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

[PATCH v2 0/4] Convert Rockchip ASoC drivers to GPIO descriptors
by Linus Walleij 09 Oct '23

09 Oct '23

The Rockchip drivers are pretty straight-forward to convert over to using GPIO descriptors. Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- Changes in v2: - Make the GPIOs properly optional in the first patch, preserving the old semantics of the driver. - Link to v1: https://lore.kernel.org/r/20230928-descriptors-asoc-rockchip-v1-0-a142a42d4… --- Linus Walleij (4): ASoC: rockchip: Convert RK3288 HDMI to GPIO descriptors ASoC: rockchip: Drop includes from RK3399 ASoC: rockchip: Drop includes from Rockchip MAX98090 ASoC: rockchip: Drop includes from Rockchip RT5645 sound/soc/rockchip/rk3288_hdmi_analog.c | 46 ++++++++++----------------------- sound/soc/rockchip/rk3399_gru_sound.c | 2 -- sound/soc/rockchip/rockchip_max98090.c | 2 -- sound/soc/rockchip/rockchip_rt5645.c | 2 -- 4 files changed, 14 insertions(+), 38 deletions(-) --- base-commit: 0bb80ecc33a8fb5a682236443c1e740d5c917d1d change-id: 20230926-descriptors-asoc-rockchip-833a4fbc7c1f Best regards, -- Linus Walleij <linus.walleij(a)linaro.org>

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time to remove COPYING.GPL from alsa-plugins? (issue #45)
by Michael Wang 09 Oct '23

09 Oct '23

Hi Alsa-devs and Mr Iwai, Sorry for the spam, I am resending this in plain text mode this time. It seems the only thing in alsa-plugins that has a GPL license is rate/rate_samplerate.c, and that is due to the license of libsamplerate. But in 2016, libsamplerate dropped the commercial/GPL part of its license and was released with the 2-Clause BSD license. So I was wondering if the license for rate_samplerate.c can be updated to remove the GPL license, which means the COPYING.GPL file can also be removed from the alsa-plugins package? I have created issue #54 in github.com/alsa-project/alsa-plugins, but so far, no action. Should I submit a pull request? Thanks, Michael -- This electronic communication and the information and any files transmitted with it, or attached to it, are confidential and are intended solely for the use of the individual or entity to whom it is addressed and may contain information that is confidential, legally privileged, protected by privacy laws, or otherwise restricted from disclosure to anyone else. If you are not the intended recipient or the person responsible for delivering the e-mail to the intended recipient, you are hereby notified that any use, copying, distributing, dissemination, forwarding, printing, or copying of this e-mail is strictly prohibited. If you received this e-mail in error, please return the e-mail to the sender, delete it from your computer, and destroy any printed copy of it.

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[PATCH v2 1/2] ASoC: dt-bindings: fsl_rpmsg: List DAPM endpoints ignoring system suspend
by Chancel Liu 09 Oct '23

09 Oct '23

Add a property to list DAPM endpoints which mark paths between these endpoints should not be disabled when system enters in suspend state. LPA means low power audio case. On asymmetric multiprocessor, there are Cortex-A core and Cortex-M core, Linux is running on Cortex-A core, RTOS or other OS is running on Cortex-M core. The audio hardware devices can be controlled by Cortex-M. LPA can be explained as a mechanism that Cortex-A allocates a large buffer and fill audio data, then Cortex-A can enter into suspend for the purpose of power saving. Cortex-M continues to play the sound during suspend phase of Cortex-A. When the data in buffer is consumed, Cortex-M will trigger the Cortex-A to wakeup to fill data. LPA requires some audio paths still enabled when Cortex-A enters into suspend. Signed-off-by: Chancel Liu <chancel.liu(a)nxp.com> --- .../devicetree/bindings/sound/fsl,rpmsg.yaml | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml index 188f38baddec..d8fd17615bf2 100644 --- a/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml +++ b/Documentation/devicetree/bindings/sound/fsl,rpmsg.yaml @@ -91,6 +91,21 @@ properties: - rpmsg-audio-channel - rpmsg-micfil-channel + fsl,lpa-widgets: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: | + A list of DAPM endpoints which mark paths between these endpoints should + not be disabled when system enters in suspend state. LPA means low power + audio case. On asymmetric multiprocessor, there are Cortex-A core and + Cortex-M core, Linux is running on Cortex-A core, RTOS or other OS is + running on Cortex-M core. The audio hardware devices can be controlled by + Cortex-M. LPA can be explained as a mechanism that Cortex-A allocates a + large buffer and fill audio data, then Cortex-A can enter into suspend + for the purpose of power saving. Cortex-M continues to play the sound + during suspend phase of Cortex-A. When the data in buffer is consumed, + Cortex-M will trigger the Cortex-A to wakeup to fill data. LPA requires + some audio paths still enabled when Cortex-A enters into suspend. + required: - compatible -- 2.25.1

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[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

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[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

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[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

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[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

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[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

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