commit 23ba28616d3063bd4c4953598ed5e439ca891101 upstream.
Currently each channel is added as list to dai channel list, however
there is danger of adding same channel to multiple dai channel list
which endups corrupting the other list where its already added.
This patch ensures that the channel is actually free before adding to
the dai channel list and also ensures that the channel is on the list
before deleting it.
This check was missing previously, and we did not hit this issue as
we were testing very simple usecases with sequence of amixer commands.
Fixes: a70d924575 ("ASoC: wcd934x: add capture dapm widgets")
Fixes: dd9eb19b56 ("ASoC: wcd934x: add playback dapm widgets")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20211130160507.22180-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4739d88ad8e1900f809f8a5c98f3c1b65bf76220 upstream.
msm_routing_put_audio_mixer() can return incorrect value in various scenarios.
scenario 1:
amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1
amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 0
return value is 0 instead of 1 eventhough value was changed
scenario 2:
amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1
amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1
return value is 1 instead of 0 eventhough the value was not changed
scenario 3:
amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 0
return value is 1 instead of 0 eventhough the value was not changed
Fix this by adding checks, so that change notifications are sent correctly.
Fixes: e3a33673e8 ("ASoC: qdsp6: q6routing: Add q6routing driver")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20211130163110.5628-1-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4999d703c0e66f9f196b6edc0b8fdeca8846b8b6 upstream.
Move the declaration of temporary arrays to somewhere that won't go out
of scope before the devm_clk_hw_register() call, lest we be at the whim
of the compiler for whether those stack variables get overwritten.
Fixes a crash seen with gcc version 11.2.1 20210728 (Red Hat 11.2.1-1)
Fixes: edbd24ea1e ("ASoC: rt5682: Drop usage of __clk_get_name()")
Signed-off-by: Rob Clark <robdclark@chromium.org>
Reviewed-by: Stephen Boyd <swboyd@chromium.org>
Link: https://lore.kernel.org/r/20211118010453.843286-1-robdclark@gmail.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit fb1af5bea4670c835e42fc0c14c49d3499468774 upstream.
Olivia Mackintosh has posted to alsa-devel reporting that
there's a potential bug that could break mixer quirks for Pioneer
devices introduced by 6d27788160362a7ee6c0d317636fe4b1ddbe59a7
"ALSA: usb-audio: Add support for the Pioneer DJM 750MK2
Mixer/Soundcard".
This happened because the DJM 750 MK2 was added last to the Pioneer DJM
device table index and defined as 0x4 but was added to snd_djm_devices[]
just after the DJM 750 (MK1) entry instead of last, after the DJM 900
NXS2. This escaped review.
To prevent that from ever happening again, Takashi Iwai suggested to use
C99 array designators in snd_djm_devices[] instead of simply reordering
the entries.
Fixes: 6d2778816036 ("ALSA: usb-audio: Add support for the Pioneer DJM 750MK2")
Reported-by: Olivia Mackintosh <livvy@base.nu>
Suggested-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Geraldo Nascimento <geraldogabriel@gmail.com>
Link: https://lore.kernel.org/r/Yau46FDzoql0SNnW@geday
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 619764cc2ec9ce1283a8bbcd89a1376a7c68293b upstream.
This fixes the SND_PCI_QUIRK(...) of the TongFang PHxTxX1 barebone. This
fixes the issue of sound not working after s3 suspend.
When waking up from s3 suspend the Coef 0x10 is set to 0x0220 instead of
0x0020. Setting the value manually makes the sound work again. This patch
does this automatically.
While being on it, I also fixed the comment formatting of the quirk and
shortened variable and function names.
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Fixes: dd6dd6e3c7 ("ALSA: hda/realtek: Add quirk for TongFang PHxTxX1")
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211202165010.876431-1-wse@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit b6409dd6bdc03aa178bbff0d80db2a30d29b63ac upstream.
When control_compat.c:copy_ctl_value_to_user() is used, by
ctl_elem_read_user() & ctl_elem_write_user(), it must also copy back the
snd_ctl_elem_id value that may have been updated (filled in) by the call
to snd_ctl_elem_read/snd_ctl_elem_write().
This matches the functionality provided by snd_ctl_elem_read_user() and
snd_ctl_elem_write_user(), via snd_ctl_build_ioff().
Without this, and without making additional calls to snd_ctl_info()
which are unnecessary when using the non-compat calls, a userspace
application will not know the numid value for the element and
consequently will not be able to use the poll/read interface on the
control file to determine which elements have updates.
Signed-off-by: Alan Young <consult.awy@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211202150607.543389-1-consult.awy@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 65cc4ad62a9ed47c0b4fcd7af667d97d7c29f19d upstream.
For cs8409, it is required to run Jack Detect on resume.
Jack Detect on cs8409+cs42l42 requires an interrupt from
cs42l42 to be sent to cs8409 which is propogated to the driver
via an unsolicited event.
However, the hda_codec drops unsolicited events if the power_state
is not set to PMSG_ON. Which is set at the end of the resume call.
This means there is a race condition between setting power_state
to PMSG_ON and receiving the interrupt.
To solve this, we can add an API to set the power_state earlier
and call that before we start Jack Detect.
This does not cause issues, since we know inside our driver that
we are already initialized, and ready to handle the unsolicited
events.
Signed-off-by: Stefan Binding <sbinding@opensource.cirrus.com>
Signed-off-by: Vitaly Rodionov <vitalyr@opensource.cirrus.com>
Cc: <stable@vger.kernel.org> # v5.15+
Link: https://lore.kernel.org/r/20211128115558.71683-1-vitalyr@opensource.cirrus.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 83de8f83816e8e15227dac985163e3d433a2bf9d upstream.
The recent change made mistakenly the stream for capture started at
prepare stage. Add the stream direction check to avoid it.
Fixes: 9c9a3b9da891 ("ALSA: usb-audio: Rename early_playback_start flag with lowlatency_playback")
Link: https://lore.kernel.org/r/20211119102629.7476-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit eee5d6f1356a016105a974fb176b491288439efa upstream.
The recent regression report revealed that the judgment of the
low-latency playback mode based on the runtime->stop_threshold cannot
work reliably at the prepare stage, as sw_params call may happen at
any time, and PCM dmix actually sets it up after the prepare call.
This ended up with the stall of the stream as PCM ack won't be issued
at all.
For addressing this, check the free-wheeling mode again at the PCM
trigger right before starting the stream again, and allow switching to
the non-LL mode at a late stage.
Fixes: d5f871f89e21 ("ALSA: usb-audio: Improved lowlatency playback support")
Reported-and-tested-by: Kirill A. Shutemov <kirill.shutemov@linux.intel.com>
Link: https://lore.kernel.org/r/20211117161855.m45mxcqszkfcetai@box.shutemov.name
Link: https://lore.kernel.org/r/20211119102459.7055-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 53451b6da8271905941eb1eb369db152c4bd92f2 upstream.
The recent support for the improved low-latency playback mode applied
the SNDRV_PCM_INFO_EXPLICIT_SYNC flag for the target streams, but this
was a slight overkill. The use of the flag above disables effectively
both PCM status and control mmaps, while basically what we want to
track is only about the appl_ptr update.
For less restriction, use a more proper flag,
SNDRV_PCM_INFO_SYNC_APPLPTR instead, which disables only the control
mmap.
Fixes: d5f871f89e21 ("ALSA: usb-audio: Improved lowlatency playback support")
Link: https://lore.kernel.org/r/20211011103650.10182-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 813a17cab9b708bbb1e0db8902e19857b57196ec upstream.
While draining a stream, ALSA PCM core stops the stream by issuing
snd_pcm_stop() after all data has been sent out. And, at PCM trigger
stop, currently USB-audio driver kills the in-flight URBs explicitly,
then at sync-stop ops, sync with the finish of all remaining URBs.
This might result in a drop of the drained samples as most of
USB-audio devices / hosts allow relatively long in-flight samples (as
a sort of FIFO).
For avoiding the trimming, this patch changes the stream-stop behavior
during PCM draining state. Under that condition, the pending URBs
won't be killed. The leftover in-flight URBs are caught by the
sync-stop operation that shall be performed after the trigger-stop
operation.
Link: https://lore.kernel.org/r/20210929080844.11583-10-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d5f871f89e21bb71827ea57bd484eedea85839a0 upstream.
This is another attempt to improve further the handling of playback
stream in the low latency mode. The latest workaround in commit
4267c5a8f3 ("ALSA: usb-audio: Work around for XRUN with low latency
playback") revealed that submitting URBs forcibly in advance may
trigger XRUN easily. In the classical mode, this problem was avoided
by practically delaying the submission of the actual data with the
pre-submissions of silent data before triggering the stream start.
But that is exactly what we want to avoid.
Now, in this patch, instead of the previous workaround, we take a
similar approach as used in the implicit feedback mode. The URBs are
queued at the PCM trigger start like before, but we check whether the
buffer has been already filled enough before each submission, and
stop queuing if the data overcomes the threshold. The remaining URBs
are kept in the ready list, and they will be retrieved in the URB
complete callback of other (already queued) URBs. In the complete
callback, we try to fill the data and submit as much as possible
again. When there is no more available in-flight URBs that may handle
the pending data, we'll check in PCM ack callback and submit and
process URBs there in addition. In this way, the amount of in-flight
URBs may vary dynamically and flexibly depending on the available data
without hitting XRUN.
The following things are changed to achieve the behavior above:
* The endpoint prepare callback is changed to return an error code;
when there is no enough data available, it may return -EAGAIN.
Currently only prepare_playback_urb() returns the error.
The evaluation of the available data is a bit messy here; we can't
check with snd_pcm_avail() at the point of prepare callback (as
runtime->status->hwptr hasn't been updated yet), hence we manually
estimate the appl_ptr and compare with the internal hwptr_done to
calculate the available frames.
* snd_usb_endpoint_start() doesn't submit full URBs if the prepare
callback returns -EAGAIN, and puts the remaining URBs to the ready
list for the later submission.
* snd_complete_urb() treats the URBs in the low-latency mode similarly
like the implicit feedback mode, and submissions are done in
(now exported) snd_usb_queue_pending_output_urbs().
* snd_usb_queue_pending_output_urbs() again checks the error value
from the prepare callback. If it's -EAGAIN for the normal stream
(i.e. not implicit feedback mode), we push it back to the ready list
again.
* PCM ack callback is introduced for the playback stream, and it calls
snd_usb_queue_pending_output_urbs() if there is no in-flight URB
while the stream is running. This corresponds to the case where the
system needs the appl_ptr update for re-submitting a new URB.
* snd_usb_queue_pending_output_urbs() and the prepare EP callback
receive in_stream_lock argument, which is a bool flag indicating the
call path from PCM ack. It's needed for avoiding the deadlock of
snd_pcm_period_elapsed() calls.
* Set the new SNDRV_PCM_INFO_EXPLICIT_SYNC flag when the new
low-latency mode is deployed. This assures catching each applptr
update even in the mmap mode.
Fixes: 4267c5a8f3 ("ALSA: usb-audio: Work around for XRUN with low latency playback")
Link: https://lore.kernel.org/r/20210929080844.11583-9-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit d215f63d49da9a8803af3e81acd6cad743686573 upstream.
This is yet more preparation for the upcoming changes.
Extend snd_usb_endpoint_next_packet_size() to check the available
frames and return -EAGAIN if the next packet size is equal or exceeds
the given size. This will be needed for avoiding XRUN during the low
latency operation.
As of this patch, avail=0 is passed, i.e. the check is skipped and no
behavior change.
Link: https://lore.kernel.org/r/20210929080844.11583-7-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit bceee75387554f682638e719d1ea60125ea78cea upstream.
When a playback stream runs in the implicit feedback mode, its
operation is passive and won't start unless the capture packet is
received. This behavior contradicts with the low-latency playback
mode, and we should turn off lowlatency_playback flag accordingly.
In theory, we may take the low-latency mode when the playback-first
quirk is set, but it still conflicts with the later operation with the
fixed packet numbers, so it's disabled all together for now.
Link: https://lore.kernel.org/r/20210929080844.11583-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit e581f1cec4f899f788f6c9477f805b1d5fef25e2 upstream.
The free-wheel stream operation like dmix may not update the appl_ptr
appropriately, and it doesn't fit with the low-latency playback mode.
Disable the low-latency playback operation when the stream is set up
in such a mode.
Link: https://lore.kernel.org/r/20210929080844.11583-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 9c9a3b9da891cc70405a544da6855700eddcbb71 upstream.
This is a preparation patch for the upcoming low-latency improvement
changes.
Rename early_playback_start flag with lowlatency_playback as it's more
intuitive. The new flag is basically a reverse meaning.
Along with the rename, factor out the code to set the flag to a
function. This makes the complex condition checks simpler.
Also, the same flag is introduced to snd_usb_endpoint, too, that is
carried from the snd_usb_substream flag. Currently the endpoint flag
isn't still referred, but will be used in later patches.
Link: https://lore.kernel.org/r/20210929080844.11583-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
commit 4e7cf1fbb34ecb472c073980458cbe413afd4d64 upstream.
When a single clock source is shared among several endpoints, we have
to keep the same rate on all active endpoints as long as the clock is
being used. For dealing with such a case, this patch adds one more
check in the hw params constraint for the rate to take the shared
clocks into account. The current rate is evaluated from the endpoint
list that applies the same clock source.
BugLink: https://bugzilla.suse.com/show_bug.cgi?id=1190418
Link: https://lore.kernel.org/r/20210929080844.11583-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
[ Upstream commit ea157c2ba821dab789a544cd9fbe44dc07036ff8 ]
Interrupt Clear registers WCD938X_INTR_CLEAR_0 - WCD938X_INTR_CLEAR_2
are not marked as volatile. This has resulted in a missing interrupt bug
while performing runtime pm. regcache_sync() during runtime pm resume path
will write to Interrupt clear registers with previous values which basically
clears the pending interrupt and actual interrupt handler never sees this
interrupt.
This issue is more visible with headset plug-in plug-out case compared to
headset button.
Fix this by adding the Interrupt clear registers to volatile range
Fixes: 8d78602aa8 ("ASoC: codecs: wcd938x: add basic driver")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20211116114623.11891-2-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 7e567b5ae06315ef2d70666b149962e2bb4b97af ]
snd_ctl_remove() has to be called with card->controls_rwsem held (when
called after the card instantiation). This patch add the missing
rwsem calls around it.
Fixes: 8a9782346d ("ASoC: topology: Add topology core")
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Link: https://lore.kernel.org/r/20211116071812.18109-1-tiwai@suse.de
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit 861afeac7990587588d057b2c0b3222331c3da29 ]
Stream IDs are reused across multiple BackEnd mixers, do not reset the
stream mixers if they are not already set for that particular FrontEnd.
Ex:
amixer cset iface=MIXER,name='SLIMBUS_0_RX Audio Mixer MultiMedia1' 1
would set the MultiMedia1 steam for SLIMBUS_0_RX, however doing below
command will reset previously setup MultiMedia1 stream, because both of them
are using MultiMedia1 PCM stream.
amixer cset iface=MIXER,name='SLIMBUS_2_RX Audio Mixer MultiMedia1' 0
reset the FrontEnd Mixers conditionally to fix this issue.
This is more noticeable in desktop setup, where in alsactl tries to restore
the alsa state and overwriting the previous mixer settings.
Fixes: e3a33673e8 ("ASoC: qdsp6: q6routing: Add q6routing driver")
Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Link: https://lore.kernel.org/r/20211116114721.12517-3-srinivas.kandagatla@linaro.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
[ Upstream commit fd572393baf0350835e8d822db588f679dc7bcb8 ]
If codec is in runtime suspend, but controller is not, hotplug events
are missed as the codec has no way to alert the controller. Problem does
not occur if both controller and codec are active, or when both are
suspended.
An easy way to reproduce is to play an audio stream on one codec (e.g.
to HDMI/DP display codec), wait for other HDA codec to go to runtime
suspend, and then plug in a headset to the suspended codec. The jack
event is not reported correctly in this case. Another way to reproduce
is to force controller to stay active with
"snd_sof_pci.sof_pci_debug=0x1"
Fix the issue by reconfiguring the WAKEEN register when powering up/down
individual links, and handling control events in the interrupt handler.
Fixes: 87fc20e4a0 ("ASoC: SOF: Intel: hda: use hdac_ext fine-grained link management")
Reported-by: Hui Wang <hui.wang@canonical.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>
Reviewed-by: Péter Ujfalusi <peter.ujfalusi@linux.intel.com>
Link: https://lore.kernel.org/r/20211105111655.668777-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
commit 174a7fb3859ae75b0f0e35ef852459d8882b55b5 upstream.
This applies a SND_PCI_QUIRK(...) to the ASRock NUC Box 1100 series. This
fixes the issue of the headphone jack not being detected unless warm
rebooted from a certain other OS.
When booting a certain other OS some coeff settings are changed that enable
the audio jack. These settings are preserved on a warm reboot and can be
easily dumped.
The relevant indexes and values where gathered by naively diff-ing and
reading a working and a non-working coeff dump.
Signed-off-by: Werner Sembach <wse@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20211112110704.1022501-1-wse@tuxedocomputers.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
