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>
Currently, when the HD-audio controller driver doesn't detect any
codecs, it tries to abort the probe. But this abort happens at the
delayed probe, i.e. the primary probe call already returned success,
hence the driver is never unbound until user does so explicitly.
As a result, it may leave the HD-audio device in the running state
without the runtime PM. More badly, if the device is a HD-audio bus
that is tied with a GPU, GPU cannot reach to the full power down and
consumes unnecessarily much power.
This patch changes the logic after no-codec situation; it continues
probing without the further codec initialization but keep the
controller driver running normally.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Tested-by: Roy Spliet <nouveau@spliet.org>
Link: https://lore.kernel.org/r/20200413082034.25166-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
snd-hda-intel driver handles the most of its probe task in the delayed
work (either via workqueue or via firmware loader). When an error
happens in the later delayed probe, we can't deregister the device
itself because the probe callback already returned success and the
device was bound. So, for now, we set hda->init_failed flag and make
the rest untouched until the device gets really unbound.
However, this leaves the device up running, keeping the resources
without any use that prevents other operations.
In this patch, we release the resources at first when a probe error
happens in the delayed probe stage, but keeps the top-level object, so
that the PM and other ops can still refer to the object itself.
Also for simplicity, snd_hda_intel object is allocated via devm, so
that we can get rid of the explicit kfree calls.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
At the error path of the firmware loading error, the driver tries to
release the card object and set NULL to drvdata. This may be referred
badly at the possible PM action, as the driver itself is still bound
and the PM callbacks read the card object.
Instead, we continue the probing as if it were no option set. This is
often a better choice than the forced abort, too.
Fixes: 5cb543dba9 ("ALSA: hda - Deferred probing with request_firmware_nowait()")
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=207043
Link: https://lore.kernel.org/r/20200413082034.25166-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent AMD platform exposes an HD-audio bus but without any actual
codecs, which is internally tied with a USB-audio device, supposedly.
It results in "no codecs" error of HD-audio bus driver, and it's
nothing but a waste of resources.
This patch introduces a static blacklist table for skipping such a
known bogus PCI SSID entry. As of writing this patch, the known SSIDs
are:
* 1043:874f - ASUS ROG Zenith II / Strix
* 1462:cb59 - MSI TRX40 Creator
* 1462:cb60 - MSI TRX40
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some recent boards (supposedly with a new AMD platform) contain the
USB audio class 2 device that is often tied with HD-audio. The device
exposes an Input Gain Pad control (id=19, control=12) but this node
doesn't behave correctly, returning an error for each inquiry of
GET_MIN and GET_MAX that should have been mandatory.
As a workaround, simply ignore this node by adding a usbmix_name_map
table entry. The currently known devices are:
* 0414:a002 - Gigabyte TRX40 Aorus Pro WiFi
* 0b05:1916 - ASUS ROG Zenith II
* 0b05:1917 - ASUS ROG Strix
* 0db0:0d64 - MSI TRX40 Creator
* 0db0:543d - MSI TRX40
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=206543
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200408140449.22319-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The commit 0d6defc7e0 ("ASoC: stm32: sai: manage rebind issue")
converts some function calls to their non-devm equivalents. The
appropriate cleanup code was added to the remove function, but not
to the probe function. Add a call to snd_dmaengine_pcm_unregister
to compensate for the call to snd_dmaengine_pcm_register in case
of subsequent failure.
Fixes: commit 0d6defc7e0 ("ASoC: stm32: sai: manage rebind issue")
Signed-off-by: Julia Lawall <Julia.Lawall@inria.fr>
Acked-by: Olivier Moysan <olivier.moysan@st.com>
Link: https://lore.kernel.org/r/1586099028-5104-1-git-send-email-Julia.Lawall@inria.fr
Signed-off-by: Mark Brown <broonie@kernel.org>
Using a Canon Lake machine with the SOF driver causes dmesg to fill
up with a ton of these messages:
[ 275.902194] sof-audio-pci 0000:00:1f.3: firmware boot complete
[ 351.529358] sof-audio-pci 0000:00:1f.3: firmware boot complete
[ 560.049047] sof-audio-pci 0000:00:1f.3: firmware boot complete
etc.
Since the DSP is powered down when not in used this happens everytime
e.g. a notification plays, polluting dmesg.
Turn this messages into a debug message, matching what the code already
does for the ""booting DSP firmware" message.
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/20200402184948.3014-2-hdegoede@redhat.com
Signed-off-by: Mark Brown <broonie@kernel.org>
An empty merge for the original fix for PCM OSS regression where the
same fix is already applied in a different form.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
[ This is essentially the same fix as commit ae769d3556, but it's
adapted to the latest code for 5.7; hence it contains no Fixes or
other tags for avoid backport confusion -- tiwai ]
The recent fix for the OOB access in PCM OSS plugins (commit
f2ecf903ef: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a
regression on OSS applications. The patch introduced the size check
in client and slave size calculations to limit to each plugin's buffer
size, but I overlooked that some code paths call those without
allocating the buffer but just for estimation.
This patch fixes the bug by skipping the size check for those code
paths while keeping checking in the actual transfer calls.
Link: https://lore.kernel.org/r/20200403073818.27943-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The recent fix for the OOB access in PCM OSS plugins (commit
f2ecf903ef: "ALSA: pcm: oss: Avoid plugin buffer overflow") caused a
regression on OSS applications. The patch introduced the size check
in client and slave size calculations to limit to each plugin's buffer
size, but I overlooked that some code paths call those without
allocating the buffer but just for estimation.
This patch fixes the bug by skipping the size check for those code
paths while keeping checking in the actual transfer calls.
Fixes: f2ecf903ef ("ALSA: pcm: oss: Avoid plugin buffer overflow")
Tested-and-reported-by: Jari Ruusu <jari.ruusu@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20200403072515.25539-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If I2C is n but SoundWire is m, building fails:
sound/soc/codecs/rt5682.c:3716:1: warning: data definition has no type or storage class
module_i2c_driver(rt5682_i2c_driver);
^~~~~~~~~~~~~~~~~
sound/soc/codecs/rt5682.c:3716:1: error: type defaults to 'int' in declaration of 'module_i2c_driver' [-Werror=implicit-int]
sound/soc/codecs/rt5682.c:3716:1: warning: parameter names (without types) in function declaration
Guard this use #ifdef CONFIG_I2C.
Fixes: 5549ea6479 ("ASoC: rt5682: fix unmet dependencies")
Signed-off-by: YueHaibing <yuehaibing@huawei.com>
Link: https://lore.kernel.org/r/20200401091055.34112-1-yuehaibing@huawei.com
Signed-off-by: Mark Brown <broonie@kernel.org>
soc_compr_trigger_fe() allows start or stop after pause_push.
In dpcm_be_dai_trigger(), however, only pause_release is allowed
command after pause_push.
So, start or stop after pause in compress offload is always
returned as error if the compress offload is used with dpcm.
To fix the problem, SND_SOC_DPCM_STATE_PAUSED should be allowed
for start or stop command.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Reviewed-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/004d01d607c1$7a3d5250$6eb7f6f0$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Since a virtual mixer has no backing registers
to decide which path to connect,
it will try to match with initial state.
This is to ensure that the default mixer choice will be
correctly powered up during initialization.
Invert flag is used to select initial state of the virtual switch.
Since actual hardware can't be disconnected by virtual switch,
connected is better choice as initial state in many cases.
Signed-off-by: Gyeongtaek Lee <gt82.lee@samsung.com>
Link: https://lore.kernel.org/r/01a301d60731$b724ea10$256ebe30$@samsung.com
Signed-off-by: Mark Brown <broonie@kernel.org>
At the moment, playing audio with PulseAudio with the qdsp6 driver
results in distorted sound. It seems like its timer-based scheduling
does not work properly with qdsp6 since setting tsched=0 in
the PulseAudio configuration avoids the issue.
Apparently this happens when the pointer() callback is not accurate
enough. There is a SNDRV_PCM_INFO_BATCH flag that can be used to stop
PulseAudio from using timer-based scheduling by default.
According to https://www.alsa-project.org/pipermail/alsa-devel/2014-March/073816.html:
The flag is being used in the sense explained in the previous audio
meeting -- the data transfer granularity isn't fine enough but aligned
to the period size (or less).
q6asm-dai reports the position as multiple of
prtd->pcm_count = snd_pcm_lib_period_bytes(substream)
so it indeed just a multiple of the period size.
Therefore adding the flag here seems appropriate and makes audio
work out of the box.
Fixes: 2a9e92d371 ("ASoC: qdsp6: q6asm: Add q6asm dai driver")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Reviewed-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20200330175210.47518-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
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>
patch_realtek.c has historically failed to properly configure the PC
Beep Hidden Register for the ALC256 codec (among others). Depending on
your kernel version, symptoms of this misconfiguration can range from
chassis noise, picked up by a poorly-shielded PCBEEP trace, getting
amplified and played on your internal speaker and/or headphones to loud
feedback, which responds to the "Headphone Mic Boost" ALSA control,
getting played through your headphones. For details of the problem, see
the patch in this series titled "ALSA: hda/realtek - Set principled PC
Beep configuration for ALC256", which fixes the configuration.
These symptoms have been most noticed on the Dell XPS 13 9350 and 9360,
popular laptops that use the ALC256. As a result, several model-specific
fixups have been introduced to try and fix the problem, the most
egregious of which locks the "Headphone Mic Boost" control as a hack to
minimize noise from a feedback loop that shouldn't have been there in
the first place.
Now that the underlying issue has been fixed, remove all these fixups.
Remaining fixups needed by the XPS 13 are all picked up by existing pin
quirks.
This change should, for the XPS 13 9350/9360
- Significantly increase volume and audio quality on headphones
- Eliminate headphone popping on suspend/resume
- Allow "Headphone Mic Boost" to be set again, making the headphone
jack fully usable as a microphone jack too.
Fixes: 8c69729b44 ("ALSA: hda - Fix headphone noise after Dell XPS 13 resume back from S3")
Fixes: 423cd78561 ("ALSA: hda - Fix headphone noise on Dell XPS 13 9360")
Fixes: e4c9fd10eb ("ALSA: hda - Apply headphone noise quirk for another Dell XPS 13 variant")
Fixes: 1099f48457 ("ALSA: hda/realtek: Reduce the Headphone static noise on XPS 9350/9360")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/b649a00edfde150cf6eebbb4390e15e0c2deb39a.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The Realtek PC Beep Hidden Register[1] is currently set by
patch_realtek.c in two different places:
In alc_fill_eapd_coef(), it's set to the value 0x5757, corresponding to
non-beep input on 1Ah and no 1Ah loopback to either headphones or
speakers. (Although, curiously, the loopback amp is still enabled.) This
write was added fairly recently by commit e3743f431143 ("ALSA:
hda/realtek - Dell headphone has noise on unmute for ALC236") and is a
safe default. However, it happens in the wrong place:
alc_fill_eapd_coef() runs on module load and cold boot but not on S3
resume, meaning the register loses its value after suspend.
Conversely, in alc256_init(), the register is updated to unset bit 13
(disable speaker loopback) and set bit 5 (set non-beep input on 1Ah).
Although this write does run on S3 resume, it's not quite enough to fix
up the register's default value of 0x3717. What's missing is a set of
bit 14 to disable headphone loopback. Without that, we end up with a
feedback loop where the headphone jack is being driven by amplified
samples of itself[2].
This change eliminates the update in alc256_init() and replaces it with
the 0x5757 write from alc_fill_eapd_coef(). Kailang says that 0x5757 is
supposed to be the codec's default value, so using it will make
debugging easier for Realtek.
Affects the ALC255, ALC256, ALC257, ALC235, and ALC236 codecs.
[1] Newly documented in Documentation/sound/hd-audio/realtek-pc-beep.rst
[2] Setting the "Headphone Mic Boost" control from userspace changes
this feedback loop and has been a widely-shared workaround for headphone
noise on laptops like the Dell XPS 13 9350. This commit eliminates the
feedback loop and makes the workaround unnecessary.
Fixes: e1e8c1fdce ("ALSA: hda/realtek - Dell headphone has noise on unmute for ALC236")
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/bf22b417d1f2474b12011c2a39ed6cf8b06d3bf5.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This codec (among others) has a hidden set of audio routes, apparently
designed to allow PC Beep output without a mixer widget on the output
path, which are controlled by an undocumented Realtek vendor register.
The default configuration of these routes means that certain inputs
aren't accessible, necessitating driver control of the register.
However, Realtek has provided no documentation of the register, instead
opting to fix issues by providing magic numbers, most of which have been
at least somewhat erroneous. These magic numbers then get copied by
others into model-specific fixups, leading to a fragmented and buggy set
of configurations.
To get out of this situation, I've reverse engineered the register by
flipping bits and observing how the codec's behavior changes. This
commit documents my findings. It does not change any code.
Cc: stable@vger.kernel.org
Signed-off-by: Thomas Hebb <tommyhebb@gmail.com>
Link: https://lore.kernel.org/r/bd69dfdeaf40ff31c4b7b797c829bb320031739c.1585584498.git.tommyhebb@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
As of commit:
ASoC: soc-core: care .ignore_suspend for Component suspend
function soc-core::snd_soc_suspend no longer ignores 'ignore_suspend'
flag for dai links. While BE dai link for System Pin is
supposed to follow standard suspend-resume flow, appended
'ignore_suspend' flag disturbs that flow and causes audio to break
right after resume. Remove the flag to address this.
Link to first message in conversation:
https://lkml.org/lkml/2020/3/18/54
Cezary Rojewski (4):
ASoC: Intel: broadwell: Remove ignore_suspend flag from SSP0 dai link
ASoC: Intel: haswell: Remove ignore_suspend flag from SSP0 dai link
ASoC: Intel: bdw-rt5677: Remove ignore_suspend flag from SSP0 dai link
ASoC: Intel: bdw-rt5650: Remove ignore_suspend flag from SSP0 dai link
sound/soc/intel/boards/bdw-rt5650.c | 1 -
sound/soc/intel/boards/bdw-rt5677.c | 1 -
sound/soc/intel/boards/broadwell.c | 1 -
sound/soc/intel/boards/haswell.c | 1 -
4 files changed, 4 deletions(-)
--
2.17.1
