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Since the recent refactoring, it's been reported that some USB-audio
devices (typically webcams) are no longer detected properly by
PulseAudio. The debug session revealed that it's failing at probing
by PA to try the sample rate 44.1kHz while the device has discrete
sample rates other than 44.1kHz. But the puzzle was that arecord
works as is, and some other devices with the discrete rates work,
either.
After all, this turned out to be the lack of the dependencies in a few
hw constraint rules: snd_pcm_hw_rule_add() has the (variable)
arguments specifying the dependent parameters, and some functions
didn't set the target parameter itself as the dependencies. This
resulted in an invalid parameter that could be generated only in a
certain call pattern. This bug itself has been present in the code,
but it didn't trigger errors just because the rules were casually
avoiding such a corner case. After the recent refactoring and
cleanup, however, the hw constraints work "as expected", and the
problem surfaced now.
For fixing the problem above, this patch adds the missing dependent
parameters to each snd_pcm_hw_rule() call.
Fixes: bc4e94aa8e72 ("ALSA: usb-audio: Handle discrete rates properly in hw constraints")
BugLink: http://bugzilla.opensuse.org/show_bug.cgi?id=1181014
Link: https://lore.kernel.org/r/20210120204554.30177-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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After hibernation, HDA controller can't be runtime-suspended after
commit 215a22ed31a1 ("ALSA: hda: Refactor codjc PM to use
direct-complete optimization"), which enables direct-complete for HDA
codec.
The HDA codec driver didn't expect direct-complete will be disabled
after it returns a positive value from prepare() callback. However,
there are some places that PM core can disable direct-complete. For
instance, system hibernation or when codec has subordinates like LEDs.
So if the codec is prepared for direct-complete but PM core still calls
codec's suspend or freeze callback, partially revert the commit and take
the original approach, which uses pm_runtime_force_*() helpers to
ensure PM refcount are balanced. Meanwhile, still keep prepare() and
complete() callbacks to enable direct-complete and request a resume for
jack detection, respectively.
Reported-by: Kenneth R. Crudup <kenny@panix.com>
Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210119152145.346558-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.11
A few more fixes for v5.11, mostly around HDA jack detection, plus
a couple of updates to the MAINTAINERS entries.
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For addressing the regression on Pioneer devices, we recently
corrected the quirk code to enable the implicit feedback mode on those
devices properly. However, the devices still showed problems with the
full duplex operations with JACK, and after debug sessions, we figured
out that the older kernels that had worked with JACK also didn't use
the implicit feedback mode at all although they had the quirk code to
enable it; instead, the old code worked just to skip the normal sync
endpoint setup that would have been detected without it. IOW, what
broke without the implicit-fb quirk in the past was the application of
the normal sync endpoint that is actually the capture data endpoint on
these devices.
This patch covers the overseen piece: it modifies the quirk code again
not to enable the implicit feedback mode but just to make the driver
skipping the sync endpoint detection. This made the driver working
with JACK full-duplex mode again.
Still it's not quite clear why the implicit feedback doesn't work on
those devices yet; maybe it's about some issues in the URB setup. But
at least, with this patch, the driver should work in the level of the
older kernels again.
Fixes: 167c9dc84ec3 ("ALSA: usb-audio: Fix implicit feedback sync setup for Pioneer devices")
Link: https://lore.kernel.org/r/20210118075816.25068-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The UAC2/3 sample rate setup is based on the clock node, which is
usually shared in the interface, and can't be re-setup without
deselecting the interface once, and that's how the current code
behaves. OTOH, the sample rate setup of UAC1 is per endpoint, hence
we basically need to call for each endpoint usage even if those share
the same interface.
This patch fixes the behavior of UAC1 to call always
snd_usb_init_sample_rate() in snd_usb_endpoint_configure().
Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210118075816.25068-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The current sample rate setup function for UAC1 assumes only the first
endpoint retrieved from the interface:altset pair, but the rate set up
may be needed also for the secondary endpoint. Also, retrieving the
endpoint number from the interface descriptor is redundant; we have
already the target endpoint in the given audioformat object.
This patch simplifies the code and corrects the target endpoint as
described in the above. It simply refers to fmt->endpoint directly.
Also, this patch drops the pioneer_djm_set_format_quirk() that is
caleld from snd_usb_set_format_quirk(); this function does the sample
rate setup but for the capture endpoint (0x82), and that's exactly
what the change above fixes.
Link: https://lore.kernel.org/r/20210118075816.25068-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add HD Audio Device PCI ID for the Intel Cometlake-R platform
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Signed-off-by: Kai-Chuan Hsieh <kaichuan.hsieh@canonical.com>
Link: https://lore.kernel.org/r/20210115031515.13100-1-kaichuan.hsieh@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_seq_oss_synth_make_info() didn't check the error code from
snd_seq_oss_midi_make_info(), and this leads to the call of strlcpy()
with the uninitialized string as the source, which may lead to the
access over the limit.
Add the proper error check for avoiding the failure.
Reported-by: syzbot+e42504ff21cff05a595f@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210115093428.15882-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It turned out that VIA codecs also mute the sound in the lowest mixer
level. Turn on the dac_min_mute flag to indicate the mute-as-minimum
in TLV like already done in Conexant and IDT codecs.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210559
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210114072453.11379-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer Apire E5-575T laptop with codec ALC255 has a terrible
background noise comes from internal mic capture. And the jack
sensing dose not work for headset like some other Acer laptops.
This patch limits the internal mic boost on top of the existing
ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk for Acer Aspire E5-575T.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210114082728.74729-1-chiu@endlessos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Since the commit 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for
implicit fb sync"), we apply the hw constraints for the implicit
feedback sync to make the secondary open aligned with the already
opened stream setup. This change assumed that the secondary open is
performed after the first stream has been already set up, and adds the
hw constraints to sync with the first stream's parameters only when
the EP setup for the first stream was confirmed at the open time.
However, most of applications handling the full-duplex operations do
open both playback and capture streams at first, then set up both
streams. This results in skipping the additional hw constraints since
the counter-part stream hasn't been set up yet at the open of the
second stream, and it eventually leads to "incompatible EP" error in
the end.
This patch corrects the behavior by always applying the hw constraints
for the implicit fb sync. The hw constraint rules are defined so that
they check the sync EP dynamically at each invocation, instead. This
covers the concurrent stream setups better and lets the hw refine
calls resolving to the right configuration.
Also this patch corrects a minor error that has existed in the debug
print that isn't built as default.
Fixes: 5a6c3e11c9c9 ("ALSA: usb-audio: Add hw constraint for implicit fb sync")
Link: https://lore.kernel.org/r/20210111081611.12790-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The earlier commit to fix runtime PM in case i915 init fails,
introduces a possibility to hit a page fault.
snd_hdac_ext_bus_device_exit() is designed to be called from
dev.release(). Calling it outside device reference counting, is
not safe and may lead to calling the device_exit() function
twice. Additionally, as part of ext_bus_device_init(), the device
is also registered with snd_hdac_device_register(). Thus before
calling device_exit(), the device must be removed from device
hierarchy first.
Fix the issue by rolling back init actions by calling
hdac_device_unregister() and then releasing device with put_device().
This matches with existing code in hdac-ext module.
To complete the fix, add handling for the case where
hda_codec_load_module() returns -ENODEV, and clean up the hdac_ext
resources also in this case.
In future work, hdac-ext interface should be extended to allow clients
more flexibility to handle the life-cycle of individual devices, beyond
just the current snd_hdac_ext_bus_device_remove(), which removes all
devices.
BugLink: https://github.com/thesofproject/linux/issues/2646
Reported-by: Jaroslav Kysela <perex@perex.cz>
Fixes: 6c63c954e1c5 ("ASoC: SOF: fix a runtime pm issue in SOF when HDMI codec doesn't work")
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Rander Wang <rander.wang@intel.com>
Reviewed-by: Libin Yang <libin.yang@intel.com>
Reviewed-by: Bard Liao <bard.liao@intel.com>
Link: https://lore.kernel.org/r/20210113150715.3992635-1-kai.vehmanen@linux.intel.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add HD Audio PCI ID and HDMI codec vendor ID for Intel AlderLake-P.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Link: https://lore.kernel.org/r/20210113155629.4097057-1-kai.vehmanen@linux.intel.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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System takes a very long time to suspend after commit 215a22ed31a1
("ALSA: hda: Refactor codec PM to use direct-complete optimization"):
[ 90.065964] PM: suspend entry (s2idle)
[ 90.067337] Filesystems sync: 0.001 seconds
[ 90.185758] Freezing user space processes ... (elapsed 0.002 seconds) done.
[ 90.188713] OOM killer disabled.
[ 90.188714] Freezing remaining freezable tasks ... (elapsed 0.001 seconds) done.
[ 90.190024] printk: Suspending console(s) (use no_console_suspend to debug)
[ 90.904912] intel_pch_thermal 0000:00:12.0: CPU-PCH is cool [49C], continue to suspend
[ 321.262505] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5
[ 328.426919] snd_hda_codec_realtek ehdaudio0D0: Unable to sync register 0x2b8000. -5
[ 329.490933] ACPI: EC: interrupt blocked
That commit keeps the codec suspended during the system suspend. However,
mute/micmute LED will clear codec's direct-complete flag by
dpm_clear_superiors_direct_complete().
This doesn't play well with SOF driver. When its runtime resume is
called for system suspend, hda_codec_jack_check() schedules
jackpoll_work which uses snd_hdac_is_power_on() to check whether codec
is suspended. Because the direct-complete path isn't taken,
pm_runtime_disable() isn't called so snd_hdac_is_power_on() returns
false and jackpoll continues to run, and snd_hda_power_up_pm() cannot
power up an already suspended codec in multiple attempts, causes the
long delay on system suspend:
if (dev->power.direct_complete) {
if (pm_runtime_status_suspended(dev)) {
pm_runtime_disable(dev);
if (pm_runtime_status_suspended(dev)) {
pm_dev_dbg(dev, state, "direct-complete ");
goto Complete;
}
pm_runtime_enable(dev);
}
dev->power.direct_complete = false;
}
When direct-complete path is taken, snd_hdac_is_power_on() returns true
and hda_jackpoll_work() is skipped by accident. So this is still not
correct.
If we were to use snd_hdac_is_power_on() in system PM path,
pm_runtime_status_suspended() should be used instead of
pm_runtime_suspended(), otherwise pm_runtime_{enable,disable}() may
change the outcome of snd_hdac_is_power_on().
Because devices suspend in reverse order (i.e. child first), it doesn't
make much sense to resume an already suspended codec from audio
controller. So avoid the issue by making sure jackpoll isn't used in
system PM process.
Fixes: 215a22ed31a1 ("ALSA: hda: Refactor codec PM to use direct-complete optimization")
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210112181128.1229827-3-kai.heng.feng@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Modify hda_codec_jack_wake_enable() to also support disable WAKEEN.
In addition, this patch also moves the WAKEEN disablement call out of
hda_codec_jack_check() into hda_codec_jack_wake_enable().
This is a preparation for next patch.
No functional change intended.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210112181128.1229827-2-kai.heng.feng@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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Instead of queueing jackpoll_work, runtime resume the codec to let it
use different jack detection methods based on jackpoll_interval.
This partially matches SOF driver's behavior with commit a6e7d0a4bdb0
("ALSA: hda: fix jack detection with Realtek codecs when in D3"), the
difference is SOF unconditionally resumes the codec.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Link: https://lore.kernel.org/r/20210112181128.1229827-1-kai.heng.feng@canonical.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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We are able to power down the GPU and audio via the GPU driver
so flag these asics as supporting runtime pm.
Reviewed-by: Evan Quan <evan.quan@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Link: https://lore.kernel.org/r/20210105175245.963451-1-alexander.deucher@amd.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As snd_fw_async_midi_port.consume_bytes is unsigned int, and
NSEC_PER_SEC is 1000000000L, the second multiplication in
port->consume_bytes * 8 * NSEC_PER_SEC / 31250
always overflows on 32-bit platforms, truncating the result. Fix this
by precalculating "NSEC_PER_SEC / 31250", which is an integer constant.
Note that this assumes port->consume_bytes <= 16777.
Fixes: 531f471834227d03 ("ALSA: firewire-lib/firewire-tascam: localize async midi port")
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20210111130251.361335-3-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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As snd_ff.rx_bytes[] is unsigned int, and NSEC_PER_SEC is 1000000000L,
the second multiplication in
ff->rx_bytes[port] * 8 * NSEC_PER_SEC / 31250
always overflows on 32-bit platforms, truncating the result. Fix this
by precalculating "NSEC_PER_SEC / 31250", which is an integer constant.
Note that this assumes ff->rx_bytes[port] <= 16777.
Fixes: 19174295788de77d ("ALSA: fireface: add transaction support")
Reviewed-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Geert Uytterhoeven <geert+renesas@glider.be>
Link: https://lore.kernel.org/r/20210111130251.361335-2-geert+renesas@glider.be
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Currently hda on tegra30 fails to open a stream with an input/output error.
For example:
speaker-test -Dhw:0,3 -c 2
speaker-test 1.2.2
Playback device is hw:0,3
Stream parameters are 48000Hz, S16_LE, 2 channels
Using 16 octaves of pink noise
Rate set to 48000Hz (requested 48000Hz)
Buffer size range from 64 to 16384
Period size range from 32 to 8192
Using max buffer size 16384
Periods = 4
was set period_size = 4096
was set buffer_size = 16384
0 - Front Left
Write error: -5,Input/output error
xrun_recovery failed: -5,Input/output error
Transfer failed: Input/output error
The tegra-hda device was introduced in tegra30 but only utilized in
tegra124 until recent chips. Tegra210/186 work only due to a hardware
change. For this reason it is unknown when this issue first manifested.
Discussions with the hardware team show this applies to all current tegra
chips. It has been resolved in the tegra234, which does not have hda
support at this time.
The explanation from the hardware team is this:
Below is the striping formula referenced from HD audio spec.
{ ((num_channels * bits_per_sample) / number of SDOs) >= 8 }
The current issue is seen because Tegra HW has a problem with boundary
condition (= 8) for striping. The reason why it is not seen on
Tegra210/Tegra186 is because it uses max 2SDO lines. Max SDO lines is
read from GCAP register.
For the given stream (channels = 2, bps = 16);
ratio = (channels * bps) / NSDO = 32 / NSDO;
On Tegra30, ratio = 32/4 = 8 (FAIL)
On Tegra210/186, ratio = 32/2 = 16 (PASS)
On Tegra194, ratio = 32/4 = 8 (FAIL) ==> Earlier workaround was
applied for it
If Tegra210/186 is forced to use 4SDO, it fails there as well. So the
behavior is consistent across all these chips.
Applying the fix in [1] universally resolves this issue on tegra30-hda.
Tested on the Ouya game console and the tf201 tablet.
[1] commit 60019d8c650d ("ALSA: hda/tegra: workaround playback failure on
Tegra194")
Reviewed-by: Jon Hunter <jonathanh@nvidia.com>
Tested-by: Ion Agorria <ion@agorria.com>
Reviewed-by: Sameer Pujar <spujar@nvidia.com>
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Peter Geis <pgwipeout@gmail.com>
Link: https://lore.kernel.org/r/20210108135913.2421585-3-pgwipeout@gmail.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Pioneer devices have both playback and capture streams sharing the
same iface/altsetting, and those need to be paired as implicit
feedback. Instead of a half-baked (and broken) static quirk entry,
set up more generically for those devices by checking the number of
endpoints and the attribute of the secondary EP.
Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-6-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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There are devices that have multiple endpoints sharing the same
iface/altset not only for sync but also for the actual streams, and
the audioformat for such an endpoint needs to be handled with the
proper endpoint index; otherwise it confuses the endpoint management.
This patch extends the audioformat to annotate the endpoint index, and
put the proper ep_idx=1 to Pioneer device quirk entries accordingly.
Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-5-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The current endpoint handling assumed (more or less) a unique 1:1
relation between the endpoint and the iface/altset. The exception was
the sync EP without the implicit feedback which has usually the
secondary EP of the same altset. This works fine for most devices,
but it turned out that some unusual devices like Pinoeer's ones have
both playback and capture endpoints in the same iface/altsetting and
use both for the implicit feedback mode. For handling such a case, we
need to extend the endpoint management to take the shared interface
into account.
This patch does that: it adds a new object snd_usb_iface_ref for
managing the reference counts of the each USB interface that is used
by each endpoint. The interface setup is performed only once for the
(sharing) endpoints, and the doubly initialization is avoided.
Along with this, the resource release of endpoints and interface
refcounts are put into a single function, snd_usb_endpoint_free_all()
instead of looping in the caller side.
Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management")
Link: https://lore.kernel.org/r/20210108075219.21463-4-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The implicit feedback mode needs to handle two endpoints and the
choice of the audioformat object for the sync EP is important since
this determines the compatibility of the hw_params. The current code
uses the same audioformat object if both the main EP and the sync EP
point to the same iface/altsetting. This was done in consideration of
the non-implicit-fb sync EP handling, and it doesn't match well with
the cases where actually to endpoints are defined in the sameiface /
altsetting like a few Pioneer devices.
Modify snd_usb_find_implicit_fb_sync_format() to pick up the
audioformat that is assigned in the counter-part substreams primarily,
so that the actual capture stream can be opened properly. We keep the
same audioformat object only as a fallback in case nothing found,
though.
Fixes: 9fddc15e8039 ("ALSA: usb-audio: Factor out the implicit feedback quirk code")
Link: https://lore.kernel.org/r/20210108075219.21463-3-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The recent change in the endpoint management moved the endpoint object
creation from the stream open time to the parser of the audio
descriptor. It works fine for the standard audio, but it overlooked
the other places that create audio streams via quirks
(QUIRK_AUDIO_FIXED_ENDPOINT) like the reported a few Pioneer devices;
those call snd_usb_add_audio_stream() manually, hence they miss the
endpoints, eventually resulting in the error at opening streams.
Moreover, now the sync EP setup was moved to the explicit call of
snd_usb_audioformat_set_sync_ep(), and this needs to be added for
those places, too.
This patch addresses those regressions for quirks. It adds a local
helper function add_audio_stream_from_fixed_fmt(), which does the all
needed tasks, and replaces the calls of snd_usb_add_audio_stream()
with this new function.
Fixes: 54cb31901b83 ("ALSA: usb-audio: Create endpoint objects at parsing phase")
Reported-by: František Kučera <konference@frantovo.cz>
Link: https://lore.kernel.org/r/20210108075219.21463-2-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Sound is broken on the DragonBoard 410c (apq8016_sbc) since 5.10:
hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: error at snd_soc_component_set_jack on hdmi-audio-codec.1.auto: -95
qcom-apq8016-sbc 7702000.sound: Failed to set jack: -95
ADV7533: ASoC: error at snd_soc_link_init on ADV7533: -95
hdmi-audio-codec hdmi-audio-codec.1.auto: ASoC: error at snd_soc_component_set_jack on hdmi-audio-codec.1.auto: -95
qcom-apq8016-sbc: probe of 7702000.sound failed with error -95
This happens because apq8016_sbc calls snd_soc_component_set_jack() on
all codec DAIs and attempts to ignore failures with return code -ENOTSUPP.
-ENOTSUPP is also excluded from error logging in soc_component_ret().
However, hdmi_codec_set_jack() returns -E*OP*NOTSUPP if jack detection
is not supported, which is not handled in apq8016_sbc and soc_component_ret().
Make it return -ENOTSUPP instead to fix sound and silence the errors.
Cc: Cheng-Yi Chiang <cychiang@chromium.org>
Cc: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Fixes: 55c5cc63ab32 ("ASoC: hdmi-codec: Use set_jack ops to set jack")
Signed-off-by: Stephan Gerhold <stephan@gerhold.net>
Acked-by: Nicolin Chen <nicoleotsuka@gmail.com>
Link: https://lore.kernel.org/r/20210107165131.2535-1-stephan@gerhold.net
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.11
A collection of mostly driver specific fixes, plus a maintainership
update for TI and a fix for DAPM driver removal paths.
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* The HP ZBook Fury 15/17 G7 Mobile Workstation are using ALC285 codec
which is using 0x04 to control mute LED and 0x01 to control micmute LED.
* The right channel speaker is no sound and it needs to expose GPIO1 for
initialing AMP.
Add quirks to support them.
Signed-off-by: Jeremy Szu <jeremy.szu@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210106130549.100532-1-jeremy.szu@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here is a collection of USB- and HD-audio fixes.
Most of them are device-specific quirks while one fix is for a
regression due to an incorrect mutex unlock introduced in this merge
window"
* tag 'sound-5.11-rc3' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/via: Fix runtime PM for Clevo W35xSS
ALSA: usb-audio: Add quirk for RC-505
ALSA: hda/hdmi: Fix incorrect mutex unlock in silent_stream_disable()
ALSA: hda/realtek: Enable mute and micmute LED on HP EliteBook 850 G7
ALSA: hda/realtek: Add two "Intel Reference board" SSID in the ALC256.
ALSA: hda/realtek: Add mute LED quirk for more HP laptops
ALSA: hda/conexant: add a new hda codec CX11970
ALSA: usb-audio: Add quirk for BOSS AD-10
ALSA: usb-audio: Fix UBSAN warnings for MIDI jacks
ALSA: hda/realtek - Modify Dell platform name
ALSA: hda/realtek - Fix speaker volume control on Lenovo C940
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Clevo W35xSS_370SS with VIA codec has had the runtime PM problem that
looses the power state of some nodes after the runtime resume. This
was worked around by disabling the default runtime PM via a denylist
entry. Since 5.10.x made the runtime PM applied (casually) even
though it's disabled in the denylist, this problem was revisited. The
result was that disabling power_save_node feature suffices for the
runtime PM problem.
This patch implements the disablement of power_save_node feature in
VIA codec for the device. It also drops the former denylist entry,
too, as the runtime PM should work in the codec side properly now.
Fixes: b529ef2464ad ("ALSA: hda: Add Clevo W35xSS_370SS to the power_save blacklist")
Reported-by: Christian Labisch <clnetbox@gmail.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20210104153046.19993-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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BOSS RC-505 (shown by lsusb as "Roland Corp. RC-505") does require the
same quirk as these other BOSS devices.
Without this quirk it is neither possible to capture audio from nor to
write audio to the RC-505. Both just result in an empty audio
stream. With these changes both capture and playback seem to work
quite fine. MIDI funtionality was not tested.
Tested-by: Harry Reinold <harry.reinold@posteo.de>
Signed-off-by: Timon Reinold <tirei@agon.one>
Link: https://lore.kernel.org/r/20210102210835.21268-1-tirei@agon.one
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The silent_stream_disable() function introduced by the commit
b1a5039759cb ("ALSA: hda/hdmi: fix silent stream for first playback to
DP") takes the per_pin->lock mutex, but it unlocks the wrong one,
spec->pcm_lock, which causes a deadlock. This patch corrects it.
Fixes: b1a5039759cb ("ALSA: hda/hdmi: fix silent stream for first playback to DP")
Reported-by: Jan Alexander Steffens (heftig) <heftig@archlinux.org>
Cc: <stable@vger.kernel.org>
Acked-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Link: https://lore.kernel.org/r/20210101083852.12094-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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HP EliteBook 850 G7 uses the same GPIO pins as ALC285_FIXUP_HP_GPIO_LED
to enable mute and micmute LED. So apply the quirk to enable the LEDs.
Signed-off-by: Kai-Heng Feng <kai.heng.feng@canonical.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201230125636.45028-1-kai.heng.feng@canonical.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Add two "Intel Reference boad" SSID in the alc256.
Enable "power saving mode" and Enable "headset jack mode".
Signed-off-by: PeiSen Hou <pshou@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/5978d2267f034c28973d117925ec9c63@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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HP Pavilion 13-bb0000 (SSID 103c:87c8) needs the same
quirk as other models with ALC287.
Signed-off-by: Manuel Jiménez <mjbfm99@me.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/X+s/gKNydVrI6nLj@HP-Pavilion-13
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The current kernel does not support the cx11970 codec chip.
Add a codec configuration item to kernel.
[ Minor coding style fix by tiwai ]
Signed-off-by: bo liu <bo.liu@senarytech.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201229035226.62120-1-bo.liu@senarytech.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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BOSS AD-10 requires the very same quirk like other BOSS devices to
enable the special implicit feedback mode.
Reported-and-tested-by: Martin Passing <martin@passing.name>
Link: https://lore.kernel.org/r/20201229083428.20467-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ThinkPad X395 latop does not have the internal digital
microphone connected to the AMD's ACP bridge, but it's advertised
via BIOS. The internal microphone is connected to the HDA codec.
Use DMI to block the microphone PCM device for this platform.
BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1892115
Cc: <stable@kernel.org>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20201227164109.269973-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
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When we try to play and capture simultaneously we see that
interrupts are genrated but our handler is not being acknowledged,
After investigating further more in detail on this issue we found
that IRQ delivery via MSI from the ACP IP is unreliable and so sometimes
interrupt generated will not be acknowledged so MSI model shouldn't be used
and using legacy IRQs will resolve interrupt handling issue.
This patch replaces MSI interrupt handling with legacy IRQ model.
Issue can be reproduced easily by running below python script:
import subprocess
import time
import threading
def do2():
cmd = 'aplay -f dat -D hw:2,1 /dev/zero -d 1'
subprocess.call(cmd, stdin=subprocess.PIPE,
stderr=subprocess.PIPE, shell=True)
print('Play Done')
def run():
for i in range(1000):
do2()
def do(i):
cmd = 'arecord -f dat -D hw:2,2 /dev/null -d 1'
subprocess.call(cmd, stdout=subprocess.PIPE,
stderr=subprocess.PIPE, shell=True)
print(datetime.datetime.now(), i)
t = threading.Thread(target=run)
t.start()
for i in range(1000):
do(i)
t.join()
After applying this patch issue is resolved.
Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati@amd.com>
Link: https://lore.kernel.org/r/20201222115929.11222-1-Vishnuvardhanrao.Ravulapati@amd.com
Signed-off-by: Mark Brown <broonie@kernel.org>
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The ThinkPad E14 Gen 2 latop does not have the internal digital
microphone connected to the AMD's ACP bridge, but it's advertised
via BIOS. The internal microphone is connected to the HDA codec.
Use DMI to block the microphone PCM device for this platform.
Reported-by: Eliot Blennerhassett <eliot@blennerhassett.gen.nz>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Link: https://lore.kernel.org/r/20201227164037.269893-1-perex@perex.cz
Signed-off-by: Mark Brown <broonie@kernel.org>
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Linux 5.11-rc1
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A collection of small fixes that came up recently for 5.11.
The majority of fixes are usual HD-audio and USB-audio quirks, with a
few PCM core fixes for addressing the information leak and yet more
UBSAN fixes in the core side"
* tag 'sound-fix-5.11-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA/hda: apply jack fixup for the Acer Veriton N4640G/N6640G/N2510G
ALSA: hda/realtek: Apply jack fixup for Quanta NL3
ALSA: usb-audio: Add implicit feeback support for the BOSS GT-1
ALSA: usb-audio: Add alias entry for ASUS PRIME TRX40 PRO-S
ALSA: core: Remove redundant comments
ALSA: hda/realtek: Add quirk for MSI-GP73
ALSA: pcm: oss: Fix a few more UBSAN fixes
ALSA: pcm: Clear the full allocated memory at hw_params
ALSA: memalloc: Align buffer allocations in page size
ALSA: usb-audio: Disable sample read check if firmware doesn't give back
ALSA: pcm: Remove snd_pcm_lib_preallocate_dma_free()
ALSA: usb-audio: Add VID to support native DSD reproduction on FiiO devices
ALSA: core: memalloc: add page alignment for iram
ALSA: hda/realtek - Supported Dell fixed type headset
ALSA: hda/realtek: Remove dummy lineout on Acer TravelMate P648/P658
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The calculation of in_cables and out_cables bitmaps are done with the
bit shift by the value from the descriptor, which is an arbitrary
value, and can lead to UBSAN shift-out-of-bounds warnings.
Fix it by filtering the bad descriptor values with the check of the
upper bound 0x10 (the cable bitmaps are 16 bits).
Reported-by: syzbot+92e45ae45543f89e8c88@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201223174557.10249-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Dell platform SSID:0x0a58 change platform name.
Use the generic name instead for avoiding confusion.
Fixes: 150927c3674d ("ALSA: hda/realtek - Supported Dell fixed type headset")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/efe7c196158241aa817229df7835d645@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Cannot adjust speaker's volume on Lenovo C940.
Applying the alc298_fixup_speaker_volume function can fix the issue.
[ Additional note: C940 has I2S amp for the speaker and this needs the
same initialization as Dell machines.
The patch was slightly modified so that the quirk entry is moved
next to the corresponding Dell quirk entry. -- tiwai ]
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/ea25b4e5c468491aa2e9d6cb1f2fced3@realtek.com
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This Acer Veriton N4640G/N6640G/N2510G desktops have 2 headphone
jacks(front and rear), and a separate Mic In jack.
The rear headphone jack is actually a line out jack but always silent
while playing audio. The front 'Mic In' also fails the jack sensing.
Apply the ALC269_FIXUP_LIFEBOOK to have all audio jacks to work as
expected.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Signed-off-by: Jian-Hong Pan <jhp@endlessos.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201222150459.9545-2-chiu@endlessos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Quanta NL3 laptop has both a headphone output jack and a headset
jack, on the right edge of the chassis.
The pin information suggests that both of these are at the Front.
The PulseAudio is confused to differentiate them so one of the jack
can neither get the jack sense working nor the audio output.
The ALC269_FIXUP_LIFEBOOK chained with ALC269_FIXUP_QUANTA_MUTE can
help to differentiate 2 jacks and get the 'Auto-Mute Mode' working
correctly.
Signed-off-by: Chris Chiu <chiu@endlessos.org>
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201222150459.9545-1-chiu@endlessos.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The BOSS GT-1 (USB ID 0582:01d6) requires implicit feedback
like other similar BOSS devices. This patch adds this support.
[ rearranged the table entry in the ID order -- tiwai ]
Signed-off-by: Mike Oliphant <oliphant@nostatic.org>
Link: https://lore.kernel.org/r/20201221215533.2511-1-oliphant@nostatic.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ASUS PRIME TRX40 PRO-S mobo with 0b05:1918 needs the same quirk alias
for another ASUS mobo (0b05:1917) for the proper mixer mapping, etc.
Add the corresponding entry.
BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=210783
Cc: <stable@vger.kernel.org>
Link: https://lore.kernel.org/r/20201221080159.24468-1-tiwai@suse.de
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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