commit 175b8d89fe292796811fdee87fa39799a5b6b87a upstream.
syzbot spotted a potential out-of-bounds shift in the PCM OSS layer
where it calculates the buffer size with the arbitrary shift value
given via an ioctl.
Add a range check for avoiding the undefined behavior.
As the value can be treated by a signed integer, the max shift should
be 30.
Reported-by: syzbot+df7dc146ebdd6435eea3@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201209084552.17109-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c6dde8ffd071aea9d1ce64279178e470977b235c upstream.
The current channel-map control implementation in USB-audio driver may
lead to an error message like
"control 3:0:0:Playback Channel Map:0: access overflow"
when CONFIG_SND_CTL_VALIDATION is set. It's because the chmap get
callback clears the whole array no matter which count is set, and
rather the false-positive detection.
This patch fixes the problem by clearing only the needed array range
at usb_chmap_ctl_get().
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201211130048.6358-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 43d5ca88dfcd35e43010fdd818e067aa9a55f5ba upstream.
syzbot spotted a potential out-of-bounds shift in the USB-audio format
parser that receives the arbitrary shift value from the USB
descriptor.
Add a range check for avoiding the undefined behavior.
Reported-by: syzbot+df7dc146ebdd6435eea3@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201209084552.17109-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3fba05a2832f93b4d0cd4204f771fdae0d823114 upstream.
Fix to return a negative error code from the error handling case
instead of 0 in function wm_adsp_load(), as done elsewhere in this
function.
Fixes: 170b1e123f ("ASoC: wm_adsp: Add support for new Halo core DSPs")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Luo Meng <luomeng12@huawei.com>
Acked-by: Richard Fitzgerald <rf@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20201123133839.4073787-1-luomeng12@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 242d990c158d5b1dabd166516e21992baef5f26a upstream.
The generic parser accepts the preferred_dacs[] pairs as a hint for
assigning a DAC to each pin, but this hint doesn't work always
effectively. Currently it's merely a secondary choice after the trial
with the path index failed. This made sometimes it difficult to
assign DACs without mimicking the connection list and/or the badness
table.
This patch adds a new flag, obey_preferred_dacs, that changes the
behavior of the parser. As its name stands, the parser obeys the
given preferred_dacs[] pairs by skipping the path index matching and
giving a high penalty if no DAC is assigned by the pairs. This mode
will help for assigning the fixed DACs forcibly from the codec
driver.
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201127141104.11041-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit eeacd80fcb29b769ea915cd06b7dd35e0bf0bc25 upstream.
Some laptops like ASUS UX482EG & B9400CEA's headset audio does not work
until the quirk ALC294_FIXUP_ASUS_HPE is applied.
Signed-off-by: Jian-Hong Pan <jhp@endlessos.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201124092024.179540-1-jhp@endlessos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit aeedad2504997be262c98f6e3228173225a8d868 upstream.
HP Spectre x360 Convertible 15" version (SSID 103c:827f) needs the
same quirk to make the mute LED working like other models.
System Information
Manufacturer: HP
Product Name: HP Spectre x360 Convertible 15-bl1XX
Sound Codec:
Codec: Realtek ALC295
Vendor Id: 0x10ec0295
Subsystem Id: 0x103c827f
Revision Id: 0x100002
Reported-by: <christoph.plattner@gmx.at>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201128090015.7743-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c84bfedce60192c08455ee2d25dd13d19274a266 upstream.
ASUS Zephyrus G14 has two speaker pins, and the auto-parser tries to
assign an individual DAC to each pin as much as possible.
Unfortunately the third DAC has no volume control unlike the two DACs,
and this resulted in the inconsistent speaker volumes.
As a workaround, wire both speaker pins to the same DAC by modifying
the existing quirk (ALC289_FIXUP_ASUS_GA401) applied to this device.
Since this quirk entry is chained by another, we need to avoid
applying the DAC assignment change for it. Luckily, there is another
quirk entry (ALC289_FIXUP_ASUS_GA502) doing the very same thing, so we
can chain to the GA502 quirk instead.
Note that this patch uses a new flag of the generic parser,
obey_preferred_dacs, for enforcing the DACs.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210359
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201127141104.11041-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2d744ecf2b98405723a2138a547e5c75009bc4e5 upstream.
Automatically choose DMIC pipeline format configuration depending on
information included in NHLT.
Change the access rights of appropriate kcontrols to read-only in order
to prevent user interference.
Signed-off-by: Mateusz Gorski <mateusz.gorski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200427132727.24942-4-mateusz.gorski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org> # 5.4.x
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1b450791d517d4d6666ab9ab6d9a20c8819e3572 upstream.
For pipes supporting multiple input/output formats, kcontrol is
created and selection of pipe input and output configuration
is done based on control set.
If more than one configuration is supported, then this patch
allows user to select configuration of choice
using amixer settings.
Signed-off-by: Mateusz Gorski <mateusz.gorski@linux.intel.com>
Signed-off-by: Pavan K S <pavan.k.s@intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200427132727.24942-3-mateusz.gorski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org> # 5.4.x
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7693cadac86548b30389a6e11d78c38db654f393 upstream.
Each purge request is sent by driver after master core is powered up and
unresetted but before it is unstalled. On unstall, ROM begins processing
the request and initializing environment for FW load. Host should await
ROM's ack before moving forward. Without doing so, ROM init poll may
start too early and false timeouts can occur.
Fixes: cb6a552846 ("ASoC: Intel: cnl: Add sst library functions for cnl platform")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200305145314.32579-8-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org> # 5.4.x
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 024aa45f55ccd40704cfdef61b2a8b6d0de9cdd1 upstream.
Due to unconditional initial timeouts, firmware may fail to load during
its initialization. This issue cannot be resolved on driver side as it
is caused by external sources such as CSME but has to be accounted for
nonetheless.
Fixes: cb6a552846 ("ASoC: Intel: cnl: Add sst library functions for cnl platform")
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200305145314.32579-7-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org> # 5.4.x
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9e6c382f5a6161eb55115fb56614b9827f2e7da3 upstream.
Some configurations expose no NHLT table at all within their
/sys/firmware/acpi/tables. To prevent NULL-dereference errors from
occurring, adjust probe flow and append additional safety checks in
functions involved in NHLT lifecycle.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200305145314.32579-5-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org> # 5.4.x
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e603f11d5df8997d104ab405ff27640b90baffaa upstream.
Follow the recommendation set by hda_intel.c and enable HDMI/DP codec
wakeup during bus initialization procedure. Disable wakeup once init
completes.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200305145314.32579-4-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org> # 5.4.x
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a66f88394a78fec9a05fa6e517e9603e8eca8363 upstream.
With _reset_link removed from the probe sequence, codec_mask at the time
skl_find_hda_machine() is invoked will always be 0, so hda machine will
never be chosen. Rather than reorganizing boot flow, be permissive about
invalid mask. codec_mask will be set to proper value during probe_work -
before skl_codec_create() ever gets called.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200305145314.32579-3-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org> # 5.4.x
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 2ef81057d80456870b97890dd79c8f56a85b1242 upstream.
Skylake driver does the controller init operation twice:
- first during probe (only to stop it just before scheduling probe_work)
- and during said probe_work where the actual correct sequence is
executed
To properly complete boot sequence when iDisp codec is present, bus
initialization has to be called only after _i915_init() finishes.
With additional _reset_list preceding _i915_init(), iDisp codec never
gets the chance to enumerate on the link. Remove the superfluous
initialization to address the issue.
Signed-off-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200305145314.32579-2-cezary.rojewski@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org> # 5.4.x
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d21b96c8ed2aea7e6b7bf4735e1d2503cfbf4072 upstream.
The code change for switching to non-atomic mode brought the
unexpected mutex deadlock in get_msg(). It converted the spinlock
with the existing mutex, but there were calls with the already holding
the mutex. Since the only place that needs the extra lock is the code
path from snd_mixart_send_msg(), remove the mutex lock in get_msg()
and apply in the caller side for fixing the mutex deadlock.
Fixes: 8d3a8b5cb5 ("ALSA: mixart: Use nonatomic PCM ops")
Reported-by: Dan Carpenter <dan.carpenter@oracle.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201119121440.18945-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 95a793c3bc75cf888e0e641d656e7d080f487d8b upstream.
When processing request to add/replace user-defined element set, check
of given element identifier and decision of numeric identifier is done
in "__snd_ctl_add_replace()" helper function. When the result of check
is wrong, the helper function returns error code. The error code shall
be returned to userspace application.
Current implementation includes bug to return zero to userspace application
regardless of the result. This commit fixes the bug.
Cc: <stable@vger.kernel.org>
Fixes: e1a7bfe380 ("ALSA: control: Fix race between adding and removing a user element")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20201113092043.16148-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 54a2a3898f469a915510038fe84ef4f083131d3e upstream.
Found one more Logitech device, BCC950 ConferenceCam, which needs
the same delay here. This makes 3 out of 3 devices I have tried.
Therefore, add a delay for all Logitech devices as it does not hurt.
Signed-off-by: Joakim Tjernlund <joakim.tjernlund@infinera.com>
Cc: <stable@vger.kernel.org> # 4.19.y, 5.4.y
Link: https://lore.kernel.org/r/20201117122803.24310-1-joakim.tjernlund@infinera.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 02a9c6ee4183af2e438454c55098b828a96085fb upstream.
The spin_lock/unlock_irq() functions cannot be nested. The problem is
that presumably we would want the IRQs to be re-enabled on the second
call the spin_unlock_irq() but instead it will be enabled at the first
call so IRQs will be enabled earlier than expected.
In this situation the copy_resp_to_buf() function is only called from
one function and it is called with IRQs disabled. We can just use
the regular spin_lock/unlock() functions.
Fixes: 555e8a8f7f ("ALSA: fireworks: Add command/response functionality into hwdep interface")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201113101241.GB168908@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit bd6327fda2f3ded85b69b3c3125c99aaa51c7881 ]
lpass_pcm_data is not freed in error paths. Free it in
error paths to avoid memory leak.
Fixes: 022d00ee0b ("ASoC: lpass-platform: Fix broken pcm data usage")
Signed-off-by: Pavel Machek <pavel@ucw.cz>
Signed-off-by: V Sujith Kumar Reddy <vsujithk@codeaurora.org>
Signed-off-by: Srinivasa Rao Mandadapu <srivasam@codeaurora.org>
Link: https://lore.kernel.org/r/1605416210-14530-1-git-send-email-srivasam@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Add set_sysclk function to select preferred master input
clock on pcm512x codec, support multiple input system clocks
on SCLK master mode.
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
(cherry picked from commit 5ca64840578f9dad359d5c2e6821805df68a1608)
ASoC machine sound driver for IQAudio PiDAC plus/pro
Rev3 for iMX SoC, high resolution codec supporting
upto 384khz sample rate on SAI; Include support for
Hifiberry audio hats that uses external oscillators for
dac system clock.
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
(cherry picked from commit b52d3587cba2b3db60cf316430478969918fed7a)
[ Upstream commit 9fc149c3bce7bdbb94948a8e6bd025e3b3538603 ]
The broken jack detection should be fixed by commit a6e7d0a4bdb0 ("ALSA:
hda: fix jack detection with Realtek codecs when in D3"), let's try
enabling runtime PM by default again.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20201027130038.16463-4-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f5dac54d9d93826a776dffc848df76746f7135bb ]
Both pm_runtime_force_suspend() and pm_runtime_force_resume() have
some implicit checks, so it can make code flow more straightforward if
we separate runtime and system suspend callbacks.
High Definition Audio Specification, 4.5.9.3 Codec Wake From System S3
states that codec can wake the system up from S3 if WAKEEN is toggled.
Since HDA controller has different wakeup settings for runtime and
system susend, we also need to explicitly disable direct-complete which
can be enabled automatically by PCI core. In addition to that, avoid
waking up codec if runtime resume is for system suspend, to not break
direct-complete for codecs.
While at it, also remove AZX_DCAPS_SUSPEND_SPURIOUS_WAKEUP, as the
original bug commit a6630529aecb ("ALSA: hda: Workaround for spurious
wakeups on some Intel platforms") solves doesn't happen with this
patch.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20201027130038.16463-3-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 20afe581c9b980848ad097c4d54dde9bec7593ef ]
A delay must be introduced before the shutdown down of the mclk,
as stated in CS42L51 datasheet. Otherwise the codec may
produce some noise after the end of DAPM power down sequence.
The delay between DAC and CLOCK_SUPPLY widgets is too short.
Add a delay in mclk shutdown request to manage the shutdown delay
explicitly. From experiments, at least 10ms delay is necessary.
Set delay to 20ms as recommended in Documentation/timers/timers-howto.rst
when using msleep().
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/20201020150109.482-1-olivier.moysan@st.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3f48b6eba15ea342ef4cb420b580f5ed6605669f ]
With the current state of code, we would endup with something like
below in /proc/asound/cards for 2 machines based on this driver.
Machine 1:
0 [DB845c ]: DB845c - DB845c
DB845c
Machine 2:
0 [LenovoYOGAC6301]: Lenovo-YOGA-C63 - Lenovo-YOGA-C630-13Q50
LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216
This is not very UCM friendly both w.r.t to common up configs and
card identification, and UCM2 became totally not usefull with just
one ucm sdm845.conf for two machines which have different setups
w.r.t HDMI and other dais.
Reasons for such thing is partly because Qualcomm machine drivers never
cared to set driver_name.
This patch sets up driver name for the this driver to sort out the
UCM integration issues!
after this patch contents of /proc/asound/cards:
Machine 1:
0 [DB845c ]: sdm845 - DB845c
DB845c
Machine 2:
0 [LenovoYOGAC6301]: sdm845 - Lenovo-YOGA-C630-13Q50
LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216
with this its possible to align with what UCM2 expects and we can have
sdm845/DB845.conf
sdm845/LENOVO-81JL-LenovoYOGAC630_13Q50-LNVNB161216.conf
... for board variants. This should scale much better!
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20201023095849.22894-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 158e1886b6262c1d1c96a18c85fac5219b8bf804 ]
This is harmless, but the "addr" comes from the user and it could lead
to a negative shift or to shift wrapping if it's too high.
Fixes: 0b00a5615d ("ALSA: hdac_ext: add hdac extended controller")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20201103101807.GC1127762@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6d6bc54ab4f2404d46078abc04bf4dee4db01def ]
digital gain range is -84dB min to 40dB max, however this was not
correctly specified in the range.
Fix this by with correct range!
Fixes: 8c4f021d80 ("ASoC: wcd9335: add basic controls")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20201028154340.17090-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
When set_params return error, we should avoid to delete the component
in free function again. This may cause kernel crash.
Check cstream state to avoid such issue.
Signed-off-by: Zhang Peng <peng.zhang_8@nxp.com>
When DSP has no response with cplay and try to force exit, kernel
met crash. Reason is kernel locks a mutex proxy->lock twice then
kernel crash. Use MU_SendMessageTimeout instead MU_SendMessage
then kernel can release the mutex.
Signed-off-by: Zhang Peng <peng.zhang_8@nxp.com>
Get card name from model property, which is to align the name
with Linux Fractory.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
commit f15cfca818d756dd1c9492530091dfd583359db3 upstream.
The Zoom UAC-2 USB audio interface provides an async playback endpoint
("1 OUT (ASYNC)") and capture endpoint ("2 IN (ASYNC)"), both with
2-channel S32_LE in 44.1, 48, 88.2, 96, 176.4, or 192
kilosamples/s. The device provides explicit feedback to adjust the
host's playback rate, but the feedback appears unstable and biased
relative to the device's capture rate.
"alsaloop -t 1000" experiences playback underruns and tries to
resample the captured audio to match the varying playback
rate. Forcing the kernel to use implicit feedback appears to
produce more stable results. This causes the host to transmit one
playback sample for each capture sample received. (Zoom North America
has been notified of this change.)
Signed-off-by: Keith Winstein <keithw@cs.stanford.edu>
Tested-by: Keith Winstein <keithw@cs.stanford.edu>
Cc: <stable@vger.kernel.org>
BugLink: https://lore.kernel.org/r/20201027071841.GA164525@trolley.csail.mit.edu
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ef9ce66fab959c66d270bbee7ca79b92ee957893 upstream.
ASUS TM420 had depop circuit for headphone.
It need to turn on by COEF bit.
[ fixed the missing enum definition by tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/3d6177d7023b4783bf2793861c577ada@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8a8de09cb2adc119104f35044d1a840dd47aa9d8 upstream.
System boot with plugged headset. It will not detect headset Mic.
It will happen on cold boot restart resume state.
Quirk by SSID change to quirk by pin verb.
Fixes: 13468bfa8c58 ("ALSA: hda/realtek - set mic to auto detect on a HP AIO machine")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/f42ae1ede1cf47029ae2bef1a42caf03@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1b290ef023b3eeb4f4688b582fecb773915ef937 upstream.
Add alternative topology binary file name based on used machine driver
and fallback to use this name after failed attempt to load topology file
with name based on NHLT.
This change addresses multiple issues with current mechanism, for
example - there are devices without NHLT table, and that currently
results in tplg_name being empty.
Signed-off-by: Mateusz Gorski <mateusz.gorski@linux.intel.com>
Reviewed-by: Cezary Rojewski <cezary.rojewski@intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200427132727.24942-2-mateusz.gorski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit ed93f9750c6c2ed371347d0aac3dcd31cb9cf256 ]
Add AE-7 quirk data for setting of microphone. The AE-7 has no front
panel connector, so only rear-mic/line-in have new commands.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Link: https://lore.kernel.org/r/20200825201040.30339-19-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 22a16145af824f91014d07f8664114859900b9e6 ]
Instantiate snd_soc_dai_driver for independent symmetric control.
Otherwise the symmetric setting may be overwritten by other
instance.
Fixes: 08fdf65e37 ("ASoC: fsl_sai: Add asynchronous mode support")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Link: https://lore.kernel.org/r/1600424760-32071-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2759caad2600d503c3b0ed800e7e03d2cd7a4c05 ]
Recently we applied a fix to cover the whole OSS sequencer ioctls with
the mutex for dealing with the possible races. This works fine in
general, but in theory, this may lead to unexpectedly long stall if an
ioctl like SNDCTL_SEQ_SYNC is issued and an event with the far future
timestamp was queued.
For fixing such a potential stall, this patch changes the mutex lock
applied conditionally excluding such an ioctl command. Also, change
the mutex_lock() with the interruptible version for user to allow
escaping from the big-hammer mutex.
Fixes: 80982c7e834e ("ALSA: seq: oss: Serialize ioctls")
Suggested-by: Pavel Machek <pavel@ucw.cz>
Link: https://lore.kernel.org/r/20200922083856.28572-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 40b37136287ba6b34aa2f1f6123f3d6d205dc2f0 ]
Current code expects a single channel to be always used. Fix this
situation by forwarding the number of channels used. Then fix the
derivation of the bdiv clock rate.
Fixes: 96c3bb0023 ("ASoC: tlv320aic32x4: Dynamically Determine Clocking")
Suggested-by: Alexandre Belloni <alexandre.belloni@bootlin.com>
Signed-off-by: Miquel Raynal <miquel.raynal@bootlin.com>
Link: https://lore.kernel.org/r/20200911173140.29984-3-miquel.raynal@bootlin.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e525db7e4b44c5b2b5aac0dad24e23cb58c54d22 ]
if of_find_device_by_node() succeed, imx_es8328_probe() doesn't have
a corresponding put_device(). Thus add a jump target to fix the exception
handling for this function implementation.
Fixes: 7e7292dba2 ("ASoC: fsl: add imx-es8328 machine driver")
Signed-off-by: Yu Kuai <yukuai3@huawei.com>
Link: https://lore.kernel.org/r/20200825130224.1488694-1-yukuai3@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 753a6e17942f6f425ca622e1610625998312ad89 ]
i2sctl register value is set to 0 during hw_free(). This
impacts any ongoing concurrent session on the same i2s
port. As trigger() stop already resets enable bit to 0,
there is no need of explicit hw_free. Removing it to
fix the issue.
Fixes: 80beab8e1d ("ASoC: qcom: Add LPASS CPU DAI driver")
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-7-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5fd188215d4eb52703600d8986b22311099a5940 ]
lpass_pcm_data is never freed. Free it in close
ops to avoid memory leak.
Fixes: 022d00ee0b ("ASoC: lpass-platform: Fix broken pcm data usage")
Signed-off-by: Rohit kumar <rohitkr@codeaurora.org>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/1597402388-14112-5-git-send-email-rohitkr@codeaurora.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit ca184355db8e60290fa34bf61c13308e6f4f50d3 upstream.
The ASUS D700SA desktop's audio (1043:2390) with ALC887 cannot detect
the headset microphone and another headphone jack until
ALC887_FIXUP_ASUS_HMIC and ALC887_FIXUP_ASUS_AUDIO quirks are applied.
The NID 0x15 maps as the headset microphone and NID 0x19 maps as another
headphone jack. Also need the function like alc887_fixup_asus_jack to
enable the audio jacks.
Signed-off-by: Jian-Hong Pan <jhp@endlessos.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201007052224.22611-1-jhp@endlessos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 08befca40026136c14c3cd84f9e36c4cd20a358e upstream.
After installing archlinux, the mute led and micmute led are not working
at all. This patch fix this issue by applying a fixup from similar
model. These mute leds are confirmed working on HP Elitebook 845 G7.
Signed-off-by: Qiu Wenbo <qiuwenbo@kylinos.com.cn>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201002124454.7240-1-qiuwenbo@kylinos.com.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 13468bfa8c58731dc9ecda1cd9b22a191114f944 upstream.
Recently we enabled a HP AIO machine, we found the mic on the machine
couldn't record any sound and it couldn't detect plugging and
unplugging as well.
Through debugging we found the mic is set to manual detect mode, after
setting it to auto detect mode, it could detect plugging and
unplugging and could record sound.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200928080117.12435-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 148ebf548a1af366fc797fcc7d03f0bb92b12a79 upstream.
On a HP ZCentral, the front Mic could not be detected.
The codec of the HP ZCentrol is alc671 and it needs to override the pin
configuration to enable the headset mic.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201008105645.65505-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ce1558c285f9ad04c03b46833a028230771cc0a7 upstream.
A race exists between closing a PCM and update of ELD data. In
hdmi_pcm_close(), hinfo->nid value is modified without taking
spec->pcm_lock. If this happens concurrently while processing an ELD
update in hdmi_pcm_setup_pin(), converter assignment may be done
incorrectly.
This bug was found by hitting a WARN_ON in snd_hda_spdif_ctls_assign()
in a HDMI receiver connection stress test:
[2739.684569] WARNING: CPU: 5 PID: 2090 at sound/pci/hda/patch_hdmi.c:1898 check_non_pcm_per_cvt+0x41/0x50 [snd_hda_codec_hdmi]
...
[2739.684707] Call Trace:
[2739.684720] update_eld+0x121/0x5a0 [snd_hda_codec_hdmi]
[2739.684736] hdmi_present_sense+0x21e/0x3b0 [snd_hda_codec_hdmi]
[2739.684750] check_presence_and_report+0x81/0xd0 [snd_hda_codec_hdmi]
[2739.684842] intel_audio_codec_enable+0x122/0x190 [i915]
Fixes: 42b2987079 ("ALSA: hda - hdmi playback without monitor in dynamic pcm bind mode")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201013152628.920764-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a6e7d0a4bdb02a7a3ffe0b44aaa8842b7efdd056 upstream.
In case HDA controller becomes active, but codec is runtime suspended,
jack detection is not successful and no interrupt is raised. This has
been observed with multiple Realtek codecs and HDA controllers from
different vendors. Bug does not occur if both codec and controller are
active, or both are in suspend. Bug can be easily hit on desktop systems
with no built-in speaker.
The problem can be fixed by powering up the codec once after every
controller runtime resume. Even if codec goes back to suspend later, the
jack detection will continue to work. Add a flag to 'hda_codec' to
describe codecs that require this flow from the controller driver.
Modify __azx_runtime_resume() to use pm_request_resume() to make the
intent clearer.
Mark all Realtek codecs with the new forced_resume flag.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=209379
Cc: Kailang Yang <kailang@realtek.com>
Co-developed-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201012102704.794423-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b41c15f4e1c1f1657da15c482fa837c1b7384452 upstream.
The "count" variable needs to be capped on every path so that we don't
copy too much information to the user.
Fixes: 618eabeae7 ("ALSA: bebob: Add hwdep interface")
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Acked-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201007074928.GA2529578@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Make fsl_dsp_cpu as a separate driver, otherwise below
build error will occur when set CONFIG_SND_SOC_FSL_DSP=m
ld.lld: error: duplicate symbol: init_module
>>> defined at fsl_dsp.c
>>> sound/soc/fsl/fsl_dsp.o:(init_module)
>>> defined at fsl_dsp_cpu.c
>>> sound/soc/fsl/fsl_dsp_cpu.o:(.init.text+0x0)
Signed-off-by: Jindong <jindong.yue@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
After enable remote proc drvier, audio rpmsg
driver should not depend on HAVE_IMX_RPMSG.
Signed-off-by: Jindong <jindong.yue@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Move rpmsg wm8960 i2c driver to a new driver file,
otherwise it can not be built as module.
Signed-off-by: Jindong <jindong.yue@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Move i2s rpmsg driver to a new file: imx-i2s-rpmsg.c
This new driver should be probed after fsl_rpmsg_i2s, which
calls imx_rpmsg_platform_register (in imx-pcm-rpmsg) to fill
global variable: i2s_info_g.
Signed-off-by: Jindong <jindong.yue@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Add to check more sections for dsp firmware due to xtensa tools upgrade,
dsp firmware binary adds sections.
Signed-off-by: Zhang Peng <peng.zhang_8@nxp.com>
instantiate snd_soc_dai_link for multi instance on 8dxl evk board.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
* tag 'v5.4.70': (3051 commits)
Linux 5.4.70
netfilter: ctnetlink: add a range check for l3/l4 protonum
ep_create_wakeup_source(): dentry name can change under you...
...
Conflicts:
arch/arm/mach-imx/pm-imx6.c
arch/arm64/boot/dts/freescale/imx8mm-evk.dts
arch/arm64/boot/dts/freescale/imx8mn-ddr4-evk.dts
drivers/crypto/caam/caamalg.c
drivers/gpu/drm/imx/dw_hdmi-imx.c
drivers/gpu/drm/imx/imx-ldb.c
drivers/gpu/drm/imx/ipuv3/ipuv3-crtc.c
drivers/mmc/host/sdhci-esdhc-imx.c
drivers/net/ethernet/freescale/dpaa2/dpaa2-eth.c
drivers/net/ethernet/freescale/enetc/enetc.c
drivers/net/ethernet/freescale/enetc/enetc_pf.c
drivers/thermal/imx_thermal.c
drivers/usb/cdns3/ep0.c
drivers/xen/swiotlb-xen.c
sound/soc/fsl/fsl_esai.c
sound/soc/fsl/fsl_sai.c
Signed-off-by: Jason Liu <jason.hui.liu@nxp.com>
As per discussion with upstream this is not the correct way
to handle DSP resources. All the resource management was moved
to SOF IMX driver so this code is no longer used.
Removed it!
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
We move clocks handling from FSL DAI generic driver to IMX specific
driver as per discussion with upstream community. With this patch,
FSL DAI generic driver is no longer used and will be removed.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Because we no longer use FSL DAI generic driver, we register
supported DAI drivers now with audio DSP platform.
This will also simplify dts file.
Reviewed-by: Paul Olaru <paul.olaru@nxp.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
We no longer use FSL DAI generic driver as it is not
a good direction for upstreaming. Move clock handling
to SOF imx specific driver.
Note that clocks handled are for DAIs used by i.MX8/i.MX8X boards,
that is SAI1 and ESAI0.
Reviewed-by: Paul Olaru <paul.olaru@nxp.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
commit f73bbf639b32acb6b409e188fdde5644b301978f upstream.
On Lenovo P520, the front panel headset LED isn't lit up right now.
Realtek states that the LED needs to be enabled by ALC233's GPIO2, so
let's do it accordingly to light the LED up.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Acked-by: Hui Wang <hui.wang@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200914070231.13192-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3f74249057827c5f6676c41c18f6be12ce1469ce upstream.
We found a Mic detection issue on many Lenovo laptops, those laptops
belong to differnt models and they have different audio design like
internal mic connects to the codec or PCH, they all have this problem,
the problem is if plugging a headset before powerup/reboot the
machine, after booting up, the headphone could be detected but Mic
couldn't. If we plug out and plug in the headset, both headphone and
Mic could be detected then.
Through debugging we found the codec on those laptops are same, it is
alc257, and if we don't disable the 3k pulldown in alc256_shutup(),
the issue will be fixed. So far there is no pop noise or power
consumption regression on those laptops after this change.
Cc: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200914065118.19238-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 472eb39103e885f302fd8fd6eff104fcf5503f1b ]
clang static analysis flags this problem
hpioctl.c:513:7: warning: Branch condition evaluates to
a garbage value
if (pci.ap_mem_base[idx]) {
^~~~~~~~~~~~~~~~~~~~
If there is a failure in the middle of the memory space loop,
only some of the memory spaces need to be cleaned up.
At the error handler, idx holds the number of successful
memory spaces mapped. So rework the handler loop to use the
old idx.
There is a second problem, the memory space loop conditionally
iomaps()/sets the mem_base so it is necessay to initize pci.
Fixes: 719f82d398 ("ALSA: Add support of AudioScience ASI boards")
Signed-off-by: Tom Rix <trix@redhat.com>
Link: https://lore.kernel.org/r/20200913165230.17166-1-trix@redhat.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6a0137101f47301fff2da6ba4b9048383d569909 ]
The MPMAN Converter9 2-in-1 almost fully works with out default settings.
The only problem is that it has only 1 speaker so any sounds only playing
on the right channel get lost.
Add a quirk for this model using the default settings + MONO_SPEAKER.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200901080623.4987-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f5a2cda4f1db89776b64c4f0f2c2ac609527ac70 ]
When the wm8958_mic_detect, wm8994_mic_detect functions get called from
the machine driver, e.g. from the card's late_probe() callback, the CODEC
device may be PM runtime suspended and any regmap writes have no effect.
Add PM runtime calls to these functions to ensure the device registers
are updated as expected.
This suppresses an error during boot
"wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec"
caused by the regmap access error due to the cache_only flag being set.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200827173357.31891-2-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 811c5494436789e7149487c06e0602b507ce274b ]
The WM8994_MICBIAS register is not available in the WM1811 CODEC so skip
initialization of that register for that device.
This suppresses an error during boot:
"wm8994-codec: ASoC: error at snd_soc_component_update_bits on wm8994-codec"
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200827173357.31891-1-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7ad26d6671db758c959d7e1d100b138a38483612 ]
Some sound card try to set 0 Hz as reset, but it is impossible.
This patch ignores it to avoid error return.
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/87a6yjy5sy.wl-kuninori.morimoto.gx@renesas.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a6630529aecb5a3e84370c376ed658e892e6261e ]
We've received a regression report on Intel HD-audio controller that
wakes up immediately after S3 suspend. The bisection leads to the
commit c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not
needed"). This commit replaces the system-suspend to use
pm_runtime_force_suspend() instead of the direct call of
__azx_runtime_suspend(). However, by some really mysterious reason,
pm_runtime_force_suspend() causes a spurious wakeup (although it calls
the same __azx_runtime_suspend() internally).
As an ugly workaround for now, revert the behavior to call
__azx_runtime_suspend() and __azx_runtime_resume() for those old Intel
platforms that may exhibit such a problem, while keeping the new
standard pm_runtime_force_suspend() and pm_runtime_force_resume()
pair for the remaining chips.
Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208649
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200727164443.4233-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8d6762af302d69f76fa788a277a56a9d9cd275d5 ]
HD-audio codec driver applies a tricky procedure to forcibly perform
the runtime resume by mimicking the usage count even if the device has
been runtime-suspended beforehand. This was needed to assure to
trigger the jack detection update after the system resume.
And recently we also applied the similar logic to the HD-audio
controller side. However this seems leading to some inconsistency,
and eventually PCI controller gets screwed up.
This patch is an attempt to fix and clean up those behavior: instead
of the tricky runtime resume procedure, the existing jackpoll work is
scheduled when such a forced codec resume is required. The jackpoll
work will power up the codec, and this alone should suffice for the
jack status update in usual cases. If the extra polling is requested
(by checking codec->jackpoll_interval), the manual update is invoked
after that, and the codec is powered down again.
Also, we filter the spurious wake up of the codec from the controller
runtime resume by checking codec->relaxed_resume flag. If this flag
is set, basically we don't need to wake up explicitly, but it's
supposed to be done via the audio component notifier.
Fixes: c4c8dd6ef807 ("ALSA: hda: Skip controller resume if not needed")
Link: https://lore.kernel.org/r/20200422203744.26299-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 65bd91dd6957390c42a0491b9622cf31a2cdb140 ]
pm_runtime_get_sync() increments the runtime PM usage counter even
the call returns an error code. Thus a pairing decrement is needed
on the error handling path to keep the counter balanced.
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Link: https://lore.kernel.org/r/20200529012230.5863-1-dinghao.liu@zju.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c637fa151259c0f74665fde7cba5b7eac1417ae5 ]
The unsol event handling code has a loop retrieving the read/write
indices and the arrays without locking while the append to the array
may happen concurrently. This may lead to some inconsistency.
Although there hasn't been any proof of this bad results, it's still
safer to protect the racy accesses.
This patch adds the spinlock protection around the unsol handling loop
for addressing it. Here we take bus->reg_lock as the writer side
snd_hdac_bus_queue_event() is also protected by that lock.
Link: https://lore.kernel.org/r/20200516062556.30951-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c4c8dd6ef807663e42a5f04ea77cd62029eb99fa ]
The HD-audio controller does system-suspend and resume operations by
directly calling its helpers __azx_runtime_suspend() and
__azx_runtime_resume(). However, in general, we don't have to resume
always the device fully at the system resume; typically, if a device
has been runtime-suspended, we can leave it to runtime resume.
Usually for achieving this, the driver would call
pm_runtime_force_suspend() and pm_runtime_force_resume() pairs in the
system suspend and resume ops. Unfortunately, this doesn't work for
the resume path in our case. For handling the jack detection at the
system resume, a child codec device may need the (literally) forcibly
resume even if it's been runtime-suspended, and for that, the
controller device must be also resumed even if it's been suspended.
This patch is an attempt to improve the situation. It replaces the
direct __azx_runtime_suspend()/_resume() calls with with
pm_runtime_force_suspend() and pm_runtime_force_resume() with a slight
trick as we've done for the codec side. More exactly:
- azx_has_pm_runtime() check is dropped from azx_runtime_suspend() and
azx_runtime_resume(), so that it can be properly executed from the
system-suspend/resume path
- The WAKEEN handling depends on the card's power state now; it's set
and cleared only for the runtime-suspend
- azx_resume() checks whether any codec may need the forcible resume
beforehand. If the forcible resume is required, it does temporary
PM refcount up/down for actually triggering the runtime resume.
- A new helper function, hda_codec_need_resume(), is introduced for
checking whether the codec needs a forcible runtime-resume, and the
existing code is rewritten with that.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5c6cd7021a05a02fcf37f360592d7c18d4d807fb ]
The Miditech MIDIFACE 16x16 (USB ID 1290:1749) has more than one extra
endpoint descriptor.
The first extra descriptor is: 0x06 0x30 0x00 0x00 0x00 0x00
As the code in snd_usbmidi_get_ms_info() looks only at the
first extra descriptor to find USB_DT_CS_ENDPOINT the device
as such is recognized but there is neither input nor output
configured.
The patch iterates through the extra descriptors to find the
proper one. With this patch the device is correctly configured.
Signed-off-by: Andreas Steinmetz <ast@domdv.de>
Link: https://lore.kernel.org/r/1c3b431a86f69e1d60745b6110cdb93c299f120b.camel@domdv.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1919b42ca4ad75a2397081164661af3ce5a7b8f4 ]
In tx_wait_done the ipc payload is copied before the DSP transaction
error code is checked. This might lead to corrupted data in kernel side
even though the error would be handled later. It is also pointless to
copy the data in case of error. So change the order of error check and
copy.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Jaska Uimonen <jaska.uimonen@linux.intel.com>
Link: https://lore.kernel.org/r/20200228231850.9226-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit e9a0ef0b5ddcbc0d56c65aefc0f18d16e6f71207 ]
Some USB-audio descriptors provide a bogus volume range (e.g. volume
min and max are identical), which confuses user-space.
This patch makes the driver skipping such a control element.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206221
Link: https://lore.kernel.org/r/20200214144928.23628-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6d011d5057ff88ee556c000ac6fe0be23bdfcd72 ]
RIRB interrupt status getting cleared after the write pointer is read
causes a race condition, where last response(s) into RIRB may remain
unserviced by IRQ, eventually causing azx_rirb_get_response to fall
back to polling mode. Clearing the RIRB interrupt status ahead of
write pointer access ensures that this condition is avoided.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Signed-off-by: Viswanath L <viswanathl@nvidia.com>
Link: https://lore.kernel.org/r/1580983853-351-1-git-send-email-viswanathl@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit acb874a7c049ec49d8fc66c893170fb42c01bdf7 ]
It was observed Baytrail-based chromebooks could cause continuous PLL
unlocked when using playback stream and capture stream simultaneously.
Specifically, starting a capture stream after started a playback stream.
As a result, the audio data could corrupt or turn completely silent.
As the datasheet suggested, the maximum PLL lock time should be 7 msec.
The workaround resets the codec softly by toggling SHDN off and on if
PLL failed to lock for 10 msec. Notably, there is no suggested hold
time for SHDN off.
On Baytrail-based chromebooks, it would easily happen continuous PLL
unlocked if there is a 10 msec delay between SHDN off and on. Removes
the msleep().
Signed-off-by: Tzung-Bi Shih <tzungbi@google.com>
Link: https://lore.kernel.org/r/20191122073114.219945-2-tzungbi@google.com
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 175fc928198236037174e5c5c066fe3c4691903e ]
Propagate the error code from request_irq(), rather than returning
-EBUSY.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Link: https://lore.kernel.org/r/E1iNIqh-0000tW-EZ@rmk-PC.armlinux.org.uk
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit fc19d559b0d31b5b831fd468b10d7dadafc0d0ec upstream.
The Mic connects to the Nid 0x19, but the configuration of Nid 0x19
is not defined to Mic, and also need to set the coeff to enable the
auto detection on the Nid 0x19. After this change, the Mic plugging
in or plugging out could be detected and could record the sound from
the Mic.
And the coeff value is suggested by Kailang of Realtek.
Cc: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200909020041.8967-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c3cdf189276c2a63da62ee250615bd55e3fb680d upstream.
The GX502 requires a few steps to enable the headset i/o: pincfg,
verbs to enable and unmute the amp used for headpone out, and
a jacksense callback to toggle output via internal or jack using
a verb.
Signed-off-by: Luke D Jones <luke@ljones.dev>
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208005
Link: https://lore.kernel.org/r/20200907081959.56186-1-luke@ljones.dev
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
This reverts commit 9825056150 which is
commit 15cbff3fbbc631952c346744f862fb294504b5e2 upstream.
It causes know regressions and will be reverted in Linus's tree soon.
Reported-by: Hans de Goede <hdegoede@redhat.com>
Cc: Dan Crawford <dnlcrwfrd@gmail.com>
Cc: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/7efd2fe5-bf38-7f85-891a-eee3845d1493@redhat.com
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 9c4b205a20f483d8a5d1208cfec33e339347d4bd ]
On g12 and following platforms, The first channel of record with more than
2 channels ends being placed randomly on an even channel of the output.
On these SoCs, a bit was added to force the first channel to be placed at
the beginning of the output. Apparently the behavior if the bit is not set
is not easily predictable. According to the documentation, this bit is not
present on the axg series.
Set the bit on g12 and fix the problem.
Fixes: a3c23a8ad4 ("ASoC: meson: axg-toddr: add g12a support")
Reported-by: Nicolas Belin <nbelin@baylibre.com>
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200828151438.350974-1-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c1e6414cdc371f9ed82cefebba7538499a3059f9 ]
for_each_child_of_node returns a node pointer np with
refcount incremented. So when devm_kzalloc fails, a
pairing refcount decrement is needed to keep np's
refcount balanced.
Fixes: 16395ceee1 ("ASoC: qcom: common: Fix NULL pointer in of parser")
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Link: https://lore.kernel.org/r/20200820042828.10308-1-dinghao.liu@zju.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3c27ea23ffb43262da6c64964163895951aaed4e ]
On Linux 5.9-rc1 I get the following warning with apq8016-sbc:
WARNING: CPU: 2 PID: 69 at sound/core/init.c:207 snd_card_new+0x36c/0x3b0 [snd]
CPU: 2 PID: 69 Comm: kworker/2:1 Not tainted 5.9.0-rc1 #1
Workqueue: events deferred_probe_work_func
pc : snd_card_new+0x36c/0x3b0 [snd]
lr : snd_card_new+0xf4/0x3b0 [snd]
Call trace:
snd_card_new+0x36c/0x3b0 [snd]
snd_soc_bind_card+0x340/0x9a0 [snd_soc_core]
snd_soc_register_card+0xf4/0x110 [snd_soc_core]
devm_snd_soc_register_card+0x44/0xa0 [snd_soc_core]
apq8016_sbc_platform_probe+0x11c/0x140 [snd_soc_apq8016_sbc]
This warning was introduced in
commit 81033c6b584b ("ALSA: core: Warn on empty module").
It looks like we are supposed to set card->owner to THIS_MODULE.
Fix this for all the qcom ASoC drivers.
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Fixes: 79119c7986 ("ASoC: qcom: Add Storm machine driver")
Fixes: bdb052e81f ("ASoC: qcom: add apq8016 sound card support")
Fixes: a6f933f63f ("ASoC: qcom: apq8096: Add db820c machine driver")
Fixes: 6b1687bf76 ("ASoC: qcom: add sdm845 sound card support")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200820154511.203072-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 13774d81f38538c5fa2924bdcdfa509155480fa6 ]
In snd_hdac_device_init pm_runtime_set_active is called to
increase child_count in parent device. But when it is failed
to build connection with GPU for one case that integrated
graphic gpu is disabled, snd_hdac_ext_bus_device_exit will be
invoked to clean up a HD-audio extended codec base device. At
this time the child_count of parent is not decreased, which
makes parent device can't get suspended.
This patch calls pm_runtime_set_suspended to decrease child_count
in parent device in snd_hdac_device_exit to match with
snd_hdac_device_init. pm_runtime_set_suspended can make sure that
it will not decrease child_count if the device is already suspended.
Signed-off-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200902154218.1440441-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 23d63a31d9f44d7daeac0d1fb65c6a73c70e5216 ]
The WAKEEN bits are used to indicate which bits in the
STATESTS register may cause wake event during the codec
state change request. Configure the WAKEEN register for
the Tegra to detect the wake events.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20200825052415.20626-3-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 216116eae43963c662eb84729507bad95214ca6b ]
The Tegra HDA codec HW implementation has an issue related to not
swapping the 2 channel Audio Sample Packet(ASP) channel mapping.
Whatever the FL and FR mapping specified the left channel always
comes out of left speaker and right channel on right speaker. So
add condition to disallow the swapping of FL,FR during the playback.
Signed-off-by: Mohan Kumar <mkumard@nvidia.com>
Acked-by: Sameer Pujar <spujar@nvidia.com>
Link: https://lore.kernel.org/r/20200825052415.20626-2-mkumard@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Previously update pointer in mix function according to dma callback,
which is not accurate for the callback maybe triggerred very late, that
cause the read pointer and write pointer point to same period, then
there is noise in output.
So get pointer directly from dma driver to avoid such issue.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
Previously doing mixing in softirq, the mixing causing too much
time ( period size is 512 sample, the time is about 2.5ms), so
move the mixing to workqueue to reduce irq loading
Use 4 period instead of pingpong buffer for leverage instant loading.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
commit 6a6660d049f88b89fd9a4b9db3581b245f7782fa upstream.
There've been quite a few regression reports about the lowered volume
(reduced to ca 65% from the previous level) on Lenovo Thinkpad X1
after the commit d2cd795c4ece ("ALSA: hda - fixup for the bass speaker
on Lenovo Carbon X1 7th gen"). Although the commit itself does the
right thing from HD-audio POV in order to have a volume control for
bass speakers, it seems that the machine has some secret recipe under
the hood.
Through experiments, Benjamin Poirier found out that the following
routing gives the best result:
* DAC1 (NID 0x02) -> Speaker pin (NID 0x14)
* DAC2 (NID 0x03) -> Shared by both Bass Speaker pin (NID 0x17) &
Headphone pin (0x21)
* DAC3 (NID 0x06) -> Unused
DAC1 seems to have some equalizer internally applied, and you'd get
again the output in a bad quality if you connect this to the
headphone pin. Hence the headphone is connected to DAC2, which is now
shared with the bass speaker pin. DAC3 has no volume amp, hence it's
not connected at all.
For achieving the routing above, this patch introduced a couple of
workarounds:
* The connection list of bass speaker pin (NID 0x17) is reduced not to
include DAC3 (NID 0x06)
* Pass preferred_pairs array to specify the fixed connection
Here, both workarounds are needed because the generic parser prefers
the individual DAC assignment over others.
When the routing above is applied, the generic parser creates the two
volume controls "Front" and "Bass Speaker". Since we have only two
DACs for three output pins, those are not fully controlling each
output individually, and it would confuse PulseAudio. For avoiding
the pitfall, in this patch, we rename those volume controls to some
unique ones ("DAC1" and "DAC2"). Then PulseAudio ignore them and
concentrate only on the still good-working "Master" volume control.
If a user still wants to control each DAC volume, they can still
change manually via "DAC1" and "DAC2" volume controls.
Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen")
Reported-by: Benjamin Poirier <benjamin.poirier@gmail.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Tested-by: Benjamin Poirier <benjamin.poirier@gmail.com>
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207407#c10
BugLink: https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3214171
BugLink: https://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3276276
Link: https://lore/kernel.org/r/20200829112746.3118-1-benjamin.poirier@gmail.com
Link: https://lore.kernel.org/r/20200903083300.6333-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8bcea6cb2cbc1f749e574954569323dec5e2920e upstream.
The Galaxy Book Ion NT950XCJ-X716A (15 inches) uses the same ALC298
codec as other Samsung laptops which have the no headphone sound bug. I
confirmed on my own hardware that this fixes the bug.
This also correct the model name for the 13 inches version. It was
incorrectly referenced as NT950XCJ-X716A in commit e17f02d05. But it
should have been NP930XCJ-K01US.
Fixes: e17f02d0559c ("ALSA: hda/realtek: Add quirk for Samsung Galaxy Book Ion")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423
Signed-off-by: Adrien Crivelli <adrien.crivelli@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200826084014.211217-1-adrien.crivelli@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 0bd8bce897b6697bbc286b8ba473aa0705fe394b upstream.
Tascam FE-8 is known to support communication by asynchronous transaction
only. The support can be implemented in userspace application and
snd-firewire-ctl-services project has the support. However, ALSA
firewire-tascam driver is bound to the model.
This commit changes device entries so that the model is excluded. In a
commit 53b3ffee78 ("ALSA: firewire-tascam: change device probing
processing"), I addressed to the concern that version field in
configuration differs depending on installed firmware. However, as long
as I checked, the version number is fixed. It's safe to return version
number back to modalias.
Fixes: 53b3ffee78 ("ALSA: firewire-tascam: change device probing processing")
Cc: <stable@vger.kernel.org> # 4.4+
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200823075537.56255-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 15cbff3fbbc631952c346744f862fb294504b5e2 upstream.
Following Christian Lachner's patch for Gigabyte X570-based motherboards,
also patch the MSI X570-A PRO motherboard; the ALC1220 codec requires the
same workaround for Clevo laptops to enforce the DAC/mixer connection
path. Set up a quirk entry for that.
I suspect most if all X570 motherboards will require similar patches.
[ The entries reordered in the SSID order -- tiwai ]
Related buglink: https://bugzilla.kernel.org/show_bug.cgi?id=205275
Signed-off-by: Dan Crawford <dnlcrwfrd@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200829024946.5691-1-dnlcrwfrd@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 858e0ad9301d1270c02b5aca97537d2d6ee9dd68 upstream.
When system is suspended with active audio playback to HDMI/DP, two
alternative sequences can happen at resume:
a) monitor is detected first and ALSA prepare follows normal
stream setup sequence, or
b) ALSA prepare is called first, but monitor is not yet detected,
so PCM is restarted without a pin,
In case of (b), on i915 systems, haswell_verify_D0() is not called at
resume and the pin power state may be incorrect. Result is lack of audio
after resume with no error reported back to user-space.
Fix the problem by always verifying converter and pin state in the
i915_pin_cvt_fixup().
BugLink: https://github.com/thesofproject/linux/issues/2388
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200826170306.701566-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 949a1ebe8cea7b342085cb6a4946b498306b9493 upstream.
The PCM OSS mulaw plugin has a check of the format of the counter part
whether it's a linear format. The check is with snd_BUG_ON() that
emits WARN_ON() when the debug config is set, and it confuses
syzkaller as if it were a serious issue. Let's drop snd_BUG_ON() for
avoiding that.
While we're at it, correct the error code to a more suitable, EINVAL.
Reported-by: syzbot+23b22dc2e0b81cbfcc95@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200901131802.18157-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7c5b892e0871655fea3294ffac6fa3cc3400b60d upstream.
This uses the same quirk as the Motu and SSL2 devices.
Tested on the UR22C.
Fixes bug 208851.
Signed-off-by: Joshua Sivec <sivec@posteo.net>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208851
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200825165515.8239-1-sivec@posteo.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit eed8f88b109aa927fbf0d0c80ff9f8d00444ca7f upstream.
This reverts commit 61eee4a7fc40 ("ALSA: hda: Add support for Loongson
7A1000 controller") to fix the following error on the Loongson LS7A
platform:
rcu: INFO: rcu_preempt self-detected stall on CPU
<SNIP>
NMI backtrace for cpu 0
CPU: 0 PID: 68 Comm: kworker/0:2 Not tainted 5.8.0+ #3
Hardware name: , BIOS
Workqueue: events azx_probe_work [snd_hda_intel]
<SNIP>
Call Trace:
[<ffffffff80211a64>] show_stack+0x9c/0x130
[<ffffffff8065a740>] dump_stack+0xb0/0xf0
[<ffffffff80665774>] nmi_cpu_backtrace+0x134/0x140
[<ffffffff80665910>] nmi_trigger_cpumask_backtrace+0x190/0x200
[<ffffffff802b1abc>] rcu_dump_cpu_stacks+0x12c/0x190
[<ffffffff802b08cc>] rcu_sched_clock_irq+0xa2c/0xfc8
[<ffffffff802b91d4>] update_process_times+0x2c/0xb8
[<ffffffff802cad80>] tick_sched_timer+0x40/0xb8
[<ffffffff802ba5f0>] __hrtimer_run_queues+0x118/0x1d0
[<ffffffff802bab74>] hrtimer_interrupt+0x12c/0x2d8
[<ffffffff8021547c>] c0_compare_interrupt+0x74/0xa0
[<ffffffff80296bd0>] __handle_irq_event_percpu+0xa8/0x198
[<ffffffff80296cf0>] handle_irq_event_percpu+0x30/0x90
[<ffffffff8029d958>] handle_percpu_irq+0x88/0xb8
[<ffffffff80296124>] generic_handle_irq+0x44/0x60
[<ffffffff80b3cfd0>] do_IRQ+0x18/0x28
[<ffffffff8067ace4>] plat_irq_dispatch+0x64/0x100
[<ffffffff80209a20>] handle_int+0x140/0x14c
[<ffffffff802402e8>] irq_exit+0xf8/0x100
Because AZX_DRIVER_GENERIC can not work well for Loongson LS7A HDA
controller, it needs some workarounds which are not merged into the
upstream kernel at this time, so it should revert this patch now.
Fixes: 61eee4a7fc40 ("ALSA: hda: Add support for Loongson 7A1000 controller")
Cc: <stable@vger.kernel.org> # 5.9-rc1+
Signed-off-by: Tiezhu Yang <yangtiezhu@loongson.cn>
Link: https://lore.kernel.org/r/1598348388-2518-1-git-send-email-yangtiezhu@loongson.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Compress dsp now only support one instance in the meantime, prevent
to play the secend one.
Fix issue The DSP stops to work if try to start the second cplay to DSP.
Signed-off-by: Zhang Peng <peng.zhang_8@nxp.com>
commit 74a2a7de81a2ef20732ec02087314e92692a7a1b upstream.
As the recent fix addressed the channel swap problem more properly,
update the comment as well.
Fixes: 1b7ecc241a67 ("ALSA: usb-audio: work around streaming quirk for MacroSilicon MS2109")
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200816084431.102151-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit f082bb59b72039a2326ec1a44496899fb8aa6d0e ]
The driver supports WM1811, WM8994, WM8958 devices but according to
documentation and the regmap definitions the WM8958_DSP2_* registers
are only available on WM8958. In current code these registers are
being accessed as if they were available on all the three chips.
When starting playback on WM1811 CODEC multiple errors like:
"wm8994-codec wm8994-codec: ASoC: error at soc_component_read_no_lock on wm8994-codec: -5"
can be seen, which is caused by attempts to read an unavailable
WM8958_DSP2_PROGRAM register. The issue has been uncovered by recent
commit "e2329ee ASoC: soc-component: add soc_component_err()".
This patch adds a check in wm8958_aif_ev() callback so the DSP2 handling
is only done for WM8958.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Link: https://lore.kernel.org/r/20200731173834.23832-1-s.nawrocki@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 23dc958689449be85e39351a8c809c3d344b155b ]
The very quiet and distorted headphone output bug that afflicted my
Samsung Notebook 9 is appearing in many other Samsung laptops. Expose
the quirk which fixed my laptop as a model so other users can try it.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423
Signed-off-by: Mike Pozulp <pozulp.kernel@gmail.com>
Link: https://lore.kernel.org/r/20200817043219.458889-1-pozulp.kernel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 470757f5b3a46bd85741bb0d8c1fd3f21048a2af ]
Capture and playback endpoints on Saffire 6 (USB 1.1) resides on the same
interface. This was not supported by the composite quirk back in the day
when initial support for this device was added, thus only playback was
enabled until now.
Fixes: 11e424e88b ("ALSA: usb-audio: Add support for Focusrite Saffire 6 USB")
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Cc: <stable.vger.kernel.org>
Link: https://lore.kernel.org/r/20200815002103.29247-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit deca195383a6085be62cb453079e03e04d618d6e ]
Calling pm_runtime_get_sync increments the counter even in case of
failure, causing incorrect ref count if pm_runtime_put is not called in
error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails.
Signed-off-by: Qiushi Wu <wu000273@umn.edu>
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Link: https://lore.kernel.org/r/20200613204422.24484-1-wu000273@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6b9fbb073636906eee9fe4d4c05a4f445b9e2a23 ]
pm_runtime_get_sync() increments the runtime PM usage counter even
when it returns an error code, causing incorrect ref count if
pm_runtime_put_noidle() is not called in error handling paths.
Thus call pm_runtime_put_noidle() if pm_runtime_get_sync() fails.
Signed-off-by: Qiushi Wu <wu000273@umn.edu>
Link: https://lore.kernel.org/r/20200614033344.1814-1-wu000273@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c4c59b95b7f7d4cef5071b151be2dadb33f3287b ]
pm_runtime_get_sync() increments the runtime PM usage counter even
when it returns an error code, causing incorrect ref count if
pm_runtime_put_noidle() is not called in error handling paths.
Thus call pm_runtime_put_noidle() if pm_runtime_get_sync() fails.
Signed-off-by: Qiushi Wu <wu000273@umn.edu>
Link: https://lore.kernel.org/r/20200614033749.2975-1-wu000273@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d96f27c80b65437a7b572647ecb4717ec9a50c98 ]
There's another HP desktop has buggy BIOS which flags the Port
Connectivity bit as no connection.
Apply force connectivity quirk to enable DP/HDMI audio.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200811095336.32396-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit cd72c317a0a11f64225b9a3f1fe503bb8c7327b5 ]
HDMI on some platforms doesn't enable audio support because its Port
Connectivity [31:30] is set to AC_JACK_PORT_NONE:
Node 0x05 [Pin Complex] wcaps 0x40778d: 8-Channels Digital Amp-Out CP
Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1
Amp-Out vals: [0x00 0x00]
Pincap 0x0b000094: OUT Detect HBR HDMI DP
Pin Default 0x58560010: [N/A] Digital Out at Int HDMI
Conn = Digital, Color = Unknown
DefAssociation = 0x1, Sequence = 0x0
Pin-ctls: 0x40: OUT
Unsolicited: tag=00, enabled=0
Power states: D0 D3 EPSS
Power: setting=D0, actual=D0
Devices: 0
Connection: 3
0x02 0x03* 0x04
For now, use a quirk to force connectivity based on SSID. If there are
more platforms affected by the same issue, we can eye for a more generic
solution.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200804155836.16252-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
When there is no asrc node, there will be below error:
imx-cs42888 sound-cs42888: ASoC: Neither/both codec name/of_node are set for HiFi-ASRC-BE
Add codec node to fix this issue
Fixes: 44dec4241b ("MLK-24612-3: ASoC: imx-cs42888: Support two client device")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
Support two client device, each client is 2 channel
48000Hz, 16bits.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
ESAI mixer is for mixing the data from clients. There is
a ping-pong buffer in the mixer for storing the mixed data.
The period size is same as the period size in client (unit is
sample number).
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
ESAI client driver is working as cpu dai in front-end.
ESAI is an unity, which can't be separated to multiple device
in hardware. so we need to separate it in software view.
Each client device can accept data from user space indepedently.
Then the data of different client is mixed together before sent
to ESAI interface.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
[ Upstream commit 062fa09f44f4fb3776a23184d5d296b0c8872eb9 ]
When power_up_sst() fails, stream needs to be freed
just like when try_module_get() fails. However, current
code is returning directly and ends up leaking memory.
Fixes: 0121327c1a ("ASoC: Intel: mfld-pcm: add control for powering up/down dsp")
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200813084112.26205-1-dinghao.liu@zju.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ff69c97ef84c9f7795adb49e9f07c9adcdd0c288 ]
For some reason interrupt set and clear register offsets are
not set correctly.
This patch corrects them!
Fixes: 585e881e5b ("ASoC: codecs: Add msm8916-wcd analog codec")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Stephan Gerhold <stephan@gerhold.net>
Link: https://lore.kernel.org/r/20200811103452.20448-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 796a58fe2b8c9b6668db00d92512ec84be663027 ]
Most of the DAPM widgets for DSP ASoC components reuse reg field
of the widgets for its internal calculations, however these are not
real registers. So read/writes to these numbers are not really
valid. However ASoC core will read these registers to get default
state during startup.
With recent changes to ASoC core, every register read/write
failures are reported very verbosely. Prior to this fails to reads
are totally ignored, so we never saw any error messages.
To fix this add dummy read/write function to return default value.
Fixes: e3a33673e8 ("ASoC: qdsp6: q6routing: Add q6routing driver")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200811120205.21805-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 56235e4bc5ae58cb8fcd9314dba4e9ab077ddda8 ]
Looks like the q6afe-dai dapm widget registers are set as "0",
which is a not correct.
As this registers will be read by ASoC core during startup
which will throw up errors, Fix this by making the registers
as SND_SOC_NOPM as these should be never used.
With recent changes to ASoC core, every register read/write
failures are reported very verbosely. Prior to this fails to reads
are totally ignored, so we never saw any error messages.
Fixes: 24c4cbcfac ("ASoC: qdsp6: q6afe: Add q6afe dai driver")
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200811120205.21805-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit e17f02d0559c174cf1f6435e45134490111eaa37 upstream.
The Galaxy Book Ion uses the same ALC298 codec as other Samsung laptops
which have the no headphone sound bug, like my Samsung Notebook. The
Galaxy Book owner confirmed that this patch fixes the bug.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423
Signed-off-by: Mike Pozulp <pozulp.kernel@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200818165446.499821-1-pozulp.kernel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f70fff83cda63bbf596f99edc131b9daaba07458 upstream.
The Flex Book uses the same ALC298 codec as other Samsung laptops which
have the no headphone sound bug, like my Samsung Notebook. The Flex Book
owner used Early Patching to confirm that this quirk fixes the bug.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423
Signed-off-by: Mike Pozulp <pozulp.kernel@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200814045346.645367-1-pozulp.kernel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Commit 89d64ed05d ("MLK-24263-8 ASoC: SOF: Add support for system /
runtime PM on i.MX8") introduced runtime PM handlers for i.MX8QM but
forgot to add the same handlers for i.MX8QXP.
Do it now!
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
[ Upstream commit 5a25de6df789cc805a9b8ba7ab5deef5067af47e ]
Freeing chip on error may lead to an Oops at the next time
the system goes to resume. Fix this by removing all
snd_echo_free() calls on error.
Fixes: 47b5d028fd ("ALSA: Echoaudio - Add suspend support #2")
Signed-off-by: Dinghao Liu <dinghao.liu@zju.edu.cn>
Link: https://lore.kernel.org/r/20200813074632.17022-1-dinghao.liu@zju.edu.cn
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
POST_PROCESS component takes care of processing algorithms. For now,
we intend to use this component for decoding/encoding library algorithms
from Cadence.
Current implementation assumes that we will still use ALSA PCM
interface. The heavy part is done on firmware side with the help
of topology files.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
commit 6e8596172ee1cd46ec0bfd5adcf4ff86371478b6 upstream.
This is just another Pioneer device with fixed endpoints. Input is dummy
but used as feedback (it always returns silence).
Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200810082502.225979-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1b7ecc241a67ad6b584e071bd791a54e0cd5f097 upstream.
Further investigation of the L-R swap problem on the MS2109 reveals that
the problem isn't that the channels are swapped, but rather that they
are swapped and also out of phase by one sample. In other words, the
issue is actually that the very first frame that comes from the hardware
is a half-frame containing only the right channel, and after that
everything becomes offset.
So introduce a new quirk field to drop the very first 2 bytes that come
in after the format is configured and a capture stream starts. This puts
the channels in phase and in the correct order.
Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200810082400.225858-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fec9008828cde0076aae595ac031bfcf49d335a4 upstream.
Adds an entry for Creative USB X-Fi to the rc_config array in
mixer_quirks.c to allow use of volume knob on the device.
Adds support for newer X-Fi Pro card, known as "Model No. SB1095"
with USB ID "041e:3263"
Signed-off-by: Mirko Dietrich <buzz@l4m1.de>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200806124850.20334-1-buzz@l4m1.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 386a6539992b82fe9ac4f9dc3f548956fd894d8c upstream.
After installing the Ubuntu Linux, the micmute led status is not
correct. Users expect that the led is on if the capture is disabled,
but with the current kernel, the led is off with the capture disabled.
We tried the old linux kernel like linux-4.15, there is no this issue.
It looks like we introduced this issue when switching to the led_cdev.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200810021659.7429-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 5aef1ff2397d021f93d874b57dff032fdfac73de ]
The fifo_depth is 64 on i.MX8QM/i.MX8QXP, 128 on i.MX8MQ, 16 on
i.MX7ULP.
Original FSL_SAI_CR1_RFW_MASK value 0x1F is not suitable for
these platform, the FIFO watermark mask should be updated
according to the fifo_depth.
Fixes: a860fac420 ("ASoC: fsl_sai: Add support for imx7ulp/imx8mq")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Link: https://lore.kernel.org/r/1596176895-28724-1-git-send-email-shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0d3f01dcdc234001f979a0af0b6b31cb9f25b6c1 ]
After carefully checking, it appears that both tdmout and tdmin require the
rising edge of the sclk they get to be synchronized with the frame sync
event (which should be a rising edge of lrclk).
TDMIN was improperly set before this patch. Remove the sclk_invert quirk
which is no longer needed and fix the sclk phase.
Fixes: 1a11d88f49 ("ASoC: meson: add tdm formatter base driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-4-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 80a254394fcfe55450b0351da298ca7231889219 ]
After carefully checking the result provided by the TDMIN on the g12a and
sm1 SoC families, the TDMIN skew offset appears to be 3 instead of 2 on the
axg.
Fixes: f01bc67f58 ("ASoC: meson: axg-tdm-formatter: rework quirks settings")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-3-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6878ba91ce84f7a07887a0615af70f969508839f ]
The .set_fmt() callback of the axg tdm interface incorrectly
test the content of SND_SOC_DAIFMT_MASTER_MASK as if it was a
bitfield, which it is not.
Implement the test correctly.
Fixes: d60e4f1e4b ("ASoC: meson: add tdm interface driver")
Signed-off-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200729154456.1983396-2-jbrunet@baylibre.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit bd054ece7d9cdd88e900df6625e951a01d9f655e ]
axg_card_add_tdm_loopback() misses to call kfree() in an error path. We
can use devm_kasprintf() to fix the issue, also improve maintainability.
So use it instead.
Fixes: c84836d7f6 ("ASoC: meson: axg-card: use modern dai_link style")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200717082242.130627-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 88cee34b776f80d2da04afb990c2a28c36799c43 ]
This field is required for ASoC cards. Not setting it will result in a
module->name pointer being NULL and generate problems such as
cat /proc/asound/modules
0 (efault)
Fixes: 76016322ec ('ASoC: Intel: Add Broxton-P machine driver')
Reported-by: Jaroslav Kysela <perex@perex.cz>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200625191308.3322-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8753889e2720c1ef7ebf03370e384f5bf5ff4fab ]
This field is required for ASoC cards. Not setting it will result in a
module->name pointer being NULL and generate problems such as
cat /proc/asound/modules
0 (efault)
Fixes: 8017b8fd37 ('ASoC: SOF: Add Nocodec machine driver support')
Reported-by: Jaroslav Kysela <perex@perex.cz>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200625191308.3322-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Use reserved memory as PCM mode need more memory.
Use ping-pong buffer for PCM LPA playback to avoid underrun.
Signed-off-by: Bing Song <bing.song@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
FW is locate on OCRAM_A when lpa. core decoder can't put in reserved
memory as HIFI4 has 1G memory limitation. Disable DSP wrapper decoder
when lpa. lpa PCM mode will use SW decoder.
Signed-off-by: Bing Song <bing.song@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Input buffer maybe return when suspending. Interrupt suspend if
MU from DSP to ARM core.
Signed-off-by: Bing Song <bing.song@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Enlarge the buffer size to 128K. So ARM can sleep longer time for
MP3 LPA playback.
Signed-off-by: Bing Song <bing.song@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
[ Upstream commit 7fcd9bb5acd01250bcae1ecc0cb8b8d4bb5b7e63 ]
When the ASoC card registration fails and the codec component driver
never probes, the codec device is not initialized and therefore
memory for codec->wcaps is not allocated. This results in a NULL pointer
dereference when the codec driver suspend callback is invoked during
system suspend. Fix this by returning without performing any actions
during codec suspend/resume if the card was not registered successfully.
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200728231011.1454066-1-ranjani.sridharan@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 80982c7e834e5d4e325b6ce33757012ecafdf0bb upstream.
Some ioctls via OSS sequencer API may race and lead to UAF when the
port create and delete are performed concurrently, as spotted by a
couple of syzkaller cases. This patch is an attempt to address it by
serializing the ioctls with the existing register_mutex.
Basically OSS sequencer API is an obsoleted interface and was designed
without much consideration of the concurrency. There are very few
applications with it, and the concurrent performance isn't asked,
hence this "big hammer" approach should be good enough.
Reported-by: syzbot+1a54a94bd32716796edd@syzkaller.appspotmail.com
Reported-by: syzbot+9d2abfef257f3e2d4713@syzkaller.appspotmail.com
Suggested-by: Hillf Danton <hdanton@sina.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200804185815.2453-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 7fe3530427e52dd53cd7366914864e29215180a4 upstream.
The ca0113 command had the wrong group_id, 0x48 when it should've been
0x30. The front microphone selection should now work.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-3-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a00dc409de455b64e6cb2f6d40cdb8237cdb2e83 upstream.
When the ZxR headphone gain control was added, the ca0132_switch_get
function was not updated, which meant that the changes to the control
state were not saved when entering/exiting alsamixer.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-1-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit cc5edb1bd3f7bfe450f767b12423f6673822427b upstream.
Add a new quirk ID for the Recon3D, as tested by me.
Signed-off-by: Connor McAdams <conmanx360@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200803002928.8638-2-conmanx360@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f1ec5be17b9aafbc5f573da023850566b43d8e5e upstream.
There are several Loongson-3 based laptops produced by CZC or Lemote,
they use alc269/alc662 codecs and need specific pin-tables, this patch
add their pin-tables.
Signed-off-by: Huacai Chen <chenhc@lemote.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1596360400-32425-1-git-send-email-chenhc@lemote.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 07c9983b567d0ef33aefc063299de95a987e12a8 upstream.
This reverts commit 9a6418487b56 ("ALSA: hda: call runtime_allow()
for all hda controllers").
The reverted patch already introduced some regressions on some
machines:
- on gemini-lake machines, the error of "azx_get_response timeout"
happens in the hda driver.
- on the machines with alc662 codec, the audio jack detection doesn't
work anymore.
Fixes: 9a6418487b56 ("ALSA: hda: call runtime_allow() for all hda controllers")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208511
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200803064638.6139-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c2c3657f0aedb8736a0fb7b2b1985adfb86e7802 upstream.
It's been reported that, when neither nouveau nor Nvidia graphics
driver is used, the screen starts flickering. And, after comparing
between the working case (stable 4.4.x) and the broken case, it turned
out that the problem comes from the audio component binding. The
Nvidia and AMD audio binding code clears the bus->keep_power flag
whenever snd_hdac_acomp_init() succeeds. But this doesn't mean that
the component is actually bound, but it merely indicates that it's
ready for binding. So, when both nouveau and Nvidia are blacklisted
or not ready, the driver keeps running without the audio component but
also with bus->keep_power = false. This made the driver runtime PM
kicked in and powering down when unused, which results in flickering
in the graphics side, as it seems.
For fixing the bug, this patch moves the bus->keep_power flag change
into generic_acomp_notifier_set() that is the function called from the
master_bind callback of component ops; i.e. it's guaranteed that the
binding succeeded.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208609
Fixes: 5a858e79c911 ("ALSA: hda - Disable audio component for legacy Nvidia HDMI codecs")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200728082033.23933-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 5649625344fe1f4695eace7c37d011e317bf66d5 upstream.
HP NB right speaker had no sound output.
This platform was connected to I2S Amp for speaker out.(None Realtek I2S Amp IC)
EC need to check codec GPIO1 pin to initial I2S Amp.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/01285f623ac7447187482fb4a8ecaa7c@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 293a92c1d9913248b9987b68f3a5d6d2f0aae62b upstream.
This patch fixes a small typo I accidently submitted with the initial patch. The board should be named GA401 not G401.
Fixes: ff53664daff2 ("ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289")
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200724140837.302763-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4b43d05a1978a93a19374c6e6b817c9c1ff4ba4b upstream.
This patch adds support for headset mic to the ASUS ROG Zephyrus
G15(GA502) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200724140616.298892-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 3da87ec67a491b9633a82045896c076b794bf938 upstream.
As expected, this requires the same quirk as the SSL2+ in order for the
clock to sync. This was suggested by, and tested on an SSL2, by Dmitry.
Suggested-by: Dmitry <dpavlushko@gmail.com>
Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200621075005.52mjjfc6dtdjnr3h@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
For PCM format the supported format is mono/stereo, S16_LE/S32_LE,
and 8KHz-192kHz frequency.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
commit 8edac489e7c3fce44208373bb3e7b5835a672c66 upstream.
we need to free all allocated tlvs, not just the one allocated in
the loop before releasing kcontrols - other the tlvs references will
leak.
Fixes: 9f90af3a99 ('ASoC: topology: Consolidate and fix asoc_tplg_dapm_widget_*_create flow')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707203749.113883-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6f0307df83f2aa6bdf656c2219c89ce96502d20e upstream.
When errors happens while loading graph components, the kernel oopses
while trying to remove all topology components. This can be
root-caused to a list pointing to memory that was already freed on
error.
remove_route() is already called on errors and will perform the
required cleanups so there's no need to free the route memory in
soc_tplg_dapm_graph_elems_load() if the route was added to the
list. We do however want to free the routes allocated but not added to
the list.
Fixes: 7df04ea7a3 ('ASoC: topology: modify dapm route loading routine and add dapm route unloading')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200707203749.113883-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b6aa06de7757667bac88997a8807b143b8436035 upstream.
When building on allyesconfig kernel for a NO_DMA=y platform (e.g.
Sun-3), CONFIG_SND_SOC_QCOM_COMMON=y, but CONFIG_SND_SOC_QDSP6_AFE=n,
leading to a link failure:
sound/soc/qcom/common.o: In function `qcom_snd_parse_of':
common.c:(.text+0x2e2): undefined reference to `q6afe_is_rx_port'
While SND_SOC_QDSP6 depends on HAS_DMA, SND_SOC_MSM8996 and SND_SOC_SDM845
don't, so the following warning is seen:
WARNING: unmet direct dependencies detected for SND_SOC_QDSP6
Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && HAS_DMA [=n]
Selected by [y]:
- SND_SOC_MSM8996 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y]
- SND_SOC_SDM845 [=y] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && QCOM_APR [=y] && CROS_EC [=y] && I2C [=y] && SOUNDWIRE [=y]
Until recently, this warning was harmless (from a compile-testing
point-of-view), but the new user of q6afe_is_rx_port() turned this into
a hard failure.
As the QDSP6 driver itself builds fine if NO_DMA=y, and it depends on
QCOM_APR (which in turns depends on ARCH_QCOM || COMPILE_TEST), it is
safe to increase compile testing coverage. Hence fix the link failure
by dropping the HAS_DMA dependency of SND_SOC_QDSP6.
Fixes: a2120089251f1fe2 ("ASoC: qcom: common: set correct directions for dailinks")
Fixes: 6b1687bf76 ("ASoC: qcom: add sdm845 sound card support")
Fixes: a6f933f63f ("ASoC: qcom: apq8096: Add db820c machine driver")
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Link: https://lore.kernel.org/r/20200629122443.21736-1-geert@linux-m68k.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 85ca6b17e2bb96b19caac3b02c003d670b66de96 upstream.
The Lenovo Miix 2 10 has a keyboard dock with extra speakers in the dock.
Rather then the ACL5672's GPIO1 pin being used as IRQ to the CPU, it is
actually used to enable the amplifier for these speakers
(the IRQ to the CPU comes directly from the jack-detect switch).
Add a quirk for having an ext speaker-amplifier enable pin on GPIO1
and replace the Lenovo Miix 2 10's dmi_system_id table entry's wrong
GPIO_DEV quirk (which needs to be renamed to GPIO1_IS_IRQ) with the
new RT5670_GPIO1_IS_EXT_SPK_EN quirk, so that we enable the external
speaker-amplifier as necessary.
Also update the ident field for the dmi_system_id table entry, the
Miix models are not Thinkpads.
Fixes: 67e03ff3f3 ("ASoC: codecs: rt5670: add Thinkpad Tablet 10 quirk")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1786723
Link: https://lore.kernel.org/r/20200628155231.71089-4-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit b3df80ab6d147d4738be242e1c91e5fdbb6b03ef ]
snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error
path. Add the missed function call to fix it.
Fixes: ba49cf6f8e ("ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect")
Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com>
Reviewed-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200714080918.148196-1-jingxiangfeng@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 5cacc6f5764e94fa753b2c1f5f7f1f3f74286e82 upstream.
The RT5670_PWR_ANLG1 register has 3 bits to select the LDO voltage,
so the correct mask is 0x7 not 0x3.
Because of this wrong mask we were programming the ldo bits
to a setting of binary 001 (0x05 & 0x03) instead of binary 101
when moving to SND_SOC_BIAS_PREPARE.
According to the datasheet 001 is a reserved value, so no idea
what it did, since the driver was working fine before I guess we
got lucky and it does something which is ok.
Fixes: 5e8351de74 ("ASoC: add RT5670 CODEC driver")
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200628155231.71089-3-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 60379ba08532eca861e933b389526a4dc89e0c42 upstream.
snd_info_get_line() has a sanity check of NULL buffer -- both buffer
itself being NULL and buffer->buffer being NULL. Basically both
checks are valid and necessary, but the problem is that it's with
snd_BUG_ON() macro that triggers WARN_ON(). The latter condition
(NULL buffer->buffer) can be met arbitrarily by user since the buffer
is allocated at the first write, so it means that user can trigger
WARN_ON() at will.
This patch addresses it by simply moving buffer->buffer NULL check out
of snd_BUG_ON() so that spurious WARNING is no longer triggered.
Reported-by: syzbot+e42d0746c3c3699b6061@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200717084023.5928-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 568e4e82128aac2c62c2c359ebebb6007fd794f9 upstream.
Fixed no headphone sound bug on laptop Samsung Notebook Pen S
(950SBE-951SBE), by using existing patch in Linus' tree, commit
14425f1f521f (ALSA: hda/realtek: Add quirk for Samsung Notebook).
This laptop uses the same ALC298 but different subsystem id 0x144dc812.
I added SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c
Signed-off-by: Joonho Wohn <doomsheart@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAHcbMh291aWDKiWSZoxXB4-Eru6OYRwGA4AVEdCZeYmVLo5ZxQ@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 56275036d8185f92eceac7479d48b858ee3dab84 ]
When HDMI PCM devices are opened in a specific order, with at least one
HDMI/DP receiver connected, ALSA PCM open fails to -EBUSY on the
connected monitor, on recent Intel platforms (ICL/JSL and newer). While
this is not a typical sequence, at least Pulseaudio does this every time
when it is started, to discover the available PCMs.
The rootcause is an invalid assumption in hdmi_add_pin(), where the
total number of converters is assumed to be known at the time the
function is called. On older Intel platforms this held true, but after
ICL/JSL, the order how pins and converters are in the subnode list as
returned by snd_hda_get_sub_nodes(), was changed. As a result,
information for some converters was not stored to per_pin->mux_nids.
And this means some pins cannot be connected to all converters, and
application instead gets -EBUSY instead at open.
The assumption that converters are always before pins in the subnode
list, is not really a valid one. Fix the problem in hdmi_parse_codec()
by introducing separate loops for discovering converters and pins.
BugLink: https://github.com/thesofproject/linux/issues/1978
BugLink: https://github.com/thesofproject/linux/issues/2216
BugLink: https://github.com/thesofproject/linux/issues/2217
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200703153818.2808592-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
In SPDIF full duplex mode in order to get both RX and TX working we
need to start RX first, and TX second, fix this by porting SPDIF RX
configuration code from XCVR FW into Kernel driver and avoid using
M0+ core for SPDIF configuration.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
commit 158ae2f5e6fead30be1f0c203037f5556871513b upstream.
ASUS UX563 speaker can't output.
Add quirk to link suitable model will enable it.
This model also could enable headset Mic.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/96dee3ab01a04c28a7b44061e88009dd@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 473fbe13fd6f9082e413aea37e624ecbce5463cc upstream.
ASUS UX533 and UX534 speaker still can't output.
End User feedback speaker didn't have output.
Add this COEF value will enable it.
Fixes: 4e05110673 ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/80334402a93b48e385f8f4841b59ae09@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit f50a121d2f32bccc1d6b94df925a1ce44ea7eff7 upstream.
The Acer TravelMate B311R-31 laptop's audio (1025:1430) with ALC256
cannot detect the headset microphone until
ALC256_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x19 as the headset
mic pin.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200713060421.62435-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ff53664daff2a65f4bf2479ac56dfb3e908deff0 upstream.
This patch adds support for headset mic to the ASUS ROG Zephyrus
G14(GA401) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.
Signed-off-by: Armas Spann <zappel@retarded.farm>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200711110557.18681-1-zappel@retarded.farm
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ef9ddb9dc4f8b1da3b975918cd1fd98ec055b918 upstream.
ASUS platform couldn't need to use Headset Mode model.
It changes to the suitable model.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/d05bcff170784ec7bb35023407148161@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9b7e5208a941e2e491a83eb5fa83d889e888fa2f upstream.
USB MIDI driver has an error recovery mechanism to resubmit the URB in
the delayed timer handler, and this may race with the standard start /
stop operations. Although both start and stop operations themselves
don't race with each other due to the umidi->mutex protection, but
this isn't applied to the timer handler.
For fixing this potential race, the following changes are applied:
- Since the timer handler can't use the mutex, we apply the
umidi->disc_lock protection at each input stream URB submission;
this also needs to change the GFP flag to GFP_ATOMIC
- Add a check of the URB refcount and skip if already submitted
- Move the timer cancel call at disconnection to the beginning of the
procedure; this assures the in-flight timer handler is gone properly
before killing all pending URBs
Reported-by: syzbot+0f4ecfe6a2c322c81728@syzkaller.appspotmail.com
Reported-by: syzbot+5f1d24c49c1d2c427497@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200710160656.16819-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 68359a1ad8447c99732ebeab8c169bfed543667a upstream.
Recently syzkaller reported a UAF in LINE6 driver, and it's likely
because we call cancel_delayed_work() at the disconnect callback
instead of cancel_delayed_work_sync(). Let's use the correct one
instead.
Reported-by: syzbot+145012a46658ac00fc9e@syzkaller.appspotmail.com
Suggested-by: Alan Stern <stern@rowland.harvard.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hlfjr4gio.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6e8a914ad619042c5f25a4feb663357c4170fd8d upstream.
LINE6 drivers create stream URBs with a fixed pipe without checking
its validity, and this may lead to a kernel WARNING at the submission
when a malformed USB descriptor is passed.
For avoiding the kernel warning, perform the similar sanity checks for
each pipe type at creating a URB.
Reported-by: syzbot+c190f6858a04ea7fbc52@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/s5hv9iv4hq8.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 73094608b8e214952444fb104651704c98a37aeb ]
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:0x16ea) uses two interfaces, but only the second
interface contains the capture stream. This patch delays the
registration until the second interface appears.
Signed-off-by: Christoffer Nielsen <cn@obviux.dk>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAOtG2YHOM3zy+ed9KS-J4HkZo_QGzcUG9MigSp4e4_-13r6B=Q@mail.gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6f4ea2074ddf689ac6f892afa58515032dabf2e4 ]
Force it to use asynchronous playback.
Same quirk has already been added for Focusrite Scarlett Solo (2nd gen)
with a commit 46f5710f0b ("ALSA: usb-audio: Add quirk for Focusrite
Scarlett Solo").
This also seems to prevent regular clicks when playing at 44100Hz
on Scarlett 2i2 (2nd gen). I did not notice any side effects.
Moved both quirks to snd_usb_audioformat_attributes_quirk() as suggested.
Signed-off-by: Gregor Pintar <grpintar@gmail.com>
Reviewed-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200420214030.2361-1-grpintar@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fd60e0683e8e9107e09cd2e4798f3e27e85d2705 ]
Similar to the Kingston HyperX AMP, the Kingston HyperX Cloud
Alpha S (0951:16d8) uses two interfaces, but only the second
interface contains the capture stream. This patch delays the
registration until the second interface appears.
Signed-off-by: Emmanuel Pescosta <emmanuelpescosta099@gmail.com>
Link: https://lore.kernel.org/r/20200404153843.9288-1-emmanuelpescosta099@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit d8695bc5b1fe88305396b1f788d3b5f218e28a30 ]
A slight refactoring of the registration quirk code. Now it uses the
table lookup for easy additions in future. Also the return type was
changed to bool, and got a few more comments.
Link: https://lore.kernel.org/r/20200325103322.2508-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 55f7326170d9e83e2d828591938e1101982a679c ]
Create a quirk that allows special processing and/or
skipping the call to snd_card_register.
For HyperX AMP, which uses two interfaces, but only has
a capture stream in the second, this allows the capture
stream to merge with the first PCM.
Signed-off-by: Chris Wulff <crwulff@gmail.com>
Link: https://lore.kernel.org/r/20200314165449.4086-3-crwulff@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 2edb84e3047b93da2f2b234219cdc304df042d9e ]
MicroBook IIc operates in UAC2 mode by default. This patch addresses
several issues with it:
- MicroBook II and IIc shares the same USB ID. We can distinguish them
by interface class.
- MaxPacketsOnly attribute is erroneously set in endpoint descriptors.
As a result this card produces noise with all sample rates other than
96 KHz. This also causes issues like IOMMU page faults and other
problems with host controller.
- Sample rate changes takes more than 2 seconds for this device. Clock
validity request returns false during that period, so the clock validity
quirk is required.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200229151815.14199-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
The Channel Status bits must be reversed.
Fixes: 86401d592b ("MLK-23725-1: ASoC: fsl_xcvr: reimplement CS read function")
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Add fixed sai mclk for PDM software decimation improve
libimxswpdm conversion algorithm performance for
voice applications by using exact division ratios for
clock configurations mclk, bitclk.
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
There is no need for S24_LE support on TX since IEC958 ALSA
plugin is now configured to cover this function.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
commit 781c90c034d994c6a4e2badf189128a95ed864c2 upstream.
The Acer Veriton N4660G desktop's audio (1025:1248) with ALC269VC cannot
detect the headset microphone until ALC269VC_FIXUP_ACER_MIC_NO_PRESENCE
quirk maps the NID 0x18 as the headset mic pin.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200706071826.39726-3-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6e15d1261d522d1d222f8f89b23c6966905e9049 upstream.
The Acer Aspire C20-820 AIO's audio (1025:1065) with ALC269VC can't
detect the headset microphone until ALC269VC_FIXUP_ACER_HEADSET_MIC
quirk maps the NID 0x18 as the headset mic pin.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200706071826.39726-2-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8eae7e9b3967f08efaa4d70403aec513cbe45ad0 upstream.
The Acer desktop vCopperbox with ALC269VC cannot detect the MIC of
headset, the line out and internal speaker until
ALC269VC_FIXUP_ACER_VCOPPERBOX_PINS quirk applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200706071826.39726-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9774dc218bb628974dcbc76412f970e9258e5f27 upstream.
1)
In snd_hda_pick_fixup(), quirks are first matched by PCI SSID and then, if
there is no match, by codec SSID. The Lenovo "ThinkPad X1 Carbon 7th" has
an audio chip with PCI SSID 0x2292 and codec SSID 0x2293[1]. Therefore, fix
the quirk meant for that device to match on .subdevice == 0x2292.
2)
The "Thinkpad X1 Yoga 7th" does not exist. The companion product to the
Carbon 7th is the Yoga 4th. That device has an audio chip with PCI SSID
0x2292 and codec SSID 0x2292[2]. Given the behavior of
snd_hda_pick_fixup(), it is not possible to have a separate quirk for the
Yoga based on SSID. Therefore, merge the quirks meant for the Carbon and
Yoga. This preserves the current behavior for the Yoga.
[1] This is the case on my own machine and can also be checked here
https://github.com/linuxhw/LsPCI/tree/master/Notebook/Lenovo/ThinkPadhttps://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3225701
[2]
https://github.com/linuxhw/LsPCI/tree/master/Convertible/Lenovo/ThinkPadhttps://gist.github.com/hamidzr/dd81e429dc86f4327ded7a2030e7d7d9#gistcomment-3176355
Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen")
Fixes: 54a6a7dc107d ("ALSA: hda/realtek - Add quirk for the bass speaker on Lenovo Yoga X1 7th gen")
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Kailang Yang <kailang@realtek.com>
Tested-by: Vincent Bernat <vincent@bernat.ch>
Tested-by: Even Brenden <evenbrenden@gmail.com>
Signed-off-by: Benjamin Poirier <benjamin.poirier@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200703080005.8942-2-benjamin.poirier@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e337bf19f6af38d5c3fa6d06cd594e0f890ca1ac upstream.
These devices claim to be 96kHz mono, but actually are 48kHz stereo with
swapped channels and unaligned transfers.
Cc: stable@vger.kernel.org
Signed-off-by: Hector Martin <marcan@marcan.st>
Link: https://lore.kernel.org/r/20200702071433.237843-1-marcan@marcan.st
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 6a6ca7881b1ab1c13fe0d70bae29211a65dd90de upstream.
We have a Dell AIO, there is neither internal speaker nor internal
mic, only a multi-function audio jack on it.
Users reported that after freshly installing the OS and plug
a headset to the audio jack, the headset can't output sound. I
reproduced this bug, at that moment, the Input Source is as below:
Simple mixer control 'Input Source',0
Capabilities: cenum
Items: 'Headphone Mic' 'Headset Mic'
Item0: 'Headphone Mic'
That is because the patch_realtek will set this audio jack as mic_in
mode if Input Source's value is hp_mic.
If it is not fresh installing, this issue will not happen since the
systemd will run alsactl restore -f /var/lib/alsa/asound.state, this
will set the 'Input Source' according to history value.
If there is internal speaker or internal mic, this issue will not
happen since there is valid sink/source in the pulseaudio, the PA will
set the 'Input Source' according to active_port.
To fix this issue, change the parser function to let the hs_mic be
stored ahead of hp_mic.
Cc: stable@vger.kernel.org
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200625083833.11264-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit ad155712bb1ea2151944cf06a0e08c315c70c1e3 upstream.
The stack object “info” in snd_opl3_ioctl() has a leaking problem.
It has 2 padding bytes which are not initialized and leaked via
“copy_to_user”.
Signed-off-by: xidongwang <wangxidong_97@163.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1594006058-30362-1-git-send-email-wangxidong_97@163.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit f79a732a8325dfbd570d87f1435019d7e5501c6d ]
On partial_drain completion we should be in SNDRV_PCM_STATE_RUNNING
state, so set that for partially draining streams in
snd_compr_drain_notify() and use a flag for partially draining streams
While at it, add locks for stream state change in
snd_compr_drain_notify() as well.
Fixes: f44f2a5417 ("ALSA: compress: fix drain calls blocking other compress functions (v6)")
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Tested-by: Charles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200629134737.105993-4-vkoul@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Add dsp mu interrupt into wakeup source when LPA mode.
So the interrupt will wakeup A core when DSP consumed
all audio bitstream data.
Signed-off-by: Bing Song <bing.song@nxp.com>
The buffer is share between A core and DSP. Enlarge it, so A core will
suspend longer when DSP is playbacking audio.
Signed-off-by: Bing Song <bing.song@nxp.com>
One big buffer shared between A core and DSP to buffer audio
bitstram. So A core can sleep more time when DSP is playing
buffered audio bit stream.
Signed-off-by: Bing Song <bing.song@nxp.com>
IMX8MP_CLK_AUDIO_ROOT and IMX8MP_CLK_AUDIO_AXI_DIV will gated when
suspend even if dsp driver ignored suspend when low power audio offload
playback. The root cause is the two clocks is controlled when audiomix
power on/off. Add the two clocks in dsp driver, so driver will add
one ref count when useing dsp, so the two clocks don't be gated when
A core suspend during low power audio playback.
Signed-off-by: Bing Song <bing.song@nxp.com>
This reverts commit aba41867dd which is
commit f0bd62b64016508938df9babe47f65c2c727d25c upstream.
It causes a number of reported issues and a fix for it has not hit
Linus's tree yet. Revert this to resolve those problems.
Cc: Alexander Tsoy <alexander@tsoy.me>
Cc: Takashi Iwai <tiwai@suse.de>
Cc: Sasha Levin <sashal@kernel.org>
Cc: Hans de Goede <jwrdegoede@fedoraproject.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Depending on sample rate the PDM must be clocked at either 24.576MHz or
22.5792MHz. CLK_DIV is later calculated as function of PDM root clock
frequency and FS. Setting PDM root clock = FS * 1024 is wrong because
for low sample rate such as 8000 the PDM root clock will be 8000 * 1024
= 8192000 Hz so PDM will be underclocked for this sample rate.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
commit b2c22910fe5aae10b7e17b0721e63a3edf0c9553 upstream.
There are two more HP systems control mute LED from HDA codec and need
to expose micmute led class so SoF can control micmute LED.
Add quirks to support them.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200617102906.16156-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a0b03952a797591d4b6d6fa7b9b7872e27783729 upstream.
MSI GE63 laptop with ALC1220 codec requires the very same quirk
(ALC1220_FIXUP_CLEVO_P950) as other MSI devices for the proper sound
output.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=208057
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200616132150.8778-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit adb36a8203831e40494a92095dacd566b2ad4a69 upstream.
These IDs are for upcoming NVIDIA chips with audio functions that are largely
similar to the existing ones.
Signed-off-by: Aaron Plattner <aplattner@nvidia.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200611180845.39942-1-aplattner@nvidia.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit f141a422159a199f4c8dedb7e0df55b3b2cf16cd ]
Calling pm_runtime_get_sync increments the counter even in case of
failure, causing incorrect ref count if pm_runtime_put is not called in
error handling paths. Call pm_runtime_put if pm_runtime_get_sync fails.
Fixes: fc05a5b222 ("ASoC: rockchip: add support for pdm controller")
Signed-off-by: Qiushi Wu <wu000273@umn.edu>
Reviewed-by: Heiko Stuebner <heiko@sntech.de>
Link: https://lore.kernel.org/r/20200613205158.27296-1-wu000273@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ed1220df6e666500ebf58c4f2fccc681941646fb ]
For mono channel, SSI will switch to Normal mode.
In Normal mode and Network mode, the Word Length Control bits
control the word length divider in clock generator, which is
different with I2S Master mode (the word length is fixed to
32bit), it should be the value of params_width(hw_params).
The condition "slots == 2" is not good for I2S Master mode,
because for Network mode and Normal mode, the slots can also
be 2. Then we need to use (ssi->i2s_net & SSI_SCR_I2S_MODE_MASK)
to check if it is I2S Master mode.
So we refine the formula for mono channel, otherwise there
will be sound issue for S24_LE.
Fixes: b0a7043d5c ("ASoC: fsl_ssi: Caculate bit clock rate using slot number and width")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/034eff1435ff6ce300b6c781130cefd9db22ab9a.1592276147.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a2120089251f1fe221305e88df99af16f940e236 ]
Currently both FE and BE dai-links are configured bi-directional,
However the DSP BE dais are only single directional,
so set the directions as supported by the BE dais.
Fixes: c25e295cd7 (ASoC: qcom: Add support to parse common audio device nodes)
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Tested-by: John Stultz <john.stultz@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200612123711.29130-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4a95737440d426e93441d49d11abf4c6526d4666 ]
This patch adds support to q6afe_is_rx_port() to get direction
of DSP BE dai port, this is useful for setting dailink
directions correctly.
Fixes: c25e295cd7 (ASoC: qcom: Add support to parse common audio device nodes)
Reported-by: John Stultz <john.stultz@linaro.org>
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20200612123711.29130-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 6476b60f32866be49d05e2e0163f337374c55b06 ]
Successful send of EOS command does not indicate that EOS is actually
finished, correct event to wait EOS is finished is EOS_RENDERED event.
EOS_RENDERED means that the DSP has finished processing all the buffers
for that particular session and stream.
This patch fixes EOS handling!
Fixes: 68fd8480bb ("ASoC: qdsp6: q6asm: Add support to audio stream apis")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200611124159.20742-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 220345e98f1cdc768eeb6e3364a0fa7ab9647fe7 upstream.
The USB-audio mixer code holds a linked list of usb_mixer_elem_list,
and several operations are performed for each mixer element. A few of
them (snd_usb_mixer_notify_id() and snd_usb_mixer_interrupt_v2())
assume each mixer element being a usb_mixer_elem_info object that is a
subclass of usb_mixer_elem_list, cast via container_of() and access it
members. This may result in an out-of-bound access when a
non-standard list element has been added, as spotted by syzkaller
recently.
This patch adds a new field, is_std_info, in usb_mixer_elem_list to
indicate that the element is the usb_mixer_elem_info type or not, and
skip the access to such an element if needed.
Reported-by: syzbot+fb14314433463ad51625@syzkaller.appspotmail.com
Reported-by: syzbot+2405ca3401e943c538b5@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200624122340.9615-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit a32a1fc99807244d920d274adc46ba04b538cc8a upstream.
We've found Samsung USBC Headset (AKG) (VID: 0x04e8, PID: 0xa051)
need a tiny delay after each class compliant request.
Otherwise the device might not be able to be recognized each times.
Signed-off-by: Chihhao Chen <chihhao.chen@mediatek.com>
Signed-off-by: Macpaul Lin <macpaul.lin@mediatek.com>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/1592910203-24035-1-git-send-email-macpaul.lin@mediatek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e7585db1b0b5b4e4eb1967bb1472df308f7ffcbf upstream.
This uses the same quirk as the Motu M2 and M4 to ensure the driver uses the
audio interface's clock. Tested on an SSL2+.
Signed-off-by: Laurence Tratt <laurie@tratt.net>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200612111807.dgnig6rwhmsl2bod@overdrive.tratt.net
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit ff58bbc7b9704a5869204176f804eff57307fef0 ]
With the recent full-duplex support of implicit feedback streams, an
endpoint can be still running after closing the capture stream as long
as the playback stream with the sync-endpoint is running. In such a
state, the URBs are still be handled and they may call retire_data_urb
callback, which tries to transfer the data from the PCM buffer. Since
the PCM stream gets closed, this may lead to use-after-free.
This patch adds the proper clearance of the callback at stopping the
capture stream for addressing the possible UAF above.
Fixes: 10ce77e4817f ("ALSA: usb-audio: Add duplex sound support for USB devices using implicit feedback")
Link: https://lore.kernel.org/r/20200616120921.12249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 79d4f823a06796656289f97b922493da5690e46c ]
The Asus T101HA uses the default jack-detect mode 3, but instead of
using an analog microphone it is using a DMIC on dmic-data-pin 1,
like the Asus T100HA. Note unlike the T100HA its jack-detect is not
inverted.
Add a DMI quirk with the correct settings for this model.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200608204634.93407-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 199a5e8fda54ab3c8c6f6bf980c004e97ebf5ccb ]
The Toshiba Encore WT10-A tablet almost fully works with the default
settings for Bay Trail CR devices. The only issue is that it uses a
digital mic. connected the the DMIC1 input instead of an analog mic.
Add a quirk for this model using the default settings with the input-map
replaced with BYT_RT5640_DMIC1_MAP.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200608204634.93407-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ba4e5abc6c4e173af7c941c03c067263b686665d ]
With additional checks on dailinks, we see errors such as
[ 3.000418] sof-nocodec sof-nocodec: CPU DAI DMIC01 Pin for rtd
NoCodec-6 does not support playback
It's not clear why we set the dpcm_playback and dpcm_capture flags
unconditionally, add a check on number of channels for each direction
to avoid invalid configurations.
Fixes: 8017b8fd37 ('ASoC: SOF: Add Nocodec machine driver support')
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200608194415.4663-5-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 607fa205a7e4dfad28b8a67ab1c985756ddbccb0 ]
Additional checks for valid DAIs expose a corner case, where existing
BE dailinks get modified, e.g. HDMI links are tagged with
dpcm_capture=1 even if the DAIs are for playback.
This patch makes those changes conditional and flags configuration
issues when a BE dailink is has no_pcm=0 but dpcm_playback or
dpcm_capture=1 (which makes no sense).
As discussed on the alsa-devel mailing list, there are redundant flags
for dpcm_playback, dpcm_capture, playback_only, capture_only. This
will have to be cleaned-up in a future update. For now only correct
and flag problematic configurations.
Fixes: 218fe9b7ec7f3 ("ASoC: soc-core: Set dpcm_playback / dpcm_capture")
Suggested-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Reviewed-by: Daniel Baluta <daniel.baluta@gmail.com>
Reviewed-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Link: https://lore.kernel.org/r/20200608194415.4663-3-pierre-louis.bossart@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 36124fb19f1ae68a500cd76a76d40c6e81bee346 ]
fsl_asrc_dma_hw_params() invokes dma_request_channel() or
fsl_asrc_get_dma_channel(), which returns a reference of the specified
dma_chan object to "pair->dma_chan[dir]" with increased refcnt.
The reference counting issue happens in one exception handling path of
fsl_asrc_dma_hw_params(). When config DMA channel failed for Back-End,
the function forgets to decrease the refcnt increased by
dma_request_channel() or fsl_asrc_get_dma_channel(), causing a refcnt
leak.
Fix this issue by calling dma_release_channel() when config DMA channel
failed.
Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Link: https://lore.kernel.org/r/1590415966-52416-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 25bf943e4e7b47282bd86ae7d39e039217ebb007 ]
Function "pm_runtime_get_sync()" is not handled by "pm_runtime_put()"
if "PTR_ERR(rst) == -EPROBE_DEFER". Fix this issue by adding
"pm_runtime_put()" into this error path.
Fixes: f65bb92ca1 ("ASoC: img-i2s-in: Add runtime PM")
Signed-off-by: Qiushi Wu <wu000273@umn.edu>
Link: https://lore.kernel.org/r/20200525055011.31925-1-wu000273@umn.edu
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0e0e10fde0e9808d1991268f5dca69fb36c025f7 ]
The Toshiba Encore WT8-A tablet almost fully works with the default
settings for non-CR Bay Trail devices. The only problem is that its
jack-detect switch is not inverted (it is active high instead of
the normal active low).
Add a quirk for this model using the default settings +
BYT_RT5640_JD_NOT_INV.
Signed-off-by: Hans de Goede <hdegoede@redhat.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200518072416.5348-1-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 10ce77e4817fef99e1166be7e6685a80c63bf77f ]
For USB sound devices using implicit feedback the endpoint used for
this feedback should be able to be opened twice, once for required
feedback and second time for audio data. This way these devices can be
put in duplex audio mode. Since this only works if the settings of the
endpoint don't change a check is included for this.
This fixes bug 207023 ("MOTU M2 regression on duplex audio") and
should also fix bug 103751 ("M-Audio Fast Track Ultra usb audio device
will not operate full-duplex")
Fixes: c249177944b6 ("ALSA: usb-audio: add implicit fb quirk for MOTU M Series")
Signed-off-by: Erwin Burema <e.burema@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207023
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=103751
Link: https://lore.kernel.org/r/2410739.SCZni40SNb@alpha-wolf
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 03990fd58d2b7c8f7d53e514ba9b8749fac260f9 ]
If an error occurs after the call to 'omap_mcbsp_init()', the reference to
'mcbsp->fclk' must be decremented, as already done in the remove function.
This can be achieved easily by using the devm_ variant of 'clk_get()'
when the reference is taken in 'omap_mcbsp_init()'
This fixes the leak in the probe and has the side effect to simplify both
the error handling path of 'omap_mcbsp_init()' and the remove function.
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Acked-by: Peter Ujfalusi <peter.ujflausi@ti.com>
Link: https://lore.kernel.org/r/20200512134325.252073-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4e8748fcaeec073e3ba794871ce86c545e4f961f ]
There are 2 issues here:
- if one of the 'of_parse_phandle' fails, calling 'mop500_of_node_put()'
is a no-op because the 'mop500_dai_links' structure has not been
initialized yet, so the referenced are not decremented
- The reference stored in 'mop500_dai_links[i].codecs' is refcounted
only once in the probe and must be decremented only once.
Fixes: 39013bd60e ("ASoC: Ux500: Dispose of device nodes correctly")
Signed-off-by: Christophe JAILLET <christophe.jaillet@wanadoo.fr>
Link: https://lore.kernel.org/r/20200512100705.246349-1-christophe.jaillet@wanadoo.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 66c705d07d784fb6b4622c6e47b6acae357472db ]
SSI BUSIF buffer is possible to overflow or underflow, especially in a
hypervisor environment. If there is no interrupt support, it will eventually
lead to errors in pcm data.
This patch adds overflow and underflow interrupt support for SSI BUSIF buffer.
Reported-by: Chen Li <licheng0822@thundersoft.com>
Signed-off-by: Yongbo Zhang <giraffesnn123@gmail.com>
Tested-by: Chen Li <licheng0822@thundersoft.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200512093003.28332-1-giraffesnn123@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7d8785bc7adbb4dc5ba8ee06994107637848ded8 ]
Fix to return negative error code -ENOMEM from the IPC init error
handling case instead of 0, as done elsewhere in this function.
Fixes: c16211d622 ("ASoC: SOF: Add Sound Open Firmware driver core")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: Wei Yongjun <weiyongjun1@huawei.com>
Link: https://lore.kernel.org/r/20200509093337.78897-1-weiyongjun1@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8304cf77c92038cd1c50c27b69d30be695cc8003 ]
Although the value of FDF is used just for outgoing stream, the assignment
to union member is done for both directions of stream. At present this
causes no issue because the value of same position is reassigned later for
opposite stream. However, it's better to add if statement.
Fixes: d3d10a4a1b ("ALSA: firewire-lib: use union for directional parameters")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200508043635.349339-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1a446873d7dd3a450f685928ce7f1907bde4583d ]
During S3 test, the following error was observed:
[ 726.174237] i2c_designware i2c_designware.0: platform_pm_resume+0x0/0x3d returned 0 after 0 usecs
[ 726.184187] max98373 i2c-MX98373:00: calling max98373_resume+0x0/0x30 [snd_soc_max98373] @ 12698, parent: i2c-11
[ 726.195589] max98373 i2c-MX98373:00: Reset command failed. (ret:-16)
When calling regmap_update_bits(), since map->reg_update_bits is NULL,
_regmap_read() is entered with the following logic:
if (!map->cache_bypass) {
ret = regcache_read(map, reg, val);
if (ret == 0)
return 0;
}
if (map->cache_only)
return -EBUSY;
regcache_read() hits -EINVAL because MAX98373_R2000_SW_RESET is volatile,
as map->cache_only is set by codec suspend, thus -EBUSY is returned.
Fix by moving max98373_reset() after cache_only set to false in max98373_resume().
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Link: https://lore.kernel.org/r/1588376661-29799-1-git-send-email-yong.zhi@intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 5b6cc38f3f3f37109ce72b60bda215a5f6892c0b ]
The linked list entry from FIFO is peeked at
queue_pending_output_urbs() but the actual element pop-out is
performed outside the spinlock, and it's potentially racy.
Do delete the link at the right place inside the spinlock.
Fixes: 8fdff6a319 ("ALSA: snd-usb: implement new endpoint streaming model")
Link: https://lore.kernel.org/r/20200424074016.14301-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit f0bd62b64016508938df9babe47f65c2c727d25c ]
For computation of the the next frame size current value of fs/fps and
accumulated fractional parts of fs/fps are used, where values are stored
in Q16.16 format. This is quite natural for computing frame size for
asynchronous endpoints driven by explicit feedback, since in this case
fs/fps is a value provided by the feedback endpoint and it's already in
the Q format. If an error is accumulated over time, the device can
adjust fs/fps value to prevent buffer overruns/underruns.
But for synchronous endpoints the accuracy provided by these computations
is not enough. Due to accumulated error the driver periodically produces
frames with incorrect size (+/- 1 audio sample).
This patch fixes this issue by implementing a different algorithm for
frame size computation. It is based on accumulating of the remainders
from division fs/fps and it doesn't accumulate errors over time. This
new method is enabled for synchronous and adaptive playback endpoints.
Signed-off-by: Alexander Tsoy <alexander@tsoy.me>
Link: https://lore.kernel.org/r/20200424022449.14972-1-alexander@tsoy.me
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3b8a299a58b2afce464ae11324b59dcf0f1d10a7 ]
free_irq() is missing in case of error, fix that.
Signed-off-by: Pavel Machek (CIP) <pavel@denx.de>
Reviewed-by: Jerome Brunet <jbrunet@baylibre.com>
Link: https://lore.kernel.org/r/20200606153103.GA17905@amd
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit c26fde3b15ed41f5f452f1da727795f787833287 ]
This provides a better separation between runtime and PM sleep
callbacks.
Only do nothing if given runtime flag is set and calback is not set.
With the current implementation, if PM sleep callback is set but runtime
callback is not set then at runtime resume we reload the firmware even
if we do not support runtime resume callback.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Link: https://lore.kernel.org/r/20200515135958.17511-2-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7f0d5053c5a9d23fe5c2d337495a9d79038d267b ]
The "header->number" comes from the ioctl and it needs to be clamped to
prevent out of bounds writes.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Link: https://lore.kernel.org/r/20200501094011.GA960082@mwanda
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit dbd13179780555ecd3c992dea1222ca31920e892 ]
Currently mute LED and micmute LED share the same GPIO polarity.
So split the polarity for mute and micmute, in case they have different
polarities.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200430083255.5093-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a697ae6ea56e23397341b027098c1b11d9ab13da ]
davinci_mcasp_get_dma_type() invokes dma_request_chan(), which returns a
reference of the specified dma_chan object to "chan" with increased
refcnt.
When davinci_mcasp_get_dma_type() returns, local variable "chan" becomes
invalid, so the refcount should be decreased to keep refcount balanced.
The reference counting issue happens in one exception handling path of
davinci_mcasp_get_dma_type(). When chan device is NULL, the function
forgets to decrease the refcnt increased by dma_request_chan(), causing
a refcnt leak.
Fix this issue by calling dma_release_channel() when chan device is
NULL.
Signed-off-by: Xiyu Yang <xiyuyang19@fudan.edu.cn>
Signed-off-by: Xin Tan <tanxin.ctf@gmail.com>
Acked-by: Peter Ujfalusi <peter.ujfalusi@ti.com>
Link: https://lore.kernel.org/r/1587818916-38730-1-git-send-email-xiyuyang19@fudan.edu.cn
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1fecbb71fe0e46b886f84e3b6decca6643c3af6d ]
Disable exception interrupt before scheduling tasklet, otherwise if
the tasklet isn't handled immediately, there will be endless xrun
interrupt.
Fixes: 7ccafa2b38 ("ASoC: fsl_esai: recover the channel swap after xrun")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/a8f2ad955aac9e52587beedc1133b3efbe746895.1587968824.git.shengjiu.wang@nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fe17e6cdc0fefca96ba9659be4b2b07487cbf0c5 ]
when do randconfig like this:
CONFIG_SND_SOC_SOF_IMX8_SUPPORT=y
CONFIG_SND_SOC_SOF_IMX8=y
CONFIG_SND_SOC_SOF_OF=y
CONFIG_IMX_DSP=m
CONFIG_IMX_SCU=y
there is a link error:
sound/soc/sof/imx/imx8.o: In function 'imx8_send_msg':
imx8.c:(.text+0x380): undefined reference to 'imx_dsp_ring_doorbell'
Select IMX_DSP in SND_SOC_SOF_IMX8_SUPPORT to fix this
Fixes: f9ad75468453 ("ASoC: SOF: imx: fix reverse CONFIG_SND_SOC_SOF_OF dependency")
Reported-by: Hulk Robot <hulkci@huawei.com>
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Link: https://lore.kernel.org/r/20200409071832.2039-2-daniel.baluta@oss.nxp.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 3ef9d5073b552d56bd6daf2af1e89b7e8d4df183 ]
The microphone-jack state needs to be masked in a case of a 4-pin jack
when microphone and ground pins are shorted. Presence of nvidia,headset
tells that WM8903 CODEC driver should mask microphone's status if short
circuit is detected, i.e headphones are inserted.
Signed-off-by: Dmitry Osipenko <digetx@gmail.com>
Link: https://lore.kernel.org/r/20200330204011.18465-3-digetx@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Resume common flow (System PM / Runtime PM) is like this:
sof_resume
-> specific device resume
-> snd_sof_load_firmware
-> snd_sof_dsp_reset (1)
-> load_modules()
-> snd_sof_run_firmware (2)
We need to implement dsp_reset callback (1) that will actually reset the
DSP but keep it stalled.
In order to implement this we do the following:
-> put DSP into reset (assert CoreReset bit from PWRCTL)
-> stall the DSP using RunStall bit from AudioDSP_REG2 Audiomix
-> take DSP out of reset (de-assert CoreReset bit from PWRCTL)
At this moment the DSP is taken out of reset and Stalled! This means
that we can load the firmware and then start the DSP (2).
Until now we resetted the DSP by turning down the Audiomix PD. This
doesn't work for Runtime PM if another IP is keeping Audiomix PD up.
By introducing dsp_reset() we no longer rely on turning off the audiomix
to reset the DSP.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
This function is used to stall the DSP and it will be used
together with PWRCTL register in order to properly reset the
DSP to handle the Power Management use case.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Use a meaningful named macro to represent RunStall bit from
AudioDSP_REG2 register.
This register is used to Run/Stall the DSP.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
commit 0c5086f5699906ec8e31ea6509239489f060f2dc upstream.
The HP Thunderbolt Dock has two separate USB devices, one is for speaker
and one is for headset. Add names for them so userspace can apply UCM
settings.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200608062630.10806-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 862b2509d157c629dd26d7ac6c6cdbf043d332eb upstream.
When a USB-audio interface gets runtime-suspended via auto-pm feature,
the driver suspends all functionality and increment
chip->num_suspended_intf. Later on, when the system gets suspended to
S3, the driver increments chip->num_suspended_intf again, skips the
device changes, and sets the card power state to
SNDRV_CTL_POWER_D3hot. In return, when the system gets resumed from
S3, the resume callback decrements chip->num_suspended_intf. Since
this refcount is still not zero (it's been runtime-suspended), the
whole resume is skipped. But there is a small pitfall here.
The problem is that the driver doesn't restore the card power state
after this resume call, leaving it as SNDRV_CTL_POWER_D3hot. So,
even after the system resume finishes, the card instance still appears
as if it were system-suspended, and this confuses many ioctl accesses
that are blocked unexpectedly.
In details, we have two issues behind the scene: one is that the card
power state is changed only when the refcount becomes zero, and
another is that the prior auto-suspend check is kept in a boolean
flag. Although the latter problem is almost negligible since the
auto-pm feature is imposed only on the primary interface, but this can
be a potential problem on the devices with multiple interfaces.
This patch addresses those issues by the following:
- Replace chip->autosuspended boolean flag with chip->system_suspend
counter
- At the first system-suspend, chip->num_suspended_intf is recorded to
chip->system_suspend
- At system-resume, the card power state is restored when the
chip->num_suspended_intf refcount reaches to chip->system_suspend,
i.e. the state returns to the auto-suspended
Also, the patch fixes yet another hidden problem by the code
refactoring along with the fixes above: namely, when some resume
procedure failed, the driver left chip->num_suspended_intf that was
already decreased, and it might lead to the refcount unbalance.
In the new code, the refcount decrement is done after the whole resume
procedure, and the problem is avoided as well.
Fixes: 0662292aec ("ALSA: usb-audio: Handle normal and auto-suspend equally")
Reported-and-tested-by: Macpaul Lin <macpaul.lin@mediatek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200603153709.6293-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e18035cf5cb3d2bf8e4f4d350a23608bd208b934 upstream.
Add and use snd_pcm_stream_lock_nested() in snd_pcm_link/unlink
implementation. The code is fine, but generates a lockdep complaint:
============================================
WARNING: possible recursive locking detected
5.7.1mq+ #381 Tainted: G O
--------------------------------------------
pulseaudio/4180 is trying to acquire lock:
ffff888402d6f508 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xda8/0xee0 [snd_pcm]
but task is already holding lock:
ffff8883f7a8cf18 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xe4e/0xee0 [snd_pcm]
other info that might help us debug this:
Possible unsafe locking scenario:
CPU0
----
lock(&group->lock);
lock(&group->lock);
*** DEADLOCK ***
May be due to missing lock nesting notation
2 locks held by pulseaudio/4180:
#0: ffffffffa1a05190 (snd_pcm_link_rwsem){++++}-{3:3}, at: snd_pcm_common_ioctl+0xca0/0xee0 [snd_pcm]
#1: ffff8883f7a8cf18 (&group->lock){-...}-{2:2}, at: snd_pcm_common_ioctl+0xe4e/0xee0 [snd_pcm]
[...]
Cc: stable@vger.kernel.org
Fixes: f57f3df03a ("ALSA: pcm: More fine-grained PCM link locking")
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Link: https://lore.kernel.org/r/37252c65941e58473b1219ca9fab03d48f47e3e3.1591610330.git.mirq-linux@rere.qmqm.pl
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
commit 951e2736f4b11b58dc44d41964fa17c3527d882a upstream.
Prevent SNDRV_PCM_IOCTL_LINK linking stream to itself - the code
can't handle it. Fixed commit is not where bug was introduced, but
changes the context significantly.
Cc: stable@vger.kernel.org
Fixes: 0888c321de ("pcm_native: switch to fdget()/fdput()")
Signed-off-by: Michał Mirosław <mirq-linux@rere.qmqm.pl>
Link: https://lore.kernel.org/r/89c4a2487609a0ed6af3ecf01cc972bdc59a7a2d.1591634956.git.mirq-linux@rere.qmqm.pl
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 573fcbfd319ccef26caa3700320242accea7fd5c upstream.
A couple of Lenovo ThinkCentre machines all have 2 front mics and they
use the same codec alc623 and have the same pin config, so add a
pintbl entry for those machines to apply the fixup
ALC283_FIXUP_HEADSET_MIC.
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Link: https://lore.kernel.org/r/20200608115541.9531-1-hui.wang@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit bbd6aac3ae15bef762af03bf62e35ace5c4292bd upstream.
128000 and 192000 are congruence modulo 32000, thus it's wrong to
distinguish them as multiple of 32000 and 48000 by modulo 32000 at
first.
Additionally, used condition statement to detect quadruple speed can
cause missing bit flag.
Furthermore, counter to ensure the configuration is wrong and it
causes false positive.
This commit fixes the above three bugs.
Cc: <stable@vger.kernel.org>
Fixes: 60aec494b3 ("ALSA: fireface: support allocate_resources operation in latter protocol")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200510074301.116224-2-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 8ba4dc3cff8cbe2c571063a5fd7116e8bde563ca upstream.
The xmax values for Master Playback Volume and Mic Boost
Capture Volume are specified incorrectly (one greater)
which results in the wrong dB gain being shown to the user
in the case of Master Playback Volume.
Signed-off-by: Pavel Dobias <dobias@2n.cz>
Cc: stable@vger.kernel.org
Link: https://lore.kernel.org/r/20200515120757.24669-1-dobias@2n.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
Similar with commit a4c2646bc92 ("MLK-24263-5 ASoC: SOF: Add support for system / runtime PM")
this adds the system/runtime PM handlers.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
SOF boots the DSP at probe and keeps it up all the time.
With this change, after booting if no one is using the DSP
the SOF core will turn off the DSP to save power.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
SOF core will not call device specific System suspend
if the DSP is not booted (this is mostly because the way
PCI handles PM, which before system suspend it activates
the devices calling runtime resume).
So, in case we use OF we need to make sure that a simple
system suspend/resume cycle where the DSP is already off
will not turn on the DSP.
This part was completely re-written in the upstream kernel,
and this change is just a summary of the changes in upstream.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
We make use of common imx8m_suspend / imx8m_resume functions
for both system PM and runtime PM.
imx8m_suspend:
- frees the MU channels
- disables the clocks
imx8m_resume
- enables the clocks
- requests the MU channels
On i.MX8MP there is no dedicated functionality to put the DSP in reset.
The only way of doing this is to POWER DOWN the Audiomix domain.
We are able to do this because turning off the clocks and freeing the
channels makes the Audiomix to have no users thus PM kernel core turns
it down.
Now, care must be taken when handling system sleep. For this we
introduce a global variable to hold if the system was suspend or not via
system PM handlers.
SOF core will not call system PM suspend handler if the DSP is already down,
but at resume it will call the system PM resume. So, we need to keep track of
the state via the suspend variable.
Few insights on how SOF core handles the PM:
- SOF core uses PM runtime autosuspend (with a timeout of 2 secs)
- at probe, SOF core boots the DSP and lets the PM runtime suspend to
turn it off, if there is no activity
- when someone opens the ALSA sound card (aplay/arecord, etc) SOF core
calls PM runtime resume to turn on the DSP
- when the ALSA sound card is closed SOF core make use of PM subsystem
to call PM runtime suspend and thus turning off the DSP.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
This provides a better separation between runtime and PM sleep
callbacks.
Only do nothing if given runtime flag is set and calback is not set.
With the current implementation, if PM sleep callback is set but runtime
callback is not set then at runtime resume we reload the firmware even
if we do not support runtime resume callback.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
SPDIF TX fails to play after resume because PHY is put in reset on suspend,
and is not re-initialized on resume. Move SPDIF TX configuration in "prepare"
callback since "prepare" is called after resume.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
For the -EPROBE_DEFER case, it is not real error, and just resource is
not ready. It needs to use debug message instead of error message for it.
It could avoid unexpected console message for every device's probe, eg
SD/USB devices plug in.
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Peter Chen <peter.chen@nxp.com>
For the -EPROBE_DEFER case, it is not real error, and just resource is
not ready. It needs to use debug message instead of error message for it.
It could avoid unexpected console message for every device's probe, eg
SD/USB devices plug in.
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Peter Chen <peter.chen@nxp.com>
For the -EPROBE_DEFER case, it is not real error, and just resource is
not ready. It needs to use debug message instead of error message for it.
It could avoid unexpected console message for every device's probe, eg
SD/USB devices plug in, suspend/resume process, etc.
SD device plug in:
mmc1: new high speed SDHC card at address aaaa
mmcblk1: mmc1:aaaa SA16G 14.8 GiB
imx-cs42888 sound-cs42888: failed to find codec platform device
USB device plug in:
usb 2-1: USB disconnect, device number 2
usb 3-1: new SuperSpeed Gen 1 USB device number 2 using cdns-usb3
usb-storage 3-1:1.0: USB Mass Storage device detected
scsi host0: usb-storage 3-1:1.0
imx-cs42888 sound-cs42888: failed to find codec platform device
scsi 0:0:0:0: Direct-Access SanDisk Ultra 1.00 PQ: 0 ANSI: 6
sd 0:0:0:0: [sda] 60062500 512-byte logical blocks: (30.8 GB/28.6 GiB)
sd 0:0:0:0: [sda] Write Protect is off
sd 0:0:0:0: [sda] Write cache: disabled, read cache: enabled, doesn't support DPO or FUA
sda: sda1
sd 0:0:0:0: [sda] Attached SCSI removable disk
imx-cs42888 sound-cs42888: failed to find codec platform device
Suspend/resume process:
OOM killer disabled.
Freezing remaining freezable tasks ... (elapsed 0.083 seconds) done.
printk: Suspending console(s) (use no_console_suspend to debug)
fec 5b040000.ethernet eth0: Link is Down
PM: suspend devices took 0.684 seconds
Disabling non-boot CPUs ...
CPU1: shutdown
psci: CPU1 killed (polled 4 ms)
CPU2: shutdown
psci: CPU2 killed (polled 0 ms)
CPU3: shutdown
psci: CPU3 killed (polled 0 ms)
Enabling non-boot CPUs ...
Detected VIPT I-cache on CPU1
GICv3: CPU1: found redistributor 1 region 0:0x0000000051b20000
CPU1: Booted secondary processor 0x0000000001 [0x410fd042]
CPU1 is up
Detected VIPT I-cache on CPU2
GICv3: CPU2: found redistributor 2 region 0:0x0000000051b40000
CPU2: Booted secondary processor 0x0000000002 [0x410fd042]
CPU2 is up
Detected VIPT I-cache on CPU3
GICv3: CPU3: found redistributor 3 region 0:0x0000000051b60000
CPU3: Booted secondary processor 0x0000000003 [0x410fd042]
CPU3 is up
imx-cs42888 sound-cs42888: failed to find codec platform device
PM: resume devices took 0.240 seconds
OOM killer enabled.
Restarting tasks ...
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Peter Chen <peter.chen@nxp.com>
[ Upstream commit d745cc1ab65945b2d17ec9c5652f38299c054649 ]
Those strings are exposed to the user space as the
card name thus used in the GUIs. The common
standard is to avoid '_' here. The worst case
is 'sof-skl_hda_card' string.
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: Mark Brown <broonie@kernel.org>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20191028164624.14334-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Compute mclk to allow low power and performance settings
based on sample_rate:
[ 8000 - 11025] low power mode
[16000 - 64000] performance mode
Update Copyright year.
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
Add DSD_U8, DSD_U16_LE and DSD_U32_LE format constrains
in order to use SAI DSD settings for handling multi
channel capture streams. One channel per dataline.
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
Add dummy codec dai DSD_U8, DSD_U16_LE and DSD_U32_LE
format support to allow machine ASoC drivers that
support DSD formats to use snd-soc-dummy-dai
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
[ Upstream commit 7f5ad9c9003425175f46c94df380e8c9e558cfb5 ]
Gigabyte TRX40 Aorus Master is equipped with two USB-audio devices,
a Realtek ALC1220-VB codec (USB ID 0414:a001) and an ESS SABRE9218 DAC
(USB ID 0414:a000). The latter serves solely for the headphone output
on the front panel while the former serves for the rest I/Os (mostly
for the I/Os in the rear panel but also including the front mic).
Both chips do work more or less with the unmodified USB-audio driver,
but there are a few glitches. The ALC1220-VB returns an error for an
inquiry to some jacks, as already seen on other TRX40-based mobos.
However this machine has a slightly incompatible configuration, hence
the existing mapping cannot be used as is.
Meanwhile the ESS chip seems working without any quirk. But since
both audio devices don't provide any specific names, both cards appear
as "USB-Audio", and it's quite confusing for users.
This patch is an attempt to overcome those issues:
- The specific mapping table for ALC1220-VB is provided, reducing the
non-working nodes and renaming the badly chosen controls.
The connector map isn't needed here unlike other TRX40 quirks.
- For both USB IDs (0414:a000 and 0414:a001), provide specific card
name strings, so that user-space can identify more easily; and more
importantly, UCM profile can be applied to each.
Reported-by: Linus Torvalds <torvalds@linux-foundation.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526082810.29506-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 4020d1ccbe55bdf67b31d718d2400506eaf4b43f ]
The Asus USB DAC is a USB type-C audio dongle for connecting to
the headset and headphone. The volume minimum value -23040 which
is 0xa600 in hexadecimal with the resolution value 1 indicates
this should be endianness issue caused by the firmware bug. Add
a volume quirk to fix the volume control problem.
Also fixes this warning:
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[5] FU [Headset Capture Volume] ch = 1, val = -23040/0/1
Warning! Unlikely big volume range (=23040), cval->res is probably wrong.
[7] FU [Headset Playback Volume] ch = 1, val = -23040/0/1
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526062613.55401-1-chiu@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 399c01aa49e548c82d40f8161915a5941dd3c60e ]
We fixed the regression of the speaker volume for some Thinkpad models
(e.g. T570) by the commit 54947cd64c ("ALSA: hda/realtek - Fix
speaker output regression on Thinkpad T570"). Essentially it fixes
the DAC / pin pairing by a static table. It was confirmed and merged
to stable kernel later.
Now, interestingly, we got another regression report for the very same
model (T570) about the similar problem, and the commit above was the
culprit. That is, by some reason, there are devices that prefer the
DAC1, and another device DAC2!
Unfortunately those have the same ID and we have no idea what can
differentiate, in this patch, a new fixup model "tpt470-dock-fix" is
provided, so that users with such a machine can apply it manually.
When model=tpt470-dock-fix option is passed to snd-hda-intel module,
it avoids the fixed DAC pairing and the DAC1 is assigned to the
speaker like the earlier versions.
Fixes: 54947cd64c ("ALSA: hda/realtek - Fix speaker output regression on Thinkpad T570")
BugLink: https://apibugzilla.suse.com/show_bug.cgi?id=1172017
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200526062406.9799-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fb8cd6481ffd126f35e9e146a0dcf0c4e8899f2e ]
The "info.index" variable can be 31 in "1 << info.index".
This might trigger an undefined behavior since 1 is signed.
Fix this by casting 1 to 1u just to be sure "1u << 31" is defined.
Signed-off-by: Changming Liu <liu.changm@northeastern.edu>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/BL0PR06MB4548170B842CB055C9AF695DE5B00@BL0PR06MB4548.namprd06.prod.outlook.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit da7a8f1a8fc3e14c6dcc52b4098bddb8f20390be ]
This is another TRX40 based motherboard with ALC1220-VB USB-audio
that requires a static mapping table.
This motherboard also has a PCI device which advertises no codecs. The
PCI ID is 1022:1487 and PCI SSID is 1022:d102. As this is using the AMD
vendor ID, don't blacklist for now in case other boards have a working
audio device with the same ssid.
alsa-info.sh report for this board:
http://alsa-project.org/db/?f=0a742f89066527497b77ce16bca486daccf8a70c
Signed-off-by: Andrew Oakley <andrew@adoakley.name>
Link: https://lore.kernel.org/r/20200503141639.35519-1-andrew@adoakley.name
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 259eb82475316672a5d682a94dc8bdd53cf8d8c3 upstream.
A few known Clevo machines (PC50, PC70, X170) with ALC1220 codec need
the existing quirk for pins for PB51 and co.
Signed-off-by: PeiSen Hou <pshou@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200519065012.13119-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d9e8fe0cffbfdd18de96fa68ee2a8b667a0b046e upstream.
The Gigabyte X570 Aorus Xtreme motherboard with ALC1220 codec
requires a similar workaround for Clevo laptops to enforce the
DAC/mixer connection path. Set up a quirk entry for that.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=205275
Signed-off-by: Christian Lachner <gladiac@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200518053844.42743-2-gladiac@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e7513c5786f8b33f0c107b3759e433bc6cbb2efa upstream.
There is a corner case that ALSA keeps increasing the hw_ptr but DMA
already stop working/updating the position for a long time.
In following log we can see the position returned from DMA driver does
not move at all but the hw_ptr got increased at some point of time so
snd_pcm_avail() will return a large number which seems to be a buffer
underrun event from user space program point of view. The program
thinks there is space in the buffer and fill more data.
[ 418.510086] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 4096 avail 12368
[ 418.510149] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 6910 avail 9554
...
[ 418.681052] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 15102 avail 1362
[ 418.681130] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0
[ 418.726515] sound pcmC0D5p: pos 96 hw_ptr 16464 appl_ptr 16464 avail 16368
This is because the hw_base will be increased by runtime->buffer_size
frames unconditionally if the hw_ptr is not updated for over half of
buffer time. As the hw_base increases, so does the hw_ptr increased
by the same number.
The avail value returned from snd_pcm_avail() could exceed the limit
(buffer_size) easily becase the hw_ptr itself got increased by same
buffer_size samples when the corner case happens. In following log,
the buffer_size is 16368 samples but the avail is 21810 samples so
CRAS server complains about it.
[ 418.851755] sound pcmC0D5p: pos 96 hw_ptr 16464 appl_ptr 27390 avail 5442
[ 418.926491] sound pcmC0D5p: pos 96 hw_ptr 32832 appl_ptr 27390 avail 21810
cras_server[1907]: pcm_avail returned frames larger than buf_size:
sof-glkda7219max: :0,5: 21810 > 16368
By updating runtime->hw_ptr_jiffies each time the HWSYNC is called,
the hw_base will keep the same when buffer stall happens at long as
the interval between each HWSYNC call is shorter than half of buffer
time.
Following is a log captured by a patched kernel. The hw_base/hw_ptr
value is fixed in this corner case and user space program should be
aware of the buffer stall and handle it.
[ 293.525543] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 4096 avail 12368
[ 293.525606] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 6880 avail 9584
[ 293.525975] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 10976 avail 5488
[ 293.611178] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 15072 avail 1392
[ 293.696429] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0
...
[ 381.139517] sound pcmC0D5p: pos 96 hw_ptr 96 appl_ptr 16464 avail 0
Signed-off-by: Brent Lu <brent.lu@intel.com>
Reviewed-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/1589776238-23877-1-git-send-email-brent.lu@intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b0cb099062b0c18246c3a20caaab4c0afc303255 upstream.
The ST Audio ADCIII is an STDSP24 card plus extension box. With commit
e8a91ae18b ("ALSA: ice1712: Add support for STAudio ADCIII") we
enabled the ADCIII ports using the model=staudio option but forgot
this part to ensure the STDSP24 card is initialized properly.
Fixes: e8a91ae18b ("ALSA: ice1712: Add support for STAudio ADCIII")
Signed-off-by: Scott Bahling <sbahling@suse.com>
Cc: <stable@vger.kernel.org>
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1048934
Link: https://lore.kernel.org/r/20200518175728.28766-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 9e43342b464f1de570a3ad8256ac77645749ef45 ]
The ASUS laptop GL503VM with ALC295 can't detect the headset microphone.
The headset microphone does not work until pin 0x19 is enabled for it.
Signed-off-by: Chris Chiu <chiu@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Link: https://lore.kernel.org/r/20200512061525.133985-1-jian-hong@endlessm.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 14425f1f521fdfe274a7bb390637c786432e08b4 ]
Some models of the Samsung Notebook 9 have very quiet and distorted
headphone output. This quirk changes the VREF value of the ALC298
codec NID 0x1a from default HIZ to new 100.
[ adjusted to 5.7-base and rearranged in SSID order -- tiwai ]
Signed-off-by: Mike Pozulp <pozulp.kernel@gmail.com>
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207423
Link: https://lore.kernel.org/r/20200510032838.1989130-1-pozulp.kernel@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 24164f434dc9c23cd34fca1e36acea9d0581bdde ]
HP new platform has new mute led feature.
COEF index 0x34 bit 5 to control playback mute led.
COEF index 0x35 bit 2 and bit 3 to control Mic mute led.
[ corrected typos by tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 431e76c3edd76d84a0ed1eb81a286b2ddecc5ee4 ]
HP Note Book supported new mute Led.
Hardware PIN was not enough to meet old LED rule.
JD2 to control playback mute led.
GPO3 to control capture mute led.
(ALC285 didn't control GPO3 via verb command)
This two PIN just could control by COEF registers.
[ corrected typos by tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Link: https://lore.kernel.org/r/6741211598ba499687362ff2aa30626b@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 1a462be52f4505a2719631fb5aa7bfdbd37bfd8d ]
In the commit 8e85def5723e ("ALSA: hda: enable regmap internal
locking"), we re-enabled the regmap lock due to the reported
regression that showed the possible concurrent accesses. It was a
temporary workaround, and there are still a few opened races even
after the revert. In this patch, we cover those still opened windows
with a proper mutex lock and disable the regmap internal lock again.
First off, the patch introduces a new snd_hdac_device.regmap_lock
mutex that is applied for each snd_hdac_regmap_*() call, including
read, write and update helpers. The mutex is applied carefully so
that it won't block the self-power-up procedure in the helper
function. Also, this assures the protection for the accesses without
regmap, too.
The snd_hdac_regmap_update_raw() is refactored to use the standard
regmap_update_bits_check() function instead of the open-code. The
non-regmap case is still open-coded but it's an easy part. The all
read and write operations are in the single mutex protection, so it's
now race-free.
In addition, a couple of new helper functions are added:
snd_hdac_regmap_update_raw_once() and snd_hdac_regmap_sync(). Both
are called from HD-audio legacy driver. The former is to initialize
the given verb bits but only once when it's not initialized yet. Due
to this condition, the function invokes regcache_cache_only(), and
it's now performed inside the regmap_lock (formerly it was racy) too.
The latter function is for simply invoking regcache_sync() inside the
regmap_lock, which is called from the codec resume call path.
Along with that, the HD-audio codec driver code is slightly modified /
simplified to adapt those new functions.
And finally, snd_hdac_regmap_read_raw(), *_write_raw(), etc are
rewritten with the helper macro. It's just for simplification because
the code logic is identical among all those functions.
Tested-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200109090104.26073-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 8a71821f12a010d7100f9cc1f7b218aff0313c4a ]
GCC reports the following warning with W=1
sound/pci/hda/patch_realtek.c: In function ‘alc269_suspend’:
sound/pci/hda/patch_realtek.c:3616:29: warning: suggest braces around
empty body in an ‘if’ statement [-Wempty-body]
3616 | alc5505_dsp_suspend(codec);
| ^
sound/pci/hda/patch_realtek.c: In function ‘alc269_resume’:
sound/pci/hda/patch_realtek.c:3651:28: warning: suggest braces around empty body in an ‘if’ statement [-Wempty-body]
3651 | alc5505_dsp_resume(codec);
| ^
This is a classic macro problem and can indeed lead to bad program
flows.
Fix by using the usual "do { } while (0)" pattern
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Link: https://lore.kernel.org/r/20200111214736.3002-2-pierre-louis.bossart@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
The original patch updates REG_MICFIL_STAT in fsl_micfil_reset
method, but not in configure_hwvad_interrupts, fix this.
Fixes: e12b0060cd ("MLK-22598: ASoC: fsl_micfil: Reset channel output
data Flag")
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
During probe SOF driver loads the topology, which will send messages
to DSP to create the Audio components.
SOF DAI component at creation time calls probe() and touches DAI
registers thus we need to make sure DAI clocks are turned on during
probe.
Using pm_runtime_get_sync() we force resume to be called which will
enable all necessary clocks.
Later we call the pm_runtime_put_sync() and mark the device as
autosuspend. This will cause suspend to be called which will disable
all clocks to save power.
Later, when userspace uses the sound card (via aplay/arecord) or any
alsa-lib calls the normal runtime resume/suspend flow will happen.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
in_filled_len need to be reset on suspend, for suspend, every state
will go back to initilization state.
Fixes: 3e87108614d1 ("MLK-24044: ASoC: fsl_easrc_m2m: Fix output length not accurate")
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
initilize the m2m->complete in open() to avoid the NULL pointer in suspend
because the suspend can be called before initilizing m2m->complete in convert
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
commit f41224efcf8aafe80ea47ac870c5e32f3209ffc8 upstream.
This reverts commit 3b36b13d5e69d6f51ff1c55d1b404a74646c9757.
Enable power save node breaks some systems with ACL225. Revert the patch
and use a platform specific quirk for the original issue isntead.
Fixes: 3b36b13d5e69 ("ALSA: hda/realtek: Fix pop noise on ALC225")
BugLink: https://bugs.launchpad.net/bugs/1875916
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200503152449.22761-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 073919e09ca445d4486968e3f851372ff44cf2b5 upstream.
Kingston HyperX headset with 0951:16ad also needs the same quirk for
delaying the frequency controls.
Signed-off-by: Jesus Ramos <jesus-ramos@live.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/BY5PR19MB3634BA68C7CCA23D8DF428E796AF0@BY5PR19MB3634.namprd19.prod.outlook.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit c1f6e3c818dd734c30f6a7eeebf232ba2cf3181d upstream.
The rawmidi core allows user to resize the runtime buffer via ioctl,
and this may lead to UAF when performed during concurrent reads or
writes: the read/write functions unlock the runtime lock temporarily
during copying form/to user-space, and that's the race window.
This patch fixes the hole by introducing a reference counter for the
runtime buffer read/write access and returns -EBUSY error when the
resize is performed concurrently against read/write.
Note that the ref count field is a simple integer instead of
refcount_t here, since the all contexts accessing the buffer is
basically protected with a spinlock, hence we need no expensive atomic
ops. Also, note that this busy check is needed only against read /
write functions, and not in receive/transmit callbacks; the race can
happen only at the spinlock hole mentioned in the above, while the
whole function is protected for receive / transmit callbacks.
Reported-by: butt3rflyh4ck <butterflyhuangxx@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/CAFcO6XMWpUVK_yzzCpp8_XP7+=oUpQvuBeCbMffEDkpe8jWrfg@mail.gmail.com
Link: https://lore.kernel.org/r/s5heerw3r5z.wl-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 1b94e59d30afecf18254ad413e953e7587645a20 upstream.
ASUS ZenBook UX431DA requires an additional COEF setup when booted
from the recent Windows 10, otherwise it produces the noisy output.
The quirk turns on COEF 0x1b bit 10 that has been cleared supposedly
due to the pop noise reduction.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207553
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20200512073203.14091-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b590b38ca305d6d7902ec7c4f7e273e0069f3bcc upstream.
Lenovo Thinkpad T530 seems to have a sensitive internal mic capture
that needs to limit the mic boost like a few other Thinkpad models.
Although we may change the quirk for ALC269_FIXUP_LENOVO_DOCK, this
hits way too many other laptop models, so let's add a new fixup model
that limits the internal mic boost on top of the existing quirk and
apply to only T530.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1171293
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200514160533.10337-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 1034872123a06b759aba772b1c99612ccb8e632a ]
The snd-firewire-lib.ko has 'amdtp-packet' event of tracepoints. Current
printk format for the event includes 'sizeof(u8)' macro expected to be
extended in compilation time. However, this is not done. As a result,
perf tools cannot parse the event for printing:
$ mount -l -t debugfs
debugfs on /sys/kernel/debug type debugfs (rw,nosuid,nodev,noexec,relatime)
$ cat /sys/kernel/debug/tracing/events/snd_firewire_lib/amdtp_packet/format
...
print fmt: "%02u %04u %04x %04x %02d %03u %02u %03u %02u %01u %02u %s",
REC->second, REC->cycle, REC->src, REC->dest, REC->channel,
REC->payload_quadlets, REC->data_blocks, REC->data_block_counter,
REC->packet_index, REC->irq, REC->index,
__print_array(__get_dynamic_array(cip_header),
__get_dynamic_array_len(cip_header),
sizeof(u8))
$ sudo perf record -e snd_firewire_lib:amdtp_packet
[snd_firewire_lib:amdtp_packet] function sizeof not defined
Error: expected type 5 but read 0
This commit fixes it by obsoleting the macro with actual size.
Cc: <stable@vger.kernel.org>
Fixes: bde2bbdb30 ("ALSA: firewire-lib: use dynamic array for CIP header of tracing events")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Link: https://lore.kernel.org/r/20200503045718.86337-1-o-takashi@sakamocchi.jp
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 52e4e36807aeac1cdd07b14e509c8a64101e1a09 ]
Commit 317d931392 ("ALSA: hda/realtek - Set default power save node to
0") makes the ALC225 have pop noise on S3 resume and cold boot.
The previous fix enable power save node universally for ALC225, however
it makes some ALC225 systems unable to produce any sound.
So let's only enable power save node for the affected Dell Wyse
platform.
Fixes: 317d931392 ("ALSA: hda/realtek - Set default power save node to 0")
BugLink: https://bugs.launchpad.net/bugs/1866357
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20200503152449.22761-2-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit ca76282b6faffc83601c25bd2a95f635c03503ef ]
A race exists between build_pcms() and build_controls() phases of codec
setup. Build_pcms() sets up notifier for jack events. If a monitor event
is received before build_controls() is run, the initial jack state is
lost and never reported via mixer controls.
The problem can be hit at least with SOF as the controller driver. SOF
calls snd_hda_codec_build_controls() in its workqueue-based probe and
this can be delayed enough to hit the race condition.
Fix the issue by invalidating the per-pin ELD information when
build_controls() is called. The existing call to hdmi_present_sense()
will update the ELD contents. This ensures initial monitor state is
correctly reflected via mixer controls.
BugLink: https://github.com/thesofproject/linux/issues/1687
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20200428123836.24512-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
Add fsl,imx-audio-ak5552 to imx-ak5558 machine ASoC
driver, limit actual number of channels for ak5552
ADC support
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
We need to accumulate the input data length for calculating the
output length because there are data need to be prefilled in
the beginning. If the fraction part beyond the integrity of
sample is not considered in calculating, which may cause the
output task timeout.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Viorel Suman <viorel.suman@nxp.com>
At SPDIF TX interrupt the firmware running on M0+ core
overwrites TX DPATH BYPASS FEM bit which turns the
expected PHY DMAC TX PLL clock 2*bitclock instead of
expected 10*bitclock if SAI PLL is used as PHY DMAC TX
PLL.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
The existing "shutdown" callback does not consider
full duplex mode, fix this.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Update drivers constrains, limit sample rates to
[8Khz - 64Khz] and up to 8 channels.
Use SND_SOC_DAIFMT_PDM format type and remove
tdm slot settings to allow capture from multiple
SAI_RxD pins.
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Signed-off-by: Adrian Alonso <adrian.alonso@nxp.com>
The criteria used to drop eARC mode into ARC mode
makes eARC function to fail. Drop it for now.
Signed-off-by: Viorel Suman <viorel.suman@nxp.com>
Reviewed-by: Shengjiu Wang <shengjiu.wang@nxp.com>
With route "Playback" -> "Playback", dapm will enable
the "Playback" widget to be power on, there is this same
widget for codec component, then codec will enter
the SND_SOC_BIAS_ON level, then mclk is enabled
When the mclk is bound with a power domain, the power
domain will be enabled always.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
On imx8mp, there is no dedicated reset controller for DSP and
there is no dedicated power domain for DSP.
The power of DSP is bound with audiomix, so we need to check
the DSP status for judging the audiomix is reset or not.
We use the PID register for status check, after reset, it will
be set to zero, when DSP is used we set it to 1.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
Reviewed-by: Daniel Baluta <daniel.baluta@nxp.com>
Fix the error handler in probe. otherwise the resource
can't be release when error happen.
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Peng Zhang <peng.zhang_8@nxp.com>
commit 977dfef40c8996b69afe23a9094d184049efb7bb upstream.
The commit 3c6fd1f07ed0 ("ALSA: hda: Add driver blacklist") added a
new blacklist for the devices that are known to have empty codecs, and
one of the entries was ASUS ROG Zenith II (PCI SSID 1043:874f).
However, it turned out that the very same PCI SSID is used for the
previous model that does have the valid HD-audio codecs and the change
broke the sound on it.
Since the empty codec problem appear on the certain AMD platform (PCI
ID 1022:1487), this patch changes the blacklist matching to both PCI
ID and SSID using pci_match_id(). Also, the entry that was removed by
the previous fix for ASUS ROG Zenigh II is re-added.
Link: https://lore.kernel.org/r/20200424061222.19792-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit 54cb6221688660670a2e430892d7f4e6370263b8 ]
Fix the rsnd_ssi_stop function to skip disabling the individual SSIs of
a multi-SSI setup, as the actual stop is performed by rsnd_ssiu_stop_gen2
- the same logic as in rsnd_ssi_start. The attempt to disable these SSIs
was harmless, but caused a "status check failed" message to be printed
for every SSI in the multi-SSI setup.
The disabling of interrupts is still performed, as they are enabled for
all SSIs in rsnd_ssi_init, but care is taken to not accidentally set the
EN bit for an SSI where it was not set by rsnd_ssi_start.
Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200417153017.1744454-3-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 0c258657ddfe81b4fc0183378d800c97ba0b7cdd ]
The master SSI of a multi-SSI setup was attached both to the
RSND_MOD_SSI slot and the RSND_MOD_SSIP slot of the rsnd_dai_stream.
This is not correct wrt. the meaning of being "parent" in the rest of
the SSI code, where it seems to indicate an SSI that provides clock and
word sync but is not transmitting/receiving audio data.
Not treating the multi-SSI master as parent allows removal of various
special cases to the rsnd_ssi_is_parent conditions introduced in commit
a09fb3f28a60 ("ASoC: rsnd: Fix parent SSI start/stop in multi-SSI mode").
It also fixes the issue that operations performed via rsnd_dai_call()
were performed twice for the master SSI. This caused some "status check
failed" spam when stopping a multi-SSI stream as the driver attempted to
stop the master SSI twice.
Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200417153017.1744454-2-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 326b509238171d37402dbe308e154cc234ed1960 ]
If we don't find any pcm, pcm will point at address at an offset from
the the list head and not a meaningful structure. Fix this by returning
correct pcm if found and NULL if not. Found with coccinelle.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200415162849.308-1-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit b94e164759b82d0c1c80d4b1c8f12c9bee83f11d ]
The HDMI?_SEL register maps up to four stereo SSI data lanes onto the
sdata[0..3] inputs of the HDMI output block. The upper half of the
register contains four blocks of 4 bits, with the most significant
controlling the sdata3 line and the least significant the sdata0 line.
The shift calculation has an off-by-one error, causing the parent SSI to
be mapped to sdata3, the first multi-SSI child to sdata0 and so forth.
As the parent SSI transmits the stereo L/R channels, and the HDMI core
expects it on the sdata0 line, this causes no audio to be output when
playing stereo audio on a multichannel capable HDMI out, and
multichannel audio has permutated channels.
Fix the shift calculation to map the parent SSI to sdata0, the first
child to sdata1 etc.
Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200415141017.384017-3-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit a09fb3f28a60ba3e928a1fa94b0456780800299d ]
The parent SSI of a multi-SSI setup must be fully setup, started and
stopped since it is also part of the playback/capture setup. So only
skip the SSI (as per commit 203cdf51f2 ("ASoC: rsnd: SSI parent cares
SWSP bit") and commit 597b046f0d ("ASoC: rsnd: control SSICR::EN
correctly")) if the SSI is parent outside of a multi-SSI setup.
Signed-off-by: Matthias Blankertz <matthias.blankertz@cetitec.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Link: https://lore.kernel.org/r/20200415141017.384017-2-matthias.blankertz@cetitec.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 26d87881590fd55ccdd8f829498d7b3033f81990 ]
As done in already existing cases, we should use le32_to_cpu macro while
accessing hdr->magic. Found with sparse.
Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski@linux.intel.com>
Link: https://lore.kernel.org/r/20200415162435.31859-2-amadeuszx.slawinski@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
