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

[RFC PATCH v5 00/11] Add audio support in v4l2 framework
by Shengjiu Wang 09 Oct '23

09 Oct '23

Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. It is a specific use case, there is no reference in current kernel. v4l2 memory to memory is the closed implementation, v4l2 current support video, image, radio, tuner, touch devices, so it is not complicated to add support for this specific audio case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the first 3 patches is for refining the code to make it can be shared by the "memory->asrc->memory" driver. The main change is in the v4l2 side, A /dev/vl4-audioX will be created, user applications only use the ioctl of v4l2 framework. Other change is to add memory to memory support for two kinds of i.MX ASRC module. changes in v5: - remove V4L2_AUDIO_FMT_LPCM - define audio pixel format like V4L2_AUDIO_FMT_S8... - remove rate and format in struct v4l2_audio_format. - Add V4L2_CID_ASRC_SOURCE_RATE and V4L2_CID_ASRC_DEST_RATE controls - updata document accordingly. changes in v4: - update document style - separate V4L2_AUDIO_FMT_LPCM and V4L2_CAP_AUDIO_M2M in separate commit changes in v3: - Modify documents for adding audio m2m support - Add audio virtual m2m driver - Defined V4L2_AUDIO_FMT_LPCM format type for audio. - Defined V4L2_CAP_AUDIO_M2M capability type for audio m2m case. - with modification in v4l-utils, pass v4l2-compliance test. changes in v2: - decouple the implementation in v4l2 and ALSA - implement the memory to memory driver as a platfrom driver and move it to driver/media - move fsl_asrc_common.h to include/sound folder Shengjiu Wang (11): ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc: move fsl_asrc_common.h to include/sound ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device media: uapi: Add V4L2_CAP_AUDIO_M2M capability flag media: v4l2: Add audio capture and output support media: uapi: Add audio rate controls support media: uapi: define audio sample format fourcc type media: imx-asrc: Add memory to memory driver media: audm2m: add virtual driver for audio memory to memory .../userspace-api/media/v4l/buffer.rst | 6 + .../userspace-api/media/v4l/common.rst | 1 + .../media/v4l/dev-audio-mem2mem.rst | 71 + .../userspace-api/media/v4l/devices.rst | 1 + .../media/v4l/ext-ctrls-asrc-rate.rst | 36 + .../userspace-api/media/v4l/pixfmt-audio.rst | 277 ++++ .../userspace-api/media/v4l/pixfmt.rst | 1 + .../media/v4l/vidioc-enum-fmt.rst | 2 + .../media/v4l/vidioc-g-ext-ctrls.rst | 4 + .../userspace-api/media/v4l/vidioc-g-fmt.rst | 4 + .../media/v4l/vidioc-querycap.rst | 3 + .../media/v4l/vidioc-queryctrl.rst | 7 + .../media/videodev2.h.rst.exceptions | 4 + .../media/common/videobuf2/videobuf2-v4l2.c | 4 + drivers/media/platform/nxp/Kconfig | 12 + drivers/media/platform/nxp/Makefile | 1 + drivers/media/platform/nxp/imx-asrc.c | 1251 +++++++++++++++++ drivers/media/test-drivers/Kconfig | 9 + drivers/media/test-drivers/Makefile | 1 + drivers/media/test-drivers/audm2m.c | 808 +++++++++++ drivers/media/v4l2-core/v4l2-ctrls-core.c | 5 + drivers/media/v4l2-core/v4l2-ctrls-defs.c | 4 + drivers/media/v4l2-core/v4l2-dev.c | 17 + drivers/media/v4l2-core/v4l2-ioctl.c | 104 ++ include/media/v4l2-ctrls.h | 1 + include/media/v4l2-dev.h | 2 + include/media/v4l2-ioctl.h | 34 + .../fsl => include/sound}/fsl_asrc_common.h | 58 + include/uapi/linux/v4l2-controls.h | 13 + include/uapi/linux/videodev2.h | 77 + sound/soc/fsl/fsl_asrc.c | 136 ++ sound/soc/fsl/fsl_asrc.h | 4 +- sound/soc/fsl/fsl_asrc_dma.c | 2 +- sound/soc/fsl/fsl_easrc.c | 214 +++ sound/soc/fsl/fsl_easrc.h | 8 +- 35 files changed, 3179 insertions(+), 3 deletions(-) create mode 100644 Documentation/userspace-api/media/v4l/dev-audio-mem2mem.rst create mode 100644 Documentation/userspace-api/media/v4l/ext-ctrls-asrc-rate.rst create mode 100644 Documentation/userspace-api/media/v4l/pixfmt-audio.rst create mode 100644 drivers/media/platform/nxp/imx-asrc.c create mode 100644 drivers/media/test-drivers/audm2m.c rename {sound/soc/fsl => include/sound}/fsl_asrc_common.h (60%) -- 2.34.1

4 18

[PATCH] ALSA: scarlett2: Add Focusrite Clarett+ 2Pre and 4Pre support
by Geoffrey D. Bennett 07 Oct '23

07 Oct '23

The Focusrite Clarett+ series uses the same protocol as the Scarlett Gen 2 and Gen 3 series. This patch adds support for the Clarett+ 2Pre and Clarett+ 4Pre similarly to the existing 8Pre support by adding appropriate entries to the scarlett2 driver. The Clarett 2Pre USB and 4Pre USB presumably use the same protocol as well, so support for them can easily be added if someone can test. Signed-off-by: Geoffrey D. Bennett <g(a)b4.vu> --- sound/usb/mixer_quirks.c | 2 + sound/usb/mixer_scarlett_gen2.c | 97 ++++++++++++++++++++++++++++++++- 2 files changed, 98 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 9911859d2750..0db94ead1b93 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -3421,6 +3421,8 @@ int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) case USB_ID(0x1235, 0x8214): /* Focusrite Scarlett 18i8 3rd Gen */ case USB_ID(0x1235, 0x8215): /* Focusrite Scarlett 18i20 3rd Gen */ case USB_ID(0x1235, 0x8208): /* Focusrite Clarett 8Pre USB */ + case USB_ID(0x1235, 0x820a): /* Focusrite Clarett+ 2Pre */ + case USB_ID(0x1235, 0x820b): /* Focusrite Clarett+ 4Pre */ case USB_ID(0x1235, 0x820c): /* Focusrite Clarett+ 8Pre */ err = snd_scarlett_gen2_init(mixer); break; diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c index 481a80e160de..57e7fdff6905 100644 --- a/sound/usb/mixer_scarlett_gen2.c +++ b/sound/usb/mixer_scarlett_gen2.c @@ -6,7 +6,7 @@ * - 6i6/18i8/18i20 Gen 2 * - Solo/2i2/4i4/8i6/18i8/18i20 Gen 3 * - Clarett 8Pre USB - * - Clarett+ 8Pre + * - Clarett+ 2Pre/4Pre/8Pre * * Copyright (c) 2018-2023 by Geoffrey D. Bennett <g at b4.vu> * Copyright (c) 2020-2021 by Vladimir Sadovnikov <sadko4u(a)gmail.com> @@ -60,6 +60,10 @@ * Support for Clarett USB 8Pre added in Sep 2023 (thanks to Philippe * Perrot for confirmation). * + * Support for Clarett+ 4Pre and 2Pre added in Sep 2023 (thanks to + * Gregory Rozzo for donating a 4Pre, and David Sherwood and Patrice + * Peterson for usbmon output). + * * This ALSA mixer gives access to (model-dependent): * - input, output, mixer-matrix muxes * - mixer-matrix gain stages @@ -832,6 +836,95 @@ static const struct scarlett2_device_info s18i20_gen3_info = { } }, }; +static const struct scarlett2_device_info clarett_2pre_info = { + .config_set = SCARLETT2_CONFIG_SET_CLARETT, + .line_out_hw_vol = 1, + .level_input_count = 2, + .air_input_count = 2, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones L", + "Headphones R", + }, + + .port_count = { + [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, + [SCARLETT2_PORT_TYPE_ANALOGUE] = { 2, 4 }, + [SCARLETT2_PORT_TYPE_SPDIF] = { 2, 0 }, + [SCARLETT2_PORT_TYPE_ADAT] = { 8, 0 }, + [SCARLETT2_PORT_TYPE_MIX] = { 10, 18 }, + [SCARLETT2_PORT_TYPE_PCM] = { 4, 12 }, + }, + + .mux_assignment = { { + { SCARLETT2_PORT_TYPE_PCM, 0, 12 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 18 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 8 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_PCM, 0, 8 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 4 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 18 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 8 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_PCM, 0, 2 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 4 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 26 }, + { 0, 0, 0 }, + } }, +}; + +static const struct scarlett2_device_info clarett_4pre_info = { + .config_set = SCARLETT2_CONFIG_SET_CLARETT, + .line_out_hw_vol = 1, + .level_input_count = 2, + .air_input_count = 4, + + .line_out_descrs = { + "Monitor L", + "Monitor R", + "Headphones 1 L", + "Headphones 1 R", + "Headphones 2 L", + "Headphones 2 R", + }, + + .port_count = { + [SCARLETT2_PORT_TYPE_NONE] = { 1, 0 }, + [SCARLETT2_PORT_TYPE_ANALOGUE] = { 8, 6 }, + [SCARLETT2_PORT_TYPE_SPDIF] = { 2, 2 }, + [SCARLETT2_PORT_TYPE_ADAT] = { 8, 0 }, + [SCARLETT2_PORT_TYPE_MIX] = { 10, 18 }, + [SCARLETT2_PORT_TYPE_PCM] = { 8, 18 }, + }, + + .mux_assignment = { { + { SCARLETT2_PORT_TYPE_PCM, 0, 18 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 }, + { SCARLETT2_PORT_TYPE_SPDIF, 0, 2 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 18 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 8 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_PCM, 0, 14 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 }, + { SCARLETT2_PORT_TYPE_SPDIF, 0, 2 }, + { SCARLETT2_PORT_TYPE_MIX, 0, 18 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 8 }, + { 0, 0, 0 }, + }, { + { SCARLETT2_PORT_TYPE_PCM, 0, 12 }, + { SCARLETT2_PORT_TYPE_ANALOGUE, 0, 6 }, + { SCARLETT2_PORT_TYPE_SPDIF, 0, 2 }, + { SCARLETT2_PORT_TYPE_NONE, 0, 24 }, + { 0, 0, 0 }, + } }, +}; + static const struct scarlett2_device_info clarett_8pre_info = { .config_set = SCARLETT2_CONFIG_SET_CLARETT, .line_out_hw_vol = 1, @@ -907,6 +1000,8 @@ static const struct scarlett2_device_entry scarlett2_devices[] = { /* Supported Clarett USB/Clarett+ devices */ { USB_ID(0x1235, 0x8208), &clarett_8pre_info, "Clarett USB" }, + { USB_ID(0x1235, 0x820a), &clarett_2pre_info, "Clarett+" }, + { USB_ID(0x1235, 0x820b), &clarett_4pre_info, "Clarett+" }, { USB_ID(0x1235, 0x820c), &clarett_8pre_info, "Clarett+" }, /* End of list */ -- 2.41.0

2 2

[PATCH 1/2] ALSA: aloop: Add support for the non-interleaved access mode
by Ivan Orlov 06 Oct '23

06 Oct '23

The current version of the loopback driver supports interleaved access mode only. This patch introduces support for the non-interleaved access mode. When in the interleaved mode, the 'copy_play_buf' function copies data from the playback to the capture buffer using one memcpy call. This call copies samples for multiple, interleaved channels. In the non-interleaved mode we have multiple channel buffers, so we have to perform multiple memcpy calls to copy samples channel after channel. Add new function called 'copy_play_buf_part_n', which copies a part of each channel buffer from playback to capture. Modify the 'copy_play_buf' to use the corresponding memory copy function(just memcpy / copy_play_buf_part_n) depending on the access mode. Signed-off-by: Ivan Orlov <ivan.orlov0322(a)gmail.com> --- sound/drivers/aloop.c | 31 ++++++++++++++++++++++++++++--- 1 file changed, 28 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index a38e602b4fc6..ab116b1fed96 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -158,6 +158,9 @@ struct loopback_pcm { unsigned long last_jiffies; /* If jiffies timer is used */ struct timer_list timer; + + /* size of per channel buffer in case of non-interleaved access */ + unsigned int channel_buf_n; }; static struct platform_device *devices[SNDRV_CARDS]; @@ -335,7 +338,8 @@ static int loopback_check_format(struct loopback_cable *cable, int stream) substream->runtime; check = runtime->format != cruntime->format || runtime->rate != cruntime->rate || - runtime->channels != cruntime->channels; + runtime->channels != cruntime->channels || + runtime->access != cruntime->access; if (!check) return 0; if (stream == SNDRV_PCM_STREAM_CAPTURE) { @@ -472,6 +476,7 @@ static int loopback_prepare(struct snd_pcm_substream *substream) dpcm->buf_pos = 0; dpcm->pcm_buffer_size = frames_to_bytes(runtime, runtime->buffer_size); + dpcm->channel_buf_n = dpcm->pcm_buffer_size / runtime->channels; if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { /* clear capture buffer */ dpcm->silent_size = dpcm->pcm_buffer_size; @@ -522,6 +527,22 @@ static void clear_capture_buf(struct loopback_pcm *dpcm, unsigned int bytes) } } +static void copy_play_buf_part_n(struct loopback_pcm *play, struct loopback_pcm *capt, + unsigned int size, unsigned int src_off, unsigned int dst_off) +{ + unsigned int channels = capt->substream->runtime->channels; + unsigned int size_p_ch = size / channels; + unsigned int src_off_ch = src_off / channels; + unsigned int dst_off_ch = dst_off / channels; + int i; + + for (i = 0; i < channels; i++) { + memcpy(capt->substream->runtime->dma_area + capt->channel_buf_n * i + dst_off_ch, + play->substream->runtime->dma_area + play->channel_buf_n * i + src_off_ch, + size_p_ch); + } +} + static void copy_play_buf(struct loopback_pcm *play, struct loopback_pcm *capt, unsigned int bytes) @@ -556,7 +577,11 @@ static void copy_play_buf(struct loopback_pcm *play, size = play->pcm_buffer_size - src_off; if (dst_off + size > capt->pcm_buffer_size) size = capt->pcm_buffer_size - dst_off; - memcpy(dst + dst_off, src + src_off, size); + if (runtime->access == SNDRV_PCM_ACCESS_RW_NONINTERLEAVED || + runtime->access == SNDRV_PCM_ACCESS_MMAP_NONINTERLEAVED) + copy_play_buf_part_n(play, capt, size, src_off, dst_off); + else + memcpy(dst + dst_off, src + src_off, size); capt->silent_size = 0; bytes -= size; if (!bytes) @@ -878,7 +903,7 @@ static const struct snd_pcm_hardware loopback_pcm_hardware = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME), + SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_NONINTERLEAVED), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | -- 2.34.1

3 6

[PATCH v3 0/4] ALSA: hda: Abstract and update HOST-stream setup procedure
by Cezary Rojewski 06 Oct '23

06 Oct '23

The patchset targets two intertwined topics: The driver shall poll SDxFIFOS to ensure a valid value is set by the controller after programming SDxFMT. Due to amount of users and limited-number of configuration available in our CI when compared to overall possibilities on the market, the check is non-blocking. Second topic relates to stream setup procedure. The procedure differs between HDAudio controller device revisions. Right now those differences are handled directly by a platform driver. Existing top-level 'if (pci->device == APL)' could be replaced by a abstraction in lower parts of the code instead. With that done, the two users are updated accordingly. In avs-driver case, this updates the flow to the recommended one. Changes in v3: - fixed issues pointed out by scripts/kernel-doc Changes in v2: - fixed ->host_setup assignment in patch 02/04 Cezary Rojewski (4): ALSA: hda: Poll SDxFIFOS after programming SDxFMT ALSA: hda: Introduce HOST stream setup mechanism ASoC: Intel: avs: Use helper to setup HOST stream ASoC: Intel: Skylake: Use helper to setup HOST stream include/sound/hda_register.h | 2 ++ include/sound/hdaudio.h | 3 +++ include/sound/hdaudio_ext.h | 3 +++ sound/hda/ext/hdac_ext_stream.c | 41 +++++++++++++++++++++++++++++++ sound/hda/hdac_stream.c | 8 ++++++ sound/soc/intel/avs/pcm.c | 2 +- sound/soc/intel/skylake/skl-pcm.c | 14 +---------- 7 files changed, 59 insertions(+), 14 deletions(-) -- 2.25.1

2 8

[PATCH][next] ALSA: 6fire: Fix undefined behavior bug in struct midi_runtime
by Gustavo A. R. Silva 06 Oct '23

06 Oct '23

`struct urb` is a flexible structure, which means that it contains a flexible-array member at the bottom. This could potentially lead to an overwrite of the objects following `out_urb` in `struct midi_runtime`, among them a function pointer. Fix this by placing the declaration of object `out_urb` at the end of `struct midi_runtime`. Fixes: c6d43ba816d1 ("ALSA: usb/6fire - Driver for TerraTec DMX 6Fire USB") Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/usb/6fire/midi.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/midi.h b/sound/usb/6fire/midi.h index 47640c845903..df3f41913274 100644 --- a/sound/usb/6fire/midi.h +++ b/sound/usb/6fire/midi.h @@ -22,12 +22,12 @@ struct midi_runtime { spinlock_t in_lock; spinlock_t out_lock; struct snd_rawmidi_substream *out; - struct urb out_urb; u8 out_serial; /* serial number of out packet */ u8 *out_buffer; int buffer_offset; void (*in_received)(struct midi_runtime *rt, u8 *data, int length); + struct urb out_urb; }; int usb6fire_midi_init(struct sfire_chip *chip); -- 2.34.1

4 3

[PATCH][next] ALSA: 6fire: Fix undefined behavior bug in struct comm_runtime
by Gustavo A. R. Silva 06 Oct '23

06 Oct '23

`struct urb` is a flexible structure, which means that it contains a flexible-array member at the bottom. This could potentially lead to an overwrite of the objects following `receiver` in `struct comm_runtime`, among them some function pointers. Fix this by placing the declaration of object `receiver` at the end of `struct comm_runtime`. Fixes: ddb6b5a96437 ("ALSA: 6fire: fix DMA issues with URB transfer_buffer usage") Cc: stable(a)vger.kernel.org Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/usb/6fire/comm.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/6fire/comm.h b/sound/usb/6fire/comm.h index 2447d7ecf179..ee81572a4eec 100644 --- a/sound/usb/6fire/comm.h +++ b/sound/usb/6fire/comm.h @@ -19,7 +19,6 @@ enum /* settings for comm */ struct comm_runtime { struct sfire_chip *chip; - struct urb receiver; u8 *receiver_buffer; u8 serial; /* urb serial */ @@ -30,6 +29,7 @@ struct comm_runtime { int (*write8)(struct comm_runtime *rt, u8 request, u8 reg, u8 value); int (*write16)(struct comm_runtime *rt, u8 request, u8 reg, u8 vh, u8 vl); + struct urb receiver; }; int usb6fire_comm_init(struct sfire_chip *chip); -- 2.34.1

4 3

[PATCH] sound: fix name of SIS7013 sound chip in comment
by Thomas Perl 06 Oct '23

06 Oct '23

While grep'ing for SIS7012, I noticed that there is only one reference to it in sound/pci/intel8x0m.c, while most of its code lives in sound/pci/intel8x0.c. This probably was a simple copy'n'paste mistake, as the sound/pci/intel8x0m.c driver implements support for SIS7013 (see DEVICE_SIS), and the two devices seem to share the same behavior / registers. Signed-off-by: Thomas Perl <m(a)thp.io> --- sound/pci/intel8x0m.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 2845cc006d0c..653ecca78238 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -918,7 +918,7 @@ static int snd_intel8x0m_ich_chip_init(struct intel8x0m *chip, int probing) } if (chip->device_type == DEVICE_SIS) { - /* unmute the output on SIS7012 */ + /* unmute the output on SIS7013 */ iputword(chip, 0x4c, igetword(chip, 0x4c) | 1); } -- 2.39.2

2 1

[PATCH v3] ALSA: pcm: oss: Fix race at SNDCTL_DSP_SETTRIGGER
by Ma Ke 06 Oct '23

06 Oct '23

There is a small race window at snd_pcm_oss_set_trigger() that is called from OSS PCM SNDCTL_DSP_SETTRIGGER ioctl; namely the function calls snd_pcm_oss_make_ready() at first, then takes the params_lock mutex for the rest. When the stream is set up again by another thread between them, it leads to inconsistency, and may result in unexpected results such as NULL dereference of OSS buffer as a fuzzer spotted recently. The fix is simply to cover snd_pcm_oss_make_ready() call into the same params_lock mutex with snd_pcm_oss_make_ready_locked() variant. Signed-off-by: Ma Ke <make_ruc2021(a)163.com> --- sound/core/oss/pcm_oss.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 728c211142d1..fd9d23c3684b 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -2083,21 +2083,16 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr psubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_PLAYBACK]; csubstream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; - if (psubstream) { - err = snd_pcm_oss_make_ready(psubstream); - if (err < 0) - return err; - } - if (csubstream) { - err = snd_pcm_oss_make_ready(csubstream); - if (err < 0) - return err; - } if (psubstream) { runtime = psubstream->runtime; cmd = 0; if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -ERESTARTSYS; + err = snd_pcm_oss_make_ready_locked(psubstream); + if (err < 0) { + mutex_unlock(&runtime->oss.params_lock); + return err; + } if (trigger & PCM_ENABLE_OUTPUT) { if (runtime->oss.trigger) goto _skip1; @@ -2128,6 +2123,11 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr cmd = 0; if (mutex_lock_interruptible(&runtime->oss.params_lock)) return -ERESTARTSYS; + err = snd_pcm_oss_make_ready_locked(csubstream); + if (err < 0) { + mutex_unlock(&runtime->oss.params_lock); + return err; + } if (trigger & PCM_ENABLE_INPUT) { if (runtime->oss.trigger) goto _skip2; -- 2.37.2

2 1

[PATCH v7 00/33] Introduce QC USB SND audio offloading support
by Wesley Cheng 05 Oct '23

05 Oct '23

Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Mathias Nyman (3): xhci: add support to allocate several interrupters xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (30): usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules dt-bindings: usb: dwc3: Limit num-hc-interrupters definition dt-bindings: usb: xhci: Add num-hc-interrupters definition usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ASoC: dt-bindings: Add Q6USB backend ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ASoC: qcom: qdsp6: q6afe: Split USB AFE dev_token param into separate API ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: qcom: qdsp6: Add SND kcontrol to select offload device ASoC: qcom: qdsp6: Add SND kcontrol for fetching offload status ASoC: qcom: qdsp6: Add headphone jack for offload connection status ALSA: usb-audio: qcom: Use card and PCM index from QMI request ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ASoC: usb: Rediscover USB SND devices on USB port add .../devicetree/bindings/sound/qcom,q6usb.yaml | 55 + .../bindings/sound/qcom,sm8250.yaml | 15 + .../devicetree/bindings/usb/snps,dwc3.yaml | 4 + .../devicetree/bindings/usb/usb-xhci.yaml | 6 + drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 5 +- drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 4 + drivers/usb/host/xhci-debugfs.c | 2 +- drivers/usb/host/xhci-hub.c | 29 +- drivers/usb/host/xhci-mem.c | 102 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 48 +- drivers/usb/host/xhci-sideband.c | 198 ++ drivers/usb/host/xhci.c | 111 +- drivers/usb/host/xhci.h | 105 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-intr.h | 86 + include/linux/usb/xhci-sideband.h | 57 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 51 + sound/soc/Makefile | 2 +- sound/soc/qcom/Kconfig | 4 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 56 + sound/soc/qcom/qdsp6/q6afe.c | 206 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 450 ++++ sound/soc/soc-usb.c | 200 ++ sound/usb/Kconfig | 15 + sound/usb/Makefile | 2 +- sound/usb/card.c | 123 ++ sound/usb/card.h | 23 + sound/usb/endpoint.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 94 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 2 + sound/usb/qcom/qc_audio_offload.c | 1854 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 45 files changed, 4896 insertions(+), 196 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6usb.yaml create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-intr.h create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

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[PATCH v2] ASoC: dt-bindings: rt5616: Convert to dtschema
by Bragatheswaran Manickavel 03 Oct '23

03 Oct '23

Convert the rt5616 audio CODEC bindings to DT schema Signed-off-by: Bragatheswaran Manickavel <bragathemanick0908(a)gmail.com> --- V1 -> V2: Changes codec to audio-codec and added spacing above .../bindings/sound/realtek,rt5616.yaml | 49 +++++++++++++++++++ .../devicetree/bindings/sound/rt5616.txt | 32 ------------ 2 files changed, 49 insertions(+), 32 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5616.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt5616.txt diff --git a/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml new file mode 100644 index 000000000000..a387f802afd1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/realtek,rt5616.yaml @@ -0,0 +1,49 @@ +# SPDX-License-Identifier: (GPL-2.0 OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/realtek,rt5616.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Realtek rt5616 ALSA SoC audio codec driver + +description: | + Pins on the device (for linking into audio routes) for RT5616: + + * IN1P + * IN2P + * IN2N + * LOUTL + * LOUTR + * HPOL + * HPOR + +maintainers: + - Bard Liao <bardliao(a)realtek.com> + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: realtek,rt5616 + + reg: + maxItems: 1 + +required: + - compatible + - reg + +unevaluatedProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + + audio-codec@1b { + compatible = "realtek,rt5616"; + reg = <0x1b>; + }; + }; diff --git a/Documentation/devicetree/bindings/sound/rt5616.txt b/Documentation/devicetree/bindings/sound/rt5616.txt deleted file mode 100644 index 540a4bf252e4..000000000000 --- a/Documentation/devicetree/bindings/sound/rt5616.txt +++ /dev/null @@ -1,32 +0,0 @@ -RT5616 audio CODEC - -This device supports I2C only. - -Required properties: - -- compatible : "realtek,rt5616". - -- reg : The I2C address of the device. - -Optional properties: - -- clocks: The phandle of the master clock to the CODEC. - -- clock-names: Should be "mclk". - -Pins on the device (for linking into audio routes) for RT5616: - - * IN1P - * IN2P - * IN2N - * LOUTL - * LOUTR - * HPOL - * HPOR - -Example: - -rt5616: codec@1b { - compatible = "realtek,rt5616"; - reg = <0x1b>; -}; -- 2.34.1

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