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commit c447374494 upstream.
The Realtek PC Beep Hidden Register[1] is currently set by
patch_realtek.c in two different places:
In alc_fill_eapd_coef(), it's set to the value 0x5757, corresponding to
non-beep input on 1Ah and no 1Ah loopback to either headphones or
speakers. (Although, curiously, the loopback amp is still enabled.) This
write was added fairly recently by commit e3743f4311 ("ALSA:
hda/realtek - Dell headphone has noise on unmute for ALC236") and is a
safe default. However, it happens in the wrong place:
alc_fill_eapd_coef() runs on module load and cold boot but not on S3
resume, meaning the register loses its value after suspend.
Conversely, in alc256_init(), the register is updated to unset bit 13
(disable speaker loopback) and set bit 5 (set non-beep input on 1Ah).
Although this write does run on S3 resume, it's not quite enough to fix
up the register's default value of 0x3717. What's missing is a set of
bit 14 to disable headphone loopback. Without that, we end up with a
feedback loop where the headphone jack is being driven by amplified
samples of itself[2].
This change eliminates the update in alc256_init() and replaces it with
the 0x5757 write from alc_fill_eapd_coef(). Kailang says that 0x5757 is
supposed to be the codec's default value, so using it will make
debugging easier for Realtek.
Affects the ALC255, ALC256, ALC257, ALC235, and ALC236 codecs.
[1] Newly documented in Documentation/sound/hd-audio/realtek-pc-beep.rst
[2] Setting the "Headphone Mic Boost" control from userspace changes
this feedback loop and has been a widely-shared workaround for headphone
noise on laptops like the Dell XPS 13 9350. This commit eliminates the
feedback loop and makes the workaround unnecessary.
Fixes: e1e8c1fdce ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/bf22b417d1f2474b12011c2a39ed6cf8b06d3bf5.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ae769d3556 upstream.
The recent fix for the OOB access in PCM OSS plugins (commit
f2ecf903ef: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a
regression on OSS applications. The patch introduced the size check
in client and slave size calculations to limit to each plugin's buffer
size, but I overlooked that some code paths call those without
allocating the buffer but just for estimation.
This patch fixes the bug by skipping the size check for those code
paths while keeping checking in the actual transfer calls.
Fixes: f2ecf903ef ("ALSA: pcm: oss: Avoid plugin buffer overflow")
Tested-and-reported-by: Jari Ruusu <jari.ruusu@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200403072515.25539-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3c6fd1f07e upstream.
The recent AMD platform exposes an HD-audio bus but without any actual
codecs, which is internally tied with a USB-audio device, supposedly.
It results in "no codecs" error of HD-audio bus driver, and it's
nothing but a waste of resources.
This patch introduces a static blacklist table for skipping such a
known bogus PCI SSID entry. As of writing this patch, the known SSIDs
are:
* 1043:874f - ASUS ROG Zenith II / Strix
* 1462:cb59 - MSI TRX40 Creator
* 1462:cb60 - MSI TRX40
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2a48218f8e upstream.
Some recent boards (supposedly with a new AMD platform) contain the
USB audio class 2 device that is often tied with HD-audio. The device
exposes an Input Gain Pad control (id=19, control=12) but this node
doesn't behave correctly, returning an error for each inquiry of
GET_MIN and GET_MAX that should have been mandatory.
As a workaround, simply ignore this node by adding a usbmix_name_map
table entry. The currently known devices are:
* 0414:a002 - Gigabyte TRX40 Aorus Pro WiFi
* 0b05:1916 - ASUS ROG Zenith II
* 0b05:1917 - ASUS ROG Strix
* 0db0:0d64 - MSI TRX40 Creator
* 0db0:543d - MSI TRX40
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 21fca8bdbb upstream.
soc_compr_trigger_fe() allows start or stop after pause_push.
In dpcm_be_dai_trigger(), however, only pause_release is allowed
command after pause_push.
So, start or stop after pause in compress offload is always
returned as error if the compress offload is used with dpcm.
To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed
for start or stop command.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3bbbb7728f upstream.
Since a virtual mixer has no backing registers
to decide which path to connect,
it will try to match with initial state.
This is to ensure that the default mixer choice will be
correctly powered up during initialization.
Invert flag is used to select initial state of the virtual switch.
Since actual hardware can't be disconnected by virtual switch,
connected is better choice as initial state in many cases.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/01a301d60731$b724ea10$256ebe30$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f2ecf903ef upstream.
Each OSS PCM plugins allocate its internal buffer per pre-calculation
of the max buffer size through the chain of plugins (calling
src_frames and dst_frames callbacks). This works for most plugins,
but the rate plugin might behave incorrectly. The calculation in the
rate plugin involves with the fractional position, i.e. it may vary
depending on the input position. Since the buffer size
pre-calculation is always done with the offset zero, it may return a
shorter size than it might be; this may result in the out-of-bound
access as spotted by fuzzer.
This patch addresses those possible buffer overflow accesses by simply
setting the upper limit per the given buffer size for each plugin
before src_frames() and after dst_frames() calls.
Reported-by: syzbot+e1fe9f44fb8ecf4fb5dd@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/000000000000b25ea005a02bcf21@google.com
Link: https://lore.kernel.org/r/20200309082148.19855-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 242c46c023 upstream.
In case of ABI version mismatch, _manifest needs to be freed as
it is just a copy of the original topology manifest. However, if
a driver manifest handler is defined, that would get executed and
the cleanup is never reached. Fix that by getting the return status
of manifest() instead of returning directly.
Fixes: 583958fa2e ("ASoC: topology: Make manifest backward compatible from ABI v4")
Signed-off-by: Dragos Tarcatu <dragos_tarcatu@mentor.com>
Link: https://lore.kernel.org/r/20200207185325.22320-3-dragos_tarcatu@mentor.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9b3193089e upstream.
commit c2caa4da46 ("ASoC: Fix widget powerdown on shutdown") added a
set of the power state during snd_soc_dapm_shutdown to ensure the
widgets powered off. However, when commit 39eb5fd13d
("ASoC: dapm: Delay w->power update until the changes are written")
added the new_power member of the widget structure, to differentiate
between the current power state and the target power state, it did not
update the shutdown to use the new_power member.
As new_power has not updated it will be left in the state set by the
last DAPM sequence, ie. 1 for active widgets. So as the DAPM sequence
for the shutdown proceeds it will turn the widgets on (despite them
already being on) rather than turning them off.
Fixes: 39eb5fd13d ("ASoC: dapm: Delay w->power update until the changes are written")
Signed-off-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200228153145.21013-1-ckeepax@opensource.cirrus.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6c89ffea60 upstream.
dpcm_show_state() invokes multiple snprintf() calls to concatenate
formatted strings on the fixed size buffer. The usage of snprintf()
is supposed for avoiding the buffer overflow, but it doesn't work as
expected because snprintf() doesn't return the actual output size but
the size to be written.
Fix this bug by replacing all snprintf() calls with scnprintf()
calls.
Fixes: f86dcef87b ("ASoC: dpcm: Add debugFS support for DPCM")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200218111737.14193-4-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 549cd0ba04 upstream.
The debugfs output of intel skl driver writes strings with multiple
snprintf() calls with the fixed size. This was supposed to avoid the
buffer overflow but actually it still would, because snprintf()
returns the expected size to be output, not the actual output size.
Fix it by replacing snprintf() calls with scnprintf().
Fixes: d14700a01f ("ASoC: Intel: Skylake: Debugfs facility to dump module config")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Acked-by: Cezary Rojewski <cezary.rojewski@intel.com>
Link: https://lore.kernel.org/r/20200218111737.14193-3-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit dc7497795e upstream.
snd_seq_check_queue() passes the current tick and time of the given
queue as a pointer to snd_seq_prioq_cell_out(), but those might be
updated concurrently by the seq timer update.
Fix it by retrieving the current tick and time via the proper helper
functions at first, and pass those values to snd_seq_prioq_cell_out()
later in the loops.
snd_seq_timer_get_cur_time() takes a new argument and adjusts with the
current system time only when it's requested so; this update isn't
needed for snd_seq_check_queue(), as it's called either from the
interrupt handler or right after queuing.
Also, snd_seq_timer_get_cur_tick() is changed to read the value in the
spinlock for the concurrency, too.
Reported-by: syzbot+fd5e0eaa1a32999173b2@syzkaller.appspotmail.com
Link: https://lore.kernel.org/r/20200214111316.26939-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 8fea78029f ]
If CONFIG_SND_ATMEL_SOC_DMA=m, build error:
sound/soc/atmel/atmel_ssc_dai.o: In function `atmel_ssc_set_audio':
(.text+0x7cd): undefined reference to `atmel_pcm_dma_platform_register'
Function atmel_pcm_dma_platform_register is defined under
CONFIG SND_ATMEL_SOC_DMA, so select SND_ATMEL_SOC_DMA in
CONFIG SND_ATMEL_SOC_SSC, same to CONFIG_SND_ATMEL_SOC_PDC.
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Chen Zhou <chenzhou10@huawei.com>
Link: https://lore.kernel.org/r/20200113133242.144550-1-chenzhou10@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f1dd4795b1 ]
A long-standing compile warning was seen during build test:
sound/sh/aica.c: In function 'load_aica_firmware':
sound/sh/aica.c:521:25: warning: passing argument 2 of 'spu_memload' discards 'const' qualifier from pointer target type [-Wdiscarded-qualifiers]
Fixes: 198de43d75 ("[ALSA] Add ALSA support for the SEGA Dreamcast PCM device")
Link: https://lore.kernel.org/r/20200105144823.29547-69-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5da116f164 ]
Remove unused variables that are left over after the conversion of new
PCM ops:
sound/sh/sh_dac_audio.c:166:26: warning: unused variable 'runtime'
sound/sh/sh_dac_audio.c:186:26: warning: unused variable 'runtime'
sound/sh/sh_dac_audio.c:205:26: warning: unused variable 'runtime'
Fixes: 1cc2f8ba0b ("ALSA: sh: Convert to the new PCM ops")
Link: https://lore.kernel.org/r/20200104110057.13875-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit df4654bd6e ]
Clang warns:
../sound/usb/usx2y/usX2Yhwdep.c:122:3: warning: misleading indentation;
statement is not part of the previous 'if' [-Wmisleading-indentation]
info->version = USX2Y_DRIVER_VERSION;
^
../sound/usb/usx2y/usX2Yhwdep.c:120:2: note: previous statement is here
if (us428->chip_status & USX2Y_STAT_CHIP_INIT)
^
1 warning generated.
This warning occurs because there is a space before the tab on this
line. Remove it so that the indentation is consistent with the Linux
kernel coding style and clang no longer warns.
This was introduced before the beginning of git history so no fixes tag.
Link: https://github.com/ClangBuiltLinux/linux/issues/831
Signed-off-by: Nathan Chancellor <natechancellor@gmail.com>
Link: https://lore.kernel.org/r/20191218034257.54535-1-natechancellor@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d61fe22c2a ]
A design of ALSA control core allows applications to execute three
operations for TLV feature; read, write and command. Furthermore, it
allows driver developers to process the operations by two ways; allocated
array or callback function. In the former, read operation is just allowed,
thus developers uses the latter when device driver supports variety of
models or the target model is expected to dynamically change information
stored in TLV container.
The core also allows applications to lock any element so that the other
applications can't perform write operation to the element for element
value and TLV information. When the element is locked, write and command
operation for TLV information are prohibited as well as element value.
Any read operation should be allowed in the case.
At present, when an element has callback function for TLV information,
TLV read operation returns EPERM if the element is locked. On the
other hand, the read operation is success when an element has allocated
array for TLV information. In both cases, read operation is success for
element value expectedly.
This commit fixes the bug. This change can be backported to v4.14
kernel or later.
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20191223093347.15279-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 93f9d1a4ac upstream.
The Audioengine D1 (0x2912:0x30c8) does support reading the sample rate,
but it returns the rate in byte-reversed order.
When setting sampling rate, the driver produces these warning messages:
[168840.944226] usb 3-2.2: current rate 4500480 is different from the runtime rate 44100
[168854.930414] usb 3-2.2: current rate 8436480 is different from the runtime rate 48000
[168905.185825] usb 3-2.1.2: current rate 30465 is different from the runtime rate 96000
As can be seen from the hexadecimal conversion, the current rate read
back is byte-reversed from the rate that was set.
44100 == 0x00ac44, 4500480 == 0x44ac00
48000 == 0x00bb80, 8436480 == 0x80bb00
96000 == 0x017700, 30465 == 0x007701
Rather than implementing a new quirk to reverse the order, just skip
checking the rate to avoid spamming the log.
Signed-off-by: Arvind Sankar <nivedita@alum.mit.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200211162235.1639889-1-nivedita@alum.mit.edu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 44eeb081b8 upstream.
Some code in HD-audio driver calls snprintf() in a loop and still
expects that the return value were actually written size, while
snprintf() returns the expected would-be length instead. When the
given buffer limit were small, this leads to a buffer overflow.
Use scnprintf() for addressing those issues. It returns the actually
written size unlike snprintf().
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200218091409.27162-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit acbf27746e ]
Currently, the trigger orders SND_SOC_DPCM_TRIGGER_PRE/POST
determine the order in which FE DAI and BE DAI are triggered.
In the case of SND_SOC_DPCM_TRIGGER_PRE, the FE DAI is
triggered before the BE DAI and in the case of
SND_SOC_DPCM_TRIGGER_POST, the BE DAI is triggered before
the FE DAI. And this order remains the same irrespective of the
trigger command.
In the case of the SOF driver, during playback, the FW
expects the BE DAI to be triggered before the FE DAI during
the START trigger. The BE DAI trigger handles the starting of
Link DMA and so it must be started before the FE DAI is started
to prevent xruns during pause/release. This can be addressed
by setting the trigger order for the FE dai link to
SND_SOC_DPCM_TRIGGER_POST. But during the STOP trigger,
the FW expects the FE DAI to be triggered before the BE DAI.
Retaining the same order during the START and STOP commands,
results in FW error as the DAI component in the FW is still
active.
The issue can be fixed by mirroring the trigger order of
FE and BE DAI's during the START and STOP trigger. So, with the
trigger order set to SND_SOC_DPCM_TRIGGER_PRE, the FE DAI will be
trigger first during SNDRV_PCM_TRIGGER_START/STOP/RESUME
and the BE DAI will be triggered first during the
STOP/SUSPEND/PAUSE commands. Conversely, with the trigger order
set to SND_SOC_DPCM_TRIGGER_POST, the BE DAI will be triggered
first during the SNDRV_PCM_TRIGGER_START/STOP/RESUME commands
and the FE DAI will be triggered first during the
SNDRV_PCM_TRIGGER_STOP/SUSPEND/PAUSE commands.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191104224812.3393-2-ranjani.sridharan@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f474808acb ]
A lot of places in the driver use onyx_read_register() without
checking the return value, and it's been working OK for ~10 years
or so, so probably never fails ... Rather than trying to check the
return value everywhere, which would be relatively intrusive, at
least make sure we don't use an uninitialized value.
Fixes: f3d9478b2c ("[ALSA] snd-aoa: add snd-aoa")
Reported-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fd14f4436f ]
If multiple serializers are connected in the system and the number of
channels will need to use more than one serializer the mask to enable the
serializers were left to 0 if tdm_mask is provided
Fixes: dd55ff8346 ("ASoC: davinci-mcasp: Add set_tdm_slots() support")
Signed-off-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6a7c59c6d9 ]
A stream may specify a rate range using 'rate_min' and 'rate_max', so a
stream may be valid and not specify any rates. However, as stream cannot
be valid and not have any channel. Let's use this condition instead to
determine if a stream is valid or not.
Fixes: cde79035c6 ("ASoC: Handle multiple codecs with split playback / capture")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 328e9f6973 ]
The error from snd_usb_mixer_apply_create_quirk() is ignored in the
current usb-audio driver code, which will continue the probing even
after the error. Let's take it more serious.
Fixes: 7b1eda223d ("ALSA: usb-mixer: factor out quirks")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8d16672008 ]
The apq8016 driver leaves the of-node refcount at aborting from the
loop of for_each_child_of_node() in the error path. Not only the
iterator node of for_each_child_of_node(), the children nodes referred
from it for codec and cpu have to be properly unreferenced.
Fixes: bdb052e81f ("ASoC: qcom: add apq8016 sound card support")
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c369c8db15 ]
A quirk in snd-usb-audio was added to automate setting sample rate to
4800k and remove the previously exposed nonfunctional microphone for
the Bowers & Wilkins PX:
commit 240a8af929https://lore.kernel.org/patchwork/patch/919689/
However the headphones where updated shortly after that to remove the
unintentional microphone functionality. I guess because of this the
headphones now crash when connecting them via USB while the quirk is
active. Dmesg:
snd-usb-audio: probe of 2-3:1.0 failed with error -22
usb 2-3: 2:1: cannot get min/max values for control 2 (id 2)
This patch removes the microfone and allows the headphones to connect
and work out of the box. It is based on the current mainline kernel
and successfully applied an tested on my machine (4.18.10.arch1-1).
Fixes: 240a8af929 ("ALSA: usb-audio: Add a quirck for B&W PX headphones")
Signed-off-by: Nicolas Huaman <nicolas@herochao.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
