<feed xmlns='http://www.w3.org/2005/Atom'>
<title>user/sven/linux.git/include/sound, branch v6.14.8</title>
<subtitle>Linux Kernel
</subtitle>
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<updated>2025-05-22T12:31:43Z</updated>
<entry>
<title>ALSA: ump: Fix a typo of snd_ump_stream_msg_device_info</title>
<updated>2025-05-22T12:31:43Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2025-05-11T14:11:45Z</published>
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<id>urn:sha1:2cdceb7fb94adea7ac457a6d451edfb0097b5990</id>
<content type='text'>
[ Upstream commit dd33993a9721ab1dae38bd37c9f665987d554239 ]

s/devince/device/

It's used only internally, so no any behavior changes.

Fixes: 37e0e14128e0 ("ALSA: ump: Support UMP Endpoint and Function Block parsing")
Acked-by: Greg Kroah-Hartman &lt;gregkh@linuxfoundation.org&gt;
Link: https://patch.msgid.link/20250511141147.10246-1-tiwai@suse.de
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
</entry>
<entry>
<title>ALSA: ump: Fix buffer overflow at UMP SysEx message conversion</title>
<updated>2025-05-09T07:55:59Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2025-04-29T12:48:41Z</published>
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<id>urn:sha1:42ef48dd4ebb082a1a90b5c3feeda2e68a9e32fe</id>
<content type='text'>
[ Upstream commit 56f1f30e6795b890463d9b20b11e576adf5a2f77 ]

The conversion function from MIDI 1.0 to UMP packet contains an
internal buffer to keep the incoming MIDI bytes, and its size is 4, as
it was supposed to be the max size for a MIDI1 UMP packet data.
However, the implementation overlooked that SysEx is handled in a
different format, and it can be up to 6 bytes, as found in
do_convert_to_ump().  It leads eventually to a buffer overflow, and
may corrupt the memory when a longer SysEx message is received.

The fix is simply to extend the buffer size to 6 to fit with the SysEx
UMP message.

Fixes: 0b5288f5fe63 ("ALSA: ump: Add legacy raw MIDI support")
Reported-by: Argusee &lt;vr@darknavy.com&gt;
Link: https://patch.msgid.link/20250429124845.25128-1-tiwai@suse.de
Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
Signed-off-by: Sasha Levin &lt;sashal@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: ops: Consistently treat platform_max as control value</title>
<updated>2025-03-05T17:25:25Z</updated>
<author>
<name>Charles Keepax</name>
<email>ckeepax@opensource.cirrus.com</email>
</author>
<published>2025-02-28T15:14:56Z</published>
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<id>urn:sha1:0eba2a7e858907a746ba69cd002eb9eb4dbd7bf3</id>
<content type='text'>
This reverts commit 9bdd10d57a88 ("ASoC: ops: Shift tested values in
snd_soc_put_volsw() by +min"), and makes some additional related
updates.

There are two ways the platform_max could be interpreted; the maximum
register value, or the maximum value the control can be set to. The
patch moved from treating the value as a control value to a register
one. When the patch was applied it was technically correct as
snd_soc_limit_volume() also used the register interpretation. However,
even then most of the other usages treated platform_max as a
control value, and snd_soc_limit_volume() has since been updated to
also do so in commit fb9ad24485087 ("ASoC: ops: add correct range
check for limiting volume"). That patch however, missed updating
snd_soc_put_volsw() back to the control interpretation, and fixing
snd_soc_info_volsw_range(). The control interpretation makes more
sense as limiting is typically done from the machine driver, so it is
appropriate to use the customer facing representation rather than the
internal codec representation. Update all the code to consistently use
this interpretation of platform_max.

Finally, also add some comments to the soc_mixer_control struct to
hopefully avoid further patches switching between the two approaches.

Fixes: fb9ad24485087 ("ASoC: ops: add correct range check for limiting volume")
Signed-off-by: Charles Keepax &lt;ckeepax@opensource.cirrus.com&gt;
Link: https://patch.msgid.link/20250228151456.3703342-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: cs35l56: Prevent races when soft-resetting using SPI control</title>
<updated>2025-02-25T15:06:58Z</updated>
<author>
<name>Richard Fitzgerald</name>
<email>rf@opensource.cirrus.com</email>
</author>
<published>2025-02-25T13:18:43Z</published>
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<id>urn:sha1:769c1b79295c38d60fde4c0a8f5f31e01360c54f</id>
<content type='text'>
When SPI is used for control, the driver must hold the SPI bus lock
while issuing the sequence of writes to perform a soft reset.

&gt;From the time the driver writes the SYSTEM_RESET command until the
driver does a write to terminate the reset, there must not be any
activity on the SPI bus lines. If there is any SPI activity during the
soft-reset, another soft-reset will be triggered. The state of the SPI
chip select is irrelevant.

A repeated soft-reset does not in itself cause any problems, and it is
not an infinite loop. The problem is a race between these resets and
the driver polling for boot completion. There is a time window between
soft resets where the driver could read HALO_STATE as 2 (fully booted)
while the chip is actually soft-resetting. Although this window is
small, it is long enough that it is possible to hit it in normal
operation.

To prevent this race and ensure the chip really is fully booted, the
driver calls spi_bus_lock() to prevent other activity while resetting.
It then issues the SYSTEM_RESET mailbox command. After allowing
sufficient time for reset to take effect, the driver issues a PING
mailbox command, which will force completion of the full soft-reset
sequence. The SPI bus lock can then be released. The mailbox is
checked for any boot or wakeup response from the firmware, before the
value in HALO_STATE will be trusted.

This does not affect SoundWire or I2C control.

Fixes: 8a731fd37f8b ("ASoC: cs35l56: Move utility functions to shared file")
Signed-off-by: Richard Fitzgerald &lt;rf@opensource.cirrus.com&gt;
Link: https://patch.msgid.link/20250225131843.113752-3-rf@opensource.cirrus.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>Merge tag 'sound-6.14-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound</title>
<updated>2025-01-24T15:54:34Z</updated>
<author>
<name>Linus Torvalds</name>
<email>torvalds@linux-foundation.org</email>
</author>
<published>2025-01-24T15:54:34Z</published>
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<id>urn:sha1:2c8d2a510c15c003749e43ac2b8e1bc79a7a00d6</id>
<content type='text'>
Pull sound updates from Takashi Iwai:
 "This was a relatively calm cycle, and most of changes are rather small
  device-specific fixes. Here are highlights:

  Core:
   - Further enhancements of ALSA rawmidi and sequencer APIs for MIDI
     2.0
   - compress-offload API extensions for ASRC support

  ASoC:
   - Allow clocking on each DAI in an audio graph card to be configured
     separately
   - Improved power management for Renesas RZ-SSI
   - KUnit testing for the Cirrus DSP framework
   - Memory to meory operation support for Freescale/NXP platforms
   - Support for pause operations in SOF
   - Support for Allwinner suinv F1C100s, Awinc AW88083, Realtek
     ALC5682I-VE

  HD- and USB-audio:
   - Add support for Focusrite Scarlett 4th Gen 16i16, 18i16, and 18i20
     interfaces via new FCP driver
   - TAS2781 SPI HD-audio sub-codec support
   - Various device-specific quirks as usual"

* tag 'sound-6.14-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (235 commits)
  ALSA: hda: tas2781-spi: Fix bogus error handling in tas2781_hda_spi_probe()
  ALSA: hda: tas2781-spi: Fix error code in tas2781_read_acpi()
  ALSA: hda: tas2781-spi: Delete some dead code
  ALSA: usb: fcp: Fix return code from poll ops
  ALSA: usb: fcp: Fix incorrect resp-&gt;opcode retrieval
  ALSA: usb: fcp: Fix meter_levels type to __le32
  ALSA: hda/realtek: Enable Mute LED on HP Laptop 14s-fq1xxx
  ALSA: hda: tas2781-spi: Fix -Wsometimes-uninitialized in tasdevice_spi_switch_book()
  ALSA: ctxfi: Simplify dao_clear_{left,right}_input() functions
  ALSA: hda: tas2781-spi: select CRC32 instead of CRC32_SARWATE
  ALSA: usb: fcp: Fix hwdep read ops types
  ALSA: scarlett2: Add device_setup option to use FCP driver
  ALSA: FCP: Add Focusrite Control Protocol driver
  ALSA: hda/tas2781: Add tas2781 hda SPI driver
  ALSA: hda/realtek - Fixed headphone distorted sound on Acer Aspire A115-31 laptop
  ASoC: xilinx: xlnx_spdif: Simpify using devm_clk_get_enabled()
  ALSA: hda: Support for Ideapad hotkey mute LEDs
  ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83JX, 83MC and 83NM
  ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83LC
  ASoC: dapm: add support for preparing streams
  ...
</content>
</entry>
<entry>
<title>Merge tag 'regmap-v6.14' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap</title>
<updated>2025-01-22T16:57:52Z</updated>
<author>
<name>Linus Torvalds</name>
<email>torvalds@linux-foundation.org</email>
</author>
<published>2025-01-22T16:57:52Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=6f10810ccd6de53ff158d3b16013591c8d7442b3'/>
<id>urn:sha1:6f10810ccd6de53ff158d3b16013591c8d7442b3</id>
<content type='text'>
Pull regmap updates from Mark Brown:
 "There's one big bit of work this time around, the addition of support
  for a greater range of MBQ access sizes to SoundWire devices together
  with support for deferred read/write.

  The MBQ register maps generally have variable register sizes, the
  variable regiseter size support allows them to be handled much more
  naturally within regmap with less open coding in drivers.

  The deferred read/write support avoids spurious errors when devices
  make use of a bus feature allowing them to indicate they're busy.
  These changes pull in a supporting SoundWire change, and there's an
  ASoC change building off the new code.

  The remainder of the changes are code cleanups"

* tag 'regmap-v6.14' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regmap:
  regmap: sdw-mbq: Add support for SDCA deferred controls
  regmap: sdw-mbq: Add support for further MBQ register sizes
  ASoC: SDCA: Update list of entity_0 controls
  soundwire: SDCA: Add additional SDCA address macros
  regmap: regmap_multi_reg_read(): make register list const
  regmap: cache: rbtree: use krealloc_array() to replace krealloc()
  regmap: cache: mapple: use kmalloc_array() to replace kmalloc()
  regmap: place foo / 8 and foo % 8 closer to each other
  regmap: Use BITS_TO_BYTES()
  regmap: cache: Use BITS_TO_BYTES()
</content>
</entry>
<entry>
<title>Merge tag 'asoc-v6.14' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus</title>
<updated>2025-01-20T15:15:07Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2025-01-20T15:15:07Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=8514d8f80e5cc87eb68ec037857fcd64d9c0bc68'/>
<id>urn:sha1:8514d8f80e5cc87eb68ec037857fcd64d9c0bc68</id>
<content type='text'>
ASoC: Updates for v6.14

This was quite a quiet release for what I imagine are holiday related
reasons, the diffstat is dominated by some Cirrus Logic Kunit tests.
There's the usual mix of small improvements and fixes, plus a few new
drivers and features.  The diffstat includes some DRM changes due to
work on HDMI audio.

 - Allow clocking on each DAI in an audio graph card to be configured
   separately.
 - Improved power management for Renesas RZ-SSI.
 - KUnit testing for the Cirrus DSP framework.
 - Memory to meory operation support for Freescale/NXP platforms.
 - Support for pause operations in SOF.
 - Support for Allwinner suinv F1C100s, Awinc AW88083, Realtek
   ALC5682I-VE
</content>
</entry>
<entry>
<title>Merge branch 'for-linus' into for-next</title>
<updated>2025-01-20T08:25:33Z</updated>
<author>
<name>Takashi Iwai</name>
<email>tiwai@suse.de</email>
</author>
<published>2025-01-20T08:25:33Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=7bba999e939f6d20e4db66effeecc179ed896496'/>
<id>urn:sha1:7bba999e939f6d20e4db66effeecc179ed896496</id>
<content type='text'>
Pull pending ASoC and HD-audio fixes for 6.14-rc1

Signed-off-by: Takashi Iwai &lt;tiwai@suse.de&gt;
</content>
</entry>
<entry>
<title>ASoC: soc-dai: add snd_soc_dai_prepare() and use it internally</title>
<updated>2025-01-15T13:11:18Z</updated>
<author>
<name>Martin Blumenstingl</name>
<email>martin.blumenstingl@googlemail.com</email>
</author>
<published>2025-01-14T21:56:16Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=d4e91adfc261de1454538ee6437e6014a2d1fca1'/>
<id>urn:sha1:d4e91adfc261de1454538ee6437e6014a2d1fca1</id>
<content type='text'>
Add a new snd_soc_dai_prepare() which can be used (in an upcoming patch)
by soc-dapm.c. Use this new function internally in
snd_soc_pcm_dai_prepare() to avoid duplicating code.

Suggested-by: Jerome Brunet &lt;jbrunet@baylibre.com&gt;
Reviewed-by: Charles Keepax &lt;ckeepax@opensource.cirrus.com&gt;
Reviewed-by: Jerome Brunet &lt;jbrunet@baylibre.com&gt;
Signed-off-by: Martin Blumenstingl &lt;martin.blumenstingl@googlemail.com&gt;
Acked-by: Kuninori Morimoto &lt;kuninori.morimoto.gx@renesas.com&gt;
Link: https://patch.msgid.link/20250114215617.336105-2-martin.blumenstingl@googlemail.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
<entry>
<title>ASoC: simple-card-utils: fix priv-&gt;dai_props indexing</title>
<updated>2025-01-15T13:11:12Z</updated>
<author>
<name>Laurentiu Mihalcea</name>
<email>laurentiu.mihalcea@nxp.com</email>
</author>
<published>2025-01-14T18:43:14Z</published>
<link rel='alternate' type='text/html' href='https://git.stealer.net/cgit.cgi/user/sven/linux.git/commit/?id=04e97fa7dd7e3eda754712f92df2136acd1d9088'/>
<id>urn:sha1:04e97fa7dd7e3eda754712f92df2136acd1d9088</id>
<content type='text'>
As of commit cb18cd26039f ("ASoC: soc-core: do rtd-&gt;id trick at
snd_soc_add_pcm_runtime()") the ID stored in the PCM runtime data can
no longer be safely used to index the priv-&gt;dai_props array. This is
because the ID may be modified during snd_soc_add_pcm_runtime(), thus
resulting in an ID that's no longer a valid array index.

To fix this, use the position of the dai_link stored inside the PCM
runtime data relative to the start of the dai_link array as index into
the priv-&gt;dai_props array.

Signed-off-by: Laurentiu Mihalcea &lt;laurentiu.mihalcea@nxp.com&gt;
Acked-by: Kuninori Morimoto &lt;kuninori.morimoto.gx@renesas.com&gt;
Link: https://patch.msgid.link/20250114184314.3583-2-laurentiumihalcea111@gmail.com
Signed-off-by: Mark Brown &lt;broonie@kernel.org&gt;
</content>
</entry>
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