai
by Matt Flax 26 Feb '21

26 Feb '21

Hi there, A large number of audio codecs allow different formats for playback and capture. This becomes very useful when there are different latencies between playback and capture hardware data lines. For example digital isolation chips typically have a 1 bit delay in propagation as the bit clock rate gets faster for higher sample rates. By setting the capture and playback formats to differ by one or two bit clock cycles, the delay problem is solved. There doesn't seem to be a simple way to detect stream direction in the codec driver's set_fmt function. The snd_soc_runtime_set_dai_fmt function : https://github.com/torvalds/linux/blob/master/sound/soc/soc-core.c#L1480 calls the snd_soc_dai_set_fmt function : https://github.com/torvalds/linux/blob/master/sound/soc/soc-dai.c#L101 which calls the set_fmt function : https://github.com/torvalds/linux/blob/master/include/sound/soc-dai.h#L189 The snd_soc_dai_ops set_fmt function is defined as : int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); Is there a simple way to find the stream direction from a snd_soc_dai ? If the stream direction can be detected then the playback and capture formats can be set independently for the codec. It there a different way to set the playback and capture formats for the codec independently at runtime, depending on the sample rate ? Matt

5 8

[PATCH alsa-ucm-conf 1/3] chtnau8824: Move DAC Channel Source selection to Speaker/Headphones EnableSeq
by Hans de Goede 23 Jan '21

23 Jan '21

Move DAC Channel Source selection to Speaker/Headphones EnableSeq. The main reason for doing this is to make it easier to allow using variables, as the Speaker/Headphones conf files are evaluated after the main HiFi.conf had a chance to set them. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- ucm2/codecs/nau8824/EnableSeq.conf | 3 --- ucm2/codecs/nau8824/HeadPhones.conf | 2 ++ ucm2/codecs/nau8824/MonoSpeaker.conf | 2 ++ ucm2/codecs/nau8824/Speaker.conf | 2 ++ 4 files changed, 6 insertions(+), 3 deletions(-) diff --git a/ucm2/codecs/nau8824/EnableSeq.conf b/ucm2/codecs/nau8824/EnableSeq.conf index 3650d06..2595e5f 100644 --- a/ucm2/codecs/nau8824/EnableSeq.conf +++ b/ucm2/codecs/nau8824/EnableSeq.conf @@ -1,7 +1,4 @@ EnableSequence [ - # Playback TDM configuration - cset "name='DACL Channel Source' 0" - cset "name='DACR Channel Source' 1" # Input Configuration cset "name='DMIC1 Enable Switch' off" cset "name='DMIC2 Enable Switch' off" diff --git a/ucm2/codecs/nau8824/HeadPhones.conf b/ucm2/codecs/nau8824/HeadPhones.conf index a807f84..46ec5d7 100644 --- a/ucm2/codecs/nau8824/HeadPhones.conf +++ b/ucm2/codecs/nau8824/HeadPhones.conf @@ -12,6 +12,8 @@ SectionDevice."Headphones" { ] EnableSequence [ + cset "name='DACL Channel Source' 0" + cset "name='DACR Channel Source' 1" cset "name='Headphone Switch' on" ] diff --git a/ucm2/codecs/nau8824/MonoSpeaker.conf b/ucm2/codecs/nau8824/MonoSpeaker.conf index 6b4ef8a..09edea5 100644 --- a/ucm2/codecs/nau8824/MonoSpeaker.conf +++ b/ucm2/codecs/nau8824/MonoSpeaker.conf @@ -11,6 +11,8 @@ SectionDevice."Speaker" { ] EnableSequence [ + cset "name='DACL Channel Source' 0" + cset "name='DACR Channel Source' 1" # nau8824 mono speaker boards have the speaker on the right chan cset "name='Speaker Left DACL Volume' 0" cset "name='Speaker Left DACR Volume' 0" diff --git a/ucm2/codecs/nau8824/Speaker.conf b/ucm2/codecs/nau8824/Speaker.conf index a995873..64d7eee 100644 --- a/ucm2/codecs/nau8824/Speaker.conf +++ b/ucm2/codecs/nau8824/Speaker.conf @@ -11,6 +11,8 @@ SectionDevice."Speaker" { ] EnableSequence [ + cset "name='DACL Channel Source' 0" + cset "name='DACR Channel Source' 1" cset "name='Speaker Left DACL Volume' 1" cset "name='Speaker Left DACR Volume' 0" cset "name='Speaker Right DACL Volume' 0" -- 2.28.0

2 6

[PATCH] ASoC: Intel: Skylake: Check the kcontrol against NULL
by Lukasz Majczak 20 Jan '21

20 Jan '21

There is no check for the kcontrol against NULL and in some cases it causes kernel to crash. Fixes: 2d744ecf2b984 ("ASoC: Intel: Skylake: Automatic DMIC format configuration according to information from NHLT") Cc: <stable(a)vger.kernel.org> # 5.4+ Signed-off-by: Lukasz Majczak <lma(a)semihalf.com> --- sound/soc/intel/skylake/skl-topology.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index ae466cd592922..c9abbe4ff0ba3 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3618,12 +3618,18 @@ static void skl_tplg_complete(struct snd_soc_component *component) int i; list_for_each_entry(dobj, &component->dobj_list, list) { - struct snd_kcontrol *kcontrol = dobj->control.kcontrol; - struct soc_enum *se = - (struct soc_enum *)kcontrol->private_value; - char **texts = dobj->control.dtexts; + struct snd_kcontrol *kcontrol; + struct soc_enum *se; + char **texts; char chan_text[4]; + kcontrol = dobj->control.kcontrol; + if(!kcontrol) + continue; + + se = (struct soc_enum *)kcontrol->private_value; + texts = dobj->control.dtexts; + if (dobj->type != SND_SOC_DOBJ_ENUM || dobj->control.kcontrol->put != skl_tplg_multi_config_set_dmic) base-commit: 69fe63aa100220c8fd1f451dd54dd0895df1441d -- 2.25.1

7 9

[PATCH] Add basic capture support for Pioneer DJ DJM-250MK2
by František Kučera 20 Jan '21

20 Jan '21

From: František Kučera <franta-linux(a)frantovo.cz> This patch extends support for DJM-250MK2 and allows recording. However, DVS is not possible yet (see the comment in code). Signed-off-by: František Kučera <franta-linux(a)frantovo.cz> --- sound/usb/pcm.c | 1 + sound/usb/quirks-table.h | 62 ++++++++++++++++++++++++++++++++++------ sound/usb/quirks.c | 1 + 3 files changed, 56 insertions(+), 8 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 5600751803cf..f5fa11deea4d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -374,6 +374,7 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ifnum = 2; goto add_sync_ep_from_ifnum; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ ep = 0x82; ifnum = 0; goto add_sync_ep_from_ifnum; diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index f4fb002e3ef4..c6b2f0b8d3e2 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3549,14 +3549,40 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), { /* * Pioneer DJ DJM-250MK2 - * PCM is 8 channels out @ 48 fixed (endpoints 0x01). - * The output from computer to the mixer is usable. - * - * The input (phono or line to computer) is not working. - * It should be at endpoint 0x82 and probably also 8 channels, - * but it seems that it works only with Pioneer proprietary software. - * Even on officially supported OS, the Audacity was unable to record - * and Mixxx to recognize the control vinyls. + * PCM is 8 channels out @ 48 fixed (endpoint 0x01) + * and 8 channels in @ 48 fixed (endpoint 0x82). + * + * Both playback and recording is working, even simultaneously. + * + * Playback channels could be mapped to: + * - CH1 + * - CH2 + * - AUX + * + * Recording channels could be mapped to: + * - Post CH1 Fader + * - Post CH2 Fader + * - Cross Fader A + * - Cross Fader B + * - MIC + * - AUX + * - REC OUT + * + * There is remaining problem with recording directly from PHONO/LINE. + * If we map a channel to: + * - CH1 Control Tone PHONO + * - CH1 Control Tone LINE + * - CH2 Control Tone PHONO + * - CH2 Control Tone LINE + * it is silent. + * There is no signal even on other operating systems with official drivers. + * The signal appears only when a supported application is started. + * This needs to be investigated yet... + * (there is quite a lot communication on the USB in both directions) + * + * In current version this mixer could be used for playback + * and for recording from vinyls (through Post CH* Fader) + * but not for DVS (Digital Vinyl Systems) like in Mixxx. */ USB_DEVICE_VENDOR_SPEC(0x2b73, 0x0017), .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { @@ -3580,6 +3606,26 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .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 = 8, // inputs + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .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 } } }, { diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index abf99b814a0f..6a0be7e7f4fe 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1493,6 +1493,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, set_format_emu_quirk(subs, fmt); break; case USB_ID(0x2b73, 0x000a): /* Pioneer DJ DJM-900NXS2 */ + case USB_ID(0x2b73, 0x0017): /* Pioneer DJ DJM-250MK2 */ pioneer_djm_set_format_quirk(subs); break; case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ -- 2.20.1

5 9

[PATCH 00/14] MFD/extcon/ASoC: Add support for Intel Bay Trail boards with WM5102 codec
by Hans de Goede 16 Jan '21

16 Jan '21

Hi All, This patch series adds support for Intel Bay Trail based device which use a WM5102 codec for audio output/input. This was developed and tested on a Lenovo Yoga Tablet 1051L. This series consists of 3 parts: 1. Arizona MFD drv patches for ACPI bindings, better jack-detect support and misc. fixes 2. extcon-arizona driver fixes and improved jack reporting (this depends on the MFD changes) 3. ASoC patches in the form of a quirk for BYTCR detecting, a new machine driver for BYT + WM5102 and jack-detect support for the new machine driver (which again depends on the MFD changes). Given that 2. and 3. depend on the MFD changes I believe that it is best if all patches in this series are merged through the MFD tree (once reviewed and acked) and then Lee can provide a immutable branch for the ASoC and extcon maintainers to merge into their trees. I have a patch with matching UCM profile changes available here: https://github.com/jwrdegoede/alsa-ucm-conf/commit/316109e7814926ba984322c1… This series + the UCM profile has been tested with both the SST and SOF ASoC drivers for BYT devices. Regards, Hans BugLink: https://github.com/thesofproject/linux/issues/2485

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[PATCH v11 0/7] Qualcomm's lpass-hdmi ASoC driver to support audio over dp port
by Srinivasa Rao Mandadapu 14 Jan '21

14 Jan '21

These patches are to support audio over DP port on Qualcomm's SC7180 LPASS Asoc. It includes machine driver, cpu driver, platform driver updates for HDMI path support, device tree documention, lpass variant structure optimization and configuration changes. These patches depends on the DP patch series https://patchwork.kernel.org/project/dri-devel/list/?series=332029 https://lore.kernel.org/patchwork/project/lkml/list/?series=464856 changes since V10: -- Moved hdmi regmap functions from lpass-hdmi.c to lpass-cpu.c -- Moved QCOM_REGMAP_FIELD_ALLOC macro from lpass-hdmi.c to lpass.h changes since V9: -- Removed unused structures lpass_hdmi.h changes since V8: -- Removed redundant structure wrapper for reg map field memebrs -- Updated lpass_hdmi_regmap_volatile API with appropriate registers as true and others as false. changes since V7: -- Fixed typo errors -- Created Separate patch for buffer size change changes since V6: -- Removed compile time define flag, which used for enabling HDMI code, based on corresponding config param is included. -- Updated reg map alloc API with reg map bulk API. -- Removed unnecessary line splits changes since V5: -- Removed unused struct regmap *map in lpass_platform_alloc_hdmidmactl_fields. -- DMA alloc and free API signature change in lpass-apq8016.c, lpass-ipq806x.c -- Keeping API "irqreturn_t lpass_platform_hdmiif_irq" under ifdef macro Changes Since v4: -- Updated with single compatible node for both I2S and HDMI. Changes Since v3: -- Removed id in lpass variant structure and used snd_soc_dai_driver id. Changes Since v2: -- Audio buffer size(i.e. LPASS_PLATFORM_BUFFER_SIZE) in lpass-platform.c increased. Changes Since v1: -- Commit messages are updated -- Addressed Rob Herring review comments V Sujith Kumar Reddy (7): ASoC: Add sc7180-lpass binding header hdmi define ASoC: dt-bindings: Add dt binding for lpass hdmi Asoc:qcom:lpass-cpu:Update dts property read API Asoc: qcom: lpass:Update lpaif_dmactl members order ASoC: qcom: Add support for lpass hdmi driver Asoc: qcom: lpass-platform : Increase buffer size ASoC: qcom: sc7180: Add support for audio over DP .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 74 ++-- include/dt-bindings/sound/sc7180-lpass.h | 1 + sound/soc/qcom/Kconfig | 5 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/lpass-apq8016.c | 4 +- sound/soc/qcom/lpass-cpu.c | 249 ++++++++++++- sound/soc/qcom/lpass-hdmi.c | 258 ++++++++++++++ sound/soc/qcom/lpass-hdmi.h | 102 ++++++ sound/soc/qcom/lpass-ipq806x.c | 4 +- sound/soc/qcom/lpass-lpaif-reg.h | 49 ++- sound/soc/qcom/lpass-platform.c | 395 +++++++++++++++++---- sound/soc/qcom/lpass-sc7180.c | 116 +++++- sound/soc/qcom/lpass.h | 124 ++++++- 13 files changed, 1240 insertions(+), 143 deletions(-) create mode 100644 sound/soc/qcom/lpass-hdmi.c create mode 100644 sound/soc/qcom/lpass-hdmi.h -- 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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USB Wireless device support
by Bastien Nocera 13 Jan '21

13 Jan '21

Hey, I've recently bought a SteelSeries Arctis 1 wireless headset: https://steelseries.com/gaming-headsets/arctis-1-wireless-xbox and wanted to write a kernel driver to export a few features of the device, and needed some help getting started. The receiver shows up as a audio class USB device, with extra features on a separate USB interface.[1] The extra features that can be accessed are: - Presence detection. I wanted to use that so PulseAudio or Pipewire could make routing decisions based on whether the headset is available, or probably better, implementing a "jack detection" emulation for the main audio interface? - A slider to change the sidetone volume - Export the battery level The latter is the only one I know how to do, so I'd be happy with any pointers or examples about the first two items. Cheers [1]: User-space code that accesses those features: https://github.com/Sapd/HeadsetControl/blob/master/src/devices/steelseries_…

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