In data blocks of common isochronous packet for MOTU devices, PCM
frames are multiplexed in a shape of '24 bit * 4 Audio Pack', described
in IEC 61883-6. The frames are not aligned to quadlet.
For capture PCM substream, ALSA firewire-motu driver constructs PCM
frames by reading data blocks byte-by-byte. However this operation
includes bug for lower byte of the PCM sample. This brings invalid
content of the PCM samples.
This commit fixes the bug.
Reported-by: Peter Sjöberg <autopeter@gmail.com>
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 4641c93940 ("ALSA: firewire-motu: add MOTU specific protocol layer")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I set 10 seconds for the timeout of the i915 audio component binding
with a hope that recent machines are fast enough to handle all probe
tasks in that period, but I was too optimistic. The binding may take
longer than that, and this caused a problem on the machine with both
audio and graphics driver modules loaded in parallel, as Paul Menzel
experienced. This problem haven't hit so often just because the KMS
driver is loaded in initrd on most machines.
As a simple workaround, extend the timeout to 60 seconds.
Fixes: f9b54e1961 ("ALSA: hda/i915: Allow delayed i915 audio component binding")
Reported-by: Paul Menzel <pmenzel+alsa-devel@molgen.mpg.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
A platform can have multiple sound cards for different audio paths.
Following is the print seen duirng device boot for jetson-xavier,
ALSA device list:
#0: nvidia,p2972-0000 at 0x3518000 irq 17
By looking at above, it is not very clear if the sound card is for
HDA. It becomes confusing when platform has registered multiple cards,
and platform model name is used for card.
This patch uses "nvidia,model" property mentioned in hda device tree
to get the card name. Since property is optional, legacy boards will
continue to use "tegra-hda". Custom name can be passed wherever needed.
This naming convention is conistent with the way sound cards are named
in general.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"nvidia,model" property is added to pass custom name for hda sound card.
This is parsed in hda driver and used for card name. This aligns with the
way with which sound cards are named in general.
This patch populates above for jetson-tx1, jetson-tx2 and jetson-xavier.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
An optional property "nvidia,model" is introduced for hda to pass custom
name for the sound card. The suffix "-hda" in the name passed is useful
to distinguish between multiple cards available for a platform.
When the property is not specified, default name("tegra-hda") mentioned
in hda driver is used. This property can be added in platform specific
file of the board and card name can relate to the board in use.
Signed-off-by: Sameer Pujar <spujar@nvidia.com>
Reviewed-by: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The ASUS UX362FA with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone. This issue can be fixed
by the quirk in the commit 4e0511067 ALSA: hda/realtek: Enable audio
jacks of ASUS UX533FD with ALC294.
Besides, ASUS UX362FA and UX533FD have the same audio initial pin config
values. So, this patch replaces SND_PCI_QUIRK of UX533FD with a new
SND_HDA_PIN_QUIRK which benefits both UX362FA and UX533FD.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Ming Shuo Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Dell Precision 5820 with ALC3234 codec (which is equivalent with
ALC255) shows click noises at (runtime) PM resume on the headphone.
The biggest source of the noise comes from the cleared headphone pin
control at resume, which is done via the standard shutup procedure.
Although we have an override of the standard shutup callback to
replace with NOP, this would skip other needed stuff (e.g. the pull
down of headset power). So, instead, this "fixes" the behavior of
alc_fixup_no_shutup() by introducing spec->no_shutup_pins flag.
When this flag is set, Realtek codec won't call the standard
snd_hda_shutup_pins() & co. Now alc_fixup_no_shutup() just sets this
flag instead of overriding spec->shutup callback itself. This allows
us to apply the similar fix for other entries easily if needed in
future.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Forgot to update the document.
Fixes: e854747d75 ("ALSA: hda/realtek - Enable headset button support for new codec")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some garbage was taken via copy-and-paste error. Clean up.
Fixes: a26d96c780 ("ALSA: hda/realtek - Comprehensive model list for ALC259 & co")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We forgot to unreference the node when aborting from the loop of
for_each_child_of_node() in snd_pmac_tumbler_init(). This leads to
unbalanced node refcount. Fix it by adding the missing of_node_put()
call.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We forgot to unreference a node obtained via of_find_node_by_name()
after its usage.
Reviewed-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ac97_of_get_child_device() take the refcount of the node explicitly
via of_node_get(), but this leads to an unbalance. The
for_each_child_of_node() loop itself takes the refcount for each
iteration node, hence you don't need to take the extra refcount
again.
Fixes: 2225a3e6af ("ALSA: ac97: add codecs devicetree binding")
Reviewed-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At least some USB devices use (MSB-aligned) audio format larger
than the actual resolution of the device. In order to expose the
actual device resolution (bBitResolution), add extra field to the
procfs stream info interface.
Signed-off-by: Jussi Laako <jussi@sonarnerd.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Fixes gcc '-Wunused-but-set-variable' warning:
sound/isa/es1688/es1688_lib.c: In function 'snd_es1688_probe':
sound/isa/es1688/es1688_lib.c:124:31: warning:
variable 'hw' set but not used [-Wunused-but-set-variable]
unsigned short major, minor, hw;
^
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This avoids bringing back the problem introduced by
62ba568f7a ("ALSA: pcm: Return 0 when size <
start_threshold in capture") and fixed in 00a399cad1
("ALSA: pcm: Revert capture stream behavior change in
blocking mode"), which prevented the user from starting
capture from another thread.
Signed-off-by: Ricardo Biehl Pasquali <pasqualirb@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BE dai links only have internal PCM's and their substream ops may
not be set. Suspending these PCM's will result in their
ops->trigger() being invoked and cause a kernel oops.
So skip suspending PCM's if their ops are NULL.
[ NOTE: this change is required now for following the recent PCM core
change to get rid of snd_pcm_suspend() call. Since DPCM BE takes
the runtime carried from FE while keeping NULL ops, it can hit this
bug. See details at:
https://github.com/thesofproject/linux/pull/582
-- tiwai ]
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In the commit 62ba568f7a ("ALSA: pcm: Return 0 when size <
start_threshold in capture"), we changed the behavior of
__snd_pcm_lib_xfer() to return immediately with 0 when a capture
stream has a high start_threshold. This was intended to be a
correction of the behavior consistency and looked harmless, but this
was the culprit of the recent breakage reported by syzkaller, which
was fixed by the commit e190161f96 ("ALSA: pcm: Fix tight loop of
OSS capture stream").
At the time for the OSS fix, I didn't touch the behavior for ALSA
native API, as assuming that this behavior actually is good. But this
turned out to be also broken actually for a similar deployment,
e.g. one thread goes to a write loop in blocking mode while another
thread controls the start/stop of the stream manually.
Overall, the original commit is harmful, and it brings less merit to
keep that behavior. Let's revert it.
Fixes: 62ba568f7a ("ALSA: pcm: Return 0 when size < start_threshold in capture")
Fixes: e190161f96 ("ALSA: pcm: Fix tight loop of OSS capture stream")
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Now all callers no longer check the return value from
snd_pcm_lib_preallocate_pages() and co, let's make them to return
void, so that any new code won't fall into the same pitfall.
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Lots and lots of new drivers so far, a highlight being the MediaTek
BTCVSD which is a driver for a Bluetooth radio chip - the first such
driver we've had upstream. Hopefully we will soon also see a baseband
with an upstream driver!
- Support for only powering up channels that are actively being used.
- Quite a few improvements to simplify the generic card drivers,
especially the merge of the SCU cards into the main generic drivers.
- Lots of fixes for probing on Intel systems, trying to rationalize
things to look more standard from a framework point of view.
- New drivers for Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341,
Google ChromeOS embedded controllers, Ingenic JZ4725B, MediaTek
BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328, Spreadtrum
DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM formatters.
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Merge tag 'asoc-v5.1' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v5.1
Lots and lots of new drivers so far, a highlight being the MediaTek
BTCVSD which is a driver for a Bluetooth radio chip - the first such
driver we've had upstream. Hopefully we will soon also see a baseband
with an upstream driver!
- Support for only powering up channels that are actively being used.
- Quite a few improvements to simplify the generic card drivers,
especially the merge of the SCU cards into the main generic drivers.
- Lots of fixes for probing on Intel systems, trying to rationalize
things to look more standard from a framework point of view.
- New drivers for Asahi Kasei Microdevices AK4497, Cirrus Logic CS4341,
Google ChromeOS embedded controllers, Ingenic JZ4725B, MediaTek
BTCVSD, MT8183 and MT6358, NXP MICFIL, Rockchip RK3328, Spreadtrum
DMA controllers, Qualcomm WCD9335, Xilinx S/PDIF and PCM formatters.
A selection of driver specific fixes here, along with a few core fixes:
- A fixup for some MFD devices that were broken by the previous fixes
for deferred probe.
- A fix for potential out of bounds array accesses when ordering DAPM
power/up down sequences.
- Avoid use after free issue when unloading and reloading drivers using
topologies.
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Merge tag 'asoc-fix-v5.0-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
A selection of driver specific fixes here, along with a few core fixes:
- A fixup for some MFD devices that were broken by the previous fixes
for deferred probe.
- A fix for potential out of bounds array accesses when ordering DAPM
power/up down sequences.
- Avoid use after free issue when unloading and reloading drivers using
topologies.
The commit a60945fd08 ("ALSA: usb-audio: move implicit fb quirks to
separate function") introduced an error in the handling of quirks for
implicit feedback endpoints. This commit fixes this.
If a quirk successfully sets up an implicit feedback endpoint, usb-audio
no longer tries to find the implicit fb endpoint itself.
Fixes: a60945fd08 ("ALSA: usb-audio: move implicit fb quirks to separate function")
Signed-off-by: Manuel Reinhardt <manuel.rhdt@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This enables mute LED support and fixes switching jacks when the laptop
is docked.
Signed-off-by: Jurica Vukadin <jurica.vukadin@rt-rk.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change the header comment to use C++ style, so that it looks more
consistent with the rest of ASoC.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Show the knob to enable or disable the jz4740-codec driver, add a
proper description, and add a dependency on MIPS || COMPILE_TEST, as
this driver is only useful on MIPS.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add license information as a standard SPDX license notifier instead of
custom text.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add documentation about how to probe the jz4725b-codec driver from
devicetree.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add documentation about how to probe the jz4740-codec driver from
devicetree.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Line Playback Volume for Allwinner A10 and Allwinner A20.
Add Line Boost Volume for Allwinner A10 and Allwinner A20.
Add Line Right, Line Left, Line Playback Switch for Allwinner A10 and
Allwinner A20.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add FM Playback Volume for Allwinner A10 and Allwinner A20.
Add FM Left, FM Right, FM Playback Switch for Allwinner A10 and
Allwinner A20.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Mic1 Playback Switch and Mic2 Playback Switch for Allwinner A10 and
Allwinner A20.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Since it's now possible to have a DAPM mixer control with multiple
channels, use it to cut down the total number of controls.
Keep "Left Mixer Left DAC Playback Switch" and "Right Mixer Right DAC
Playback Switch" name & layout the same as before for compatibility.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add Mic1 Boost Volume and Mic2 Boost Volume for Allwinner A10 and for
Allwinner A20.
Those controls are in different registers per chip model, so put the
Allwinner A10 controls and the Allwinner A20 controls into the newly
split sun4i_codec_controls and sun7i_codec_controls, respectively.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Introduce sun7i_codec_controls because some of the controls are different
on Allwinner A20 compared to Allwinner A10.
Also introduce sun7i_codec_codec in order to use sun7i_codec_controls and
make sun7i_codec_quirks use sun7i_codec_codec.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add a control "Mic Playback Volume" that allows the user to control the
MIC gain stage (common for Mic1 and Mic2) leading to the output mixer.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add MIC2 Pre-Amplifier, Mic2 input for Allwinner A10 and Allwinner A20.
Previously, there only the Mic1 input and MIC1 Pre-Amplifier was exposed.
This exposes the Mic2 input and MIC2 Pre-Amplifier.
Signed-off-by: Danny Milosavljevic <dannym@scratchpost.org>
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
There is a spelling mistake in the SOC_SINGLE control name. Fix this.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Currently "0xf << 36" is used to
clear SSIU-9 internal buffer state, which overflows 32-bit value
according to user reference manual, it is always bit4 ~ bit7
of SSI_SYS_STATUS[1,3,5,7] registers indicate
SSIU-9's buffer state, so "0xf << 4" should be used.
This patch fix incorrect shifting issue in SSIU-9 case
Fixes: commit b7169ddea2 ("ASoC: rsnd: remove RSND_REG_ from rsnd_reg")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Add jz4725b-codec driver to support the internal CODEC found in the
JZ4725B SoC from Ingenic.
Signed-off-by: Paul Cercueil <paul@crapouillou.net>
Signed-off-by: Mark Brown <broonie@kernel.org>
commit 4d230d1271 ("ASoC: rsnd: fixup not to call clk_get/set
under non-atomic") added new rsnd_ssi_prepare() and moved
rsnd_ssi_master_clk_start() to .prepare.
But, ssi user count (= ssi->usrcnt) is incremented at .init
(= rsnd_ssi_init()).
Because of these timing exchange, ssi->usrcnt check at
rsnd_ssi_master_clk_start() should be adjusted.
Otherwise, 2nd master clock setup will be no check.
This patch fixup this issue.
Fixes: commit 4d230d1271 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic")
Reported-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Reported-by: Valentine Barshak <valentine.barshak@cogentembedded.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
snd_pcm_lib_preallocate_pages() and co always succeed, so the error
check is simply redundant. Drop it.
Acked-by: Ezequiel Garcia <ezequiel@collabora.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To detect potential errors, let's add:
a) build-time warnings when the table size isn't aligned with the enum
list
b) run-time warnings when the values are not initialized. This
requires an increase by one of all values to avoid the default 0.
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
