| Commit message (Collapse) | Author | Age | Files | Lines |
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The current implementation of the hdac_hda codec results in zero-valued
samples on capture and noise with headset playback when SOF is used on
platforms with an on-board HDaudio codec. This is root-caused to SOF
using be_hw_params_fixup, and the prepare() call using invalid runtime
fields to determine the format.
This patch moves the format handling to the hw_params() callback, as
done already for hdac_hdmi, to make sure the fixed-up information is
taken into account but keeps the codec initialization in prepare() as
the stream_tag is only available at that time. Moving everything in the
prepare() callback is possible but the code is less elegant so this
two-step solution was chosen.
The solution was tested with the SST driver with no regressions, and all
the issues with SOF playback and capture are solved.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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On HDaudio platforms, if playback is started when capture is working,
there is no audible output.
This can be root-caused to the use of the rx|tx_mask to store an HDaudio
stream tag.
If capture is stared before playback, rx_mask would be non-zero on HDaudio
platform, then the channel number of playback, which is in the same codec
dai with the capture, would be changed by soc_pcm_codec_params_fixup based
on the tx_mask at first, then overwritten by this function based on rx_mask
at last.
According to the author of tx|rx_mask, tx_mask is for playback and rx_mask
is for capture. And stream direction is checked at all other references of
tx|rx_mask in ASoC, so here should be an error. This patch checks stream
direction for tx|rx_mask for fixup function.
This issue would affect not only HDaudio+ASoC, but also I2S codecs if the
channel number based on rx_mask is not equal to the one for tx_mask. It could
be rarely reproduecd because most drivers in kernel set the same channel number
to tx|rx_mask or rx_mask is zero.
Tested on all platforms using stream_tag & HDaudio and intel I2S platforms.
Signed-off-by: Rander Wang <rander.wang@linux.intel.com>
Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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This patch sets missing stream_name of capture part of the DAI driver
so we can define DAPM routing properly also for the capture stream.
While at it "Playback" suffix is added to the playback stream names
to clearly identify playback/capture.
Together with related dts patch this fixes NULL pointer dereference
when opening ALSA device for recording on Odroid XU3.
Fixes: 64aba9bca5bd ("ASoC: samsung: i2s: Add widgets and routes for DPCM support")
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Commit 78a24e10cd94 ("ASoC: soc-core: clear platform pointers on error")
re-worked the clean-up of any platform pointers that may have been
initialised by the function snd_soc_init_platform(). This commit missed
one error path where if any of the prelinks for a soundcard failed to
initialise, then these platform pointers would not be cleaned-up. This
then prevents the soundcard from being initialised following a probe
deferral when any of the soundcard prelinks cannot be found.
Fix this by ensuring that soc_cleanup_platform() is called when
initialising the soundcard prelinks fails.
Fixes: 78a24e10cd94 ("ASoC: soc-core: clear platform pointers on error")
Signed-off-by: Jonathan Hunter <jonathanh@nvidia.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Limiting the value of the passed in params->msbits in the hw_params()
callback is redundant on three counts:
1. We already specify in the DAI driver that we can only handle up to
24 bits. This means msbits will be limited to 24 via the ALSA
constraints imposed by the ASoC core, unless we have multiple codecs
that can handle more bits.
2. Nothing in our hw_params() implementation uses this value.
3. The copy of the params that we are passed by the ASoC core never
reads back the msbits value.
Consequently, this code is unnecessary and does nothing useful. Remove
it.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-5.1
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This patch fixes a bug that prevents freeing the reset gpio on unloading
the module.
aic3x_i2c_probe is called when loading the module and it calls list_add
with a probably uninitialized list entry aic3x->list (next = prev = NULL)).
So even if list_del is called it does nothing and in the end the gpio_reset
is not freed. Then a repeated module probing fails silently because
gpio_request fails.
When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move
list_del to aic3x_i2c_remove because aic3x_remove may be called
multiple times without aic3x_i2c_remove being called which leads to
a NULL pointer dereference.
Signed-off-by: Philipp Puschmann <philipp.puschmann@emlix.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Add error check on set_sync function return.
Add of_node_put() as of_get_parent() takes a reference
which has to be released.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Set OSR bit if mclk/fs ratio is 512.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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When snd_pcm_stop_xrun() is called in interrupt routine,
substream context may have already been released.
Add protection on substream context.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Change capabilities exposed in SAI S/PDIF mode, to match
actually supported formats.
In S/PDIF mode only 32 bits stereo is supported.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Allow indexation of sai iec958 controls according
to device id.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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In preparation to enabling -Wimplicit-fallthrough, mark switch
cases where we are expecting to fall through.
This patch fixes the following warning:
In file included from sound/soc/codecs/ab8500-codec.c:24:
sound/soc/codecs/ab8500-codec.c: In function ‘ab8500_codec_set_dai_fmt’:
./include/linux/device.h:1485:2: warning: this statement may fall through [-Wimplicit-fallthrough=]
_dev_err(dev, dev_fmt(fmt), ##__VA_ARGS__)
^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
sound/soc/codecs/ab8500-codec.c:2129:3: note: in expansion of macro ‘dev_err’
dev_err(dai->component->dev,
^~~~~~~
sound/soc/codecs/ab8500-codec.c:2132:2: note: here
default:
^~~~~~~
Warning level 3 was used: -Wimplicit-fallthrough=3
This patch is part of the ongoing efforts to enable
-Wimplicit-fallthrough.
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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When using the S/PDIF DAI, there is no requirement to call
snd_soc_dai_set_fmt() as there is no DAI format definition that defines
S/PDIF. In any case, S/PDIF does not have separate clocks, this is
embedded into the data stream.
Consequently, when attempting to use TDA998x in S/PDIF mode, the attempt
to configure TDA998x via the hw_params callback fails as the
hdmi_codec_daifmt is left initialised to zero.
Since the S/PDIF DAI will only be used by S/PDIF, prepare the
hdmi_codec_daifmt structure for this format.
Signed-off-by: Russell King <rmk+kernel@armlinux.org.uk>
Reviewed-by: Jyri Sarha <jsarha@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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add direct loopback path from rx to tx
Signed-off-by: KaiChieh Chuang <kaichieh.chuang@mediatek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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snd_hdac_display_power() doesn't handle the concurrent calls carefully
enough, and it may lead to the doubly get_power or put_power calls,
when a runtime PM and an async work get called in racy way.
This patch addresses it by reusing the bus->lock mutex that has been
used for protecting the link state change in ext bus code, so that it
can protect against racy display state changes. The initialization of
bus->lock was moved from snd_hdac_ext_bus_init() to
snd_hdac_bus_init() as well accordingly.
Testcase: igt/i915_pm_rpm/module-reload #glk-dsi
Reported-by: Chris Wilson <chris@chris-wilson.co.uk>
Reviewed-by: Chris Wilson <chris@chris-wilson.co.uk>
Cc: Imre Deak <imre.deak@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Recently we set CONFIG_SND_HDA_POWER_SAVE_DEFAULT to 1 when
configuring the kernel, then two machines were reported to have noise
after installing the new kernel. Put them in the blacklist, the
noise disappears.
https://bugs.launchpad.net/bugs/1821663
Cc: <stable@vger.kernel.org>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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When ioctl calls are made with non-null-terminated userspace strings,
strlcpy causes an OOB-read from within strlen. Fix by changing to use
strscpy instead.
Signed-off-by: Zubin Mithra <zsm@chromium.org>
Reviewed-by: Guenter Roeck <groeck@chromium.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This fixes the regression introduced while moving to Xen shared
buffer implementation.
Fixes: 58f9d806d16a ("ALSA: xen-front: Use Xen common shared buffer implementation")
Reviewed-by: Juergen Gross <jgross@suse.com>
Signed-off-by: Oleksandr Andrushchenko <oleksandr_andrushchenko@epam.com>
Cc: <stable@vger.kernel.org> # v5.0+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This adds a SND_PCI_QUIRK(...) line for the Tuxedo XC 1509.
The Tuxedo XC 1509 and the System76 oryp5 are the same barebone
notebooks manufactured by Clevo. To name the fixups both use after the
actual underlying hardware, this patch also changes System76_orpy5
to clevo_pb51ed in 2 enum symbols and one function name,
matching the other pci_quirk entries which are also named after the
device ODM.
Fixes: 7f665b1c3283 ("ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5")
Signed-off-by: Richard Sailer <rs@tuxedocomputers.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It will be lose Mic JD state when Chrome OS boot and headset was plugged.
Just Implement of reset combo jack JD verb for ACT_PRE_PROBE state.
Intel test result was also failed.
It test passed until changed the initial state to ACT_INIT.
Mic JD will show every time.
This patch also changed the model name as 'alc-chrome-book' for
application of Chrome OS.
Fixes: 10f5b1b85ed1 ("ALSA: hda/realtek - Fixed Headset Mic JD not stable")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer TravelMate B114-21 laptop cannot detect and record sound from
headset MIC. This patch adds the ALC233_FIXUP_ACER_HEADSET_MIC HDA verb
quirk chained with ALC233_FIXUP_ASUS_MIC_NO_PRESENCE pin quirk to fix
this issue.
[ fixed the missing brace and reordered the entry -- tiwai ]
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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On an Acer Predator Helios 500 (Ryzen version), the laptop's speakers
don't work out of the box.
The problem can be worked around with hdajackretask, remapping the
"Black Headphone, Right side" pin (0x21) to the Internal speaker.
This patch adds a quirk to change this mapping by default.
[ corrected ALC299_FIXUP_PREDATOR_SPK definition and adapted for the
latest tree by tiwai ]
Signed-off-by: Bernhard Rosenkraenzer <bero@lindev.ch>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Currently PCM core sets each opened stream forcibly to SUSPENDED state
via snd_pcm_suspend_all() call, and the user-space is responsible for
re-triggering the resume manually either via snd_pcm_resume() or
prepare call. The scheme works fine usually, but there are corner
cases where the stream can't be resumed by that call: the streams
still in OPEN state before finishing hw_params. When they are
suspended, user-space cannot perform resume or prepare because they
haven't been set up yet. The only possible recovery is to re-open the
device, which isn't nice at all. Similarly, when a stream is in
DISCONNECTED state, it makes no sense to change it to SUSPENDED
state. Ditto for in SETUP state; which you can re-prepare directly.
So, this patch addresses these issues by filtering the PCM streams to
be suspended by checking the PCM state. When a stream is in either
OPEN, SETUP or DISCONNECTED as well as already SUSPENDED, the suspend
action is skipped.
To be noted, this problem was originally reported for the PCM runtime
PM on HD-audio. And, the runtime PM problem itself was already
addressed (although not intended) by the code refactoring commits
3d21ef0b49f8 ("ALSA: pcm: Suspend streams globally via device type PM
ops") and 17bc4815de58 ("ALSA: pci: Remove superfluous
snd_pcm_suspend*() calls"). These commits eliminated the
snd_pcm_suspend*() calls from the runtime PM suspend callback code
path, hence the racy OPEN state won't appear while runtime PM.
(FWIW, the race window is between snd_pcm_open_substream() and the
first power up in azx_pcm_open().)
Although the runtime PM issue was already "fixed", the same problem is
still present for the system PM, hence this patch is still needed.
And for stable trees, this patch alone should suffice for fixing the
runtime PM problem, too.
Reported-and-tested-by: Jon Hunter <jonathanh@nvidia.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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ca0132 codec driver loads the firmware selectively depending on the
model in addition to the fallback of the default firmware. The code
works good, but a minor problem is that the current code seems
confusing for Clang where it spews a warning about uninitialized
variable.
This patch simplifies the code flow for such a false-positive
warning. After this refactoring, the ca0132_spec.alt_firmware_present
field is no longer used, hence it's eliminated as well.
Reported-and-tested-by: Arnd Bergmann <arnd@arndb.de>
Reviewed-by: Nathan Chancellor <natechancellor@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The PCM OSS emulation converts and transfers the data on the fly via
"plugins". The data is converted over the dynamically allocated
buffer for each plugin, and recently syzkaller caught OOB in this
flow.
Although the bisection by syzbot pointed out to the commit
65766ee0bf7f ("ALSA: oss: Use kvzalloc() for local buffer
allocations"), this is merely a commit to replace vmalloc() with
kvmalloc(), hence it can't be the cause. The further debug action
revealed that this happens in the case where a slave PCM doesn't
support only the stereo channels while the OSS stream is set up for a
mono channel. Below is a brief explanation:
At each OSS parameter change, the driver sets up the PCM hw_params
again in snd_pcm_oss_change_params_lock(). This is also the place
where plugins are created and local buffers are allocated. The
problem is that the plugins are created before the final hw_params is
determined. Namely, two snd_pcm_hw_param_near() calls for setting the
period size and periods may influence on the final result of channels,
rates, etc, too, while the current code has already created plugins
beforehand with the premature values. So, the plugin believes that
channels=1, while the actual I/O is with channels=2, which makes the
driver reading/writing over the allocated buffer size.
The fix is simply to move the plugin allocation code after the final
hw_params call.
Reported-by: syzbot+d4503ae45b65c5bc1194@syzkaller.appspotmail.com
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ASUS X430UN and X512DK with ALC256 cannot detect the headset MIC
until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ASUS laptop P5440FF with ALC256 can't detect the headset microphone
until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied.
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>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ASUS laptop X441MB and X705FD with ALC256 cannot detect the headset
MIC until ALC256_FIXUP_ASUS_MIC_NO_PRESENCE quirk applied.
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>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer laptop Aspire E5-523G and ES1-432 with ALC255 can't detect
the headset microphone until ALC255_FIXUP_ACER_MIC_NO_PRESENCE quirk
applied.
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>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer Aspire Z24-890 cannot detect the headset MIC until
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk applied.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Daniel Drake <drake@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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dev is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/core/seq/oss/seq_oss_synth.c:626 snd_seq_oss_synth_make_info() warn: potential spectre issue 'dp->synths' [w] (local cap)
Fix this by sanitizing dev before using it to index dp->synths.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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info->stream is indirectly controlled by user-space, hence leading to
a potential exploitation of the Spectre variant 1 vulnerability.
This issue was detected with the help of Smatch:
sound/core/rawmidi.c:604 __snd_rawmidi_info_select() warn: potential spectre issue 'rmidi->streams' [r] (local cap)
Fix this by sanitizing info->stream before using it to index
rmidi->streams.
Notice that given that speculation windows are large, the policy is
to kill the speculation on the first load and not worry if it can be
completed with a dependent load/store [1].
[1] https://lore.kernel.org/lkml/20180423164740.GY17484@dhcp22.suse.cz/
Cc: stable@vger.kernel.org
Signed-off-by: Gustavo A. R. Silva <gustavo@embeddedor.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Some Acer AIO desktops like Veriton Z6860G, Z4860G and Z4660G cannot
record sound from headset MIC. This patch adds the
ALC286_FIXUP_ACER_AIO_HEADSET_MIC quirk to fix this issue.
Fixes: 9f8aefed9623 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4660G")
Fixes: b72f936f6b32 ("ALSA: hda/realtek: Fix mic issue on Acer AIO Veriton Z4860G/Z6860G")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Reviewed-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Recently we found the audio jack detection stop working after suspend
on many machines with Realtek codec. Sometimes the audio selection
dialogue didn't show up after users plugged headhphone/headset into
the headset jack, sometimes after uses plugged headphone/headset, then
click the sound icon on the upper-right corner of gnome-desktop, it
also showed the speaker rather than the headphone.
The root cause is that before suspend, the codec already call the
runtime_suspend since this codec is not used by any apps, then in
resume, it will not call runtime_resume for this codec. But for some
realtek codec (so far, alc236, alc255 and alc891) with the specific
BIOS, if it doesn't run runtime_resume after suspend, all codec
functions including jack detection stop working anymore.
This problem existed for a long time, but it was not exposed, that is
because when problem happens, if users play sound or open
sound-setting to check audio device, this will trigger calling to
runtime_resume (via snd_hda_power_up), then the codec starts working
again before users notice this problem.
Since we don't know how many codec and BIOS combinations have this
problem, to fix it, let the driver call runtime_resume for all codecs
in pm_resume, maybe for some codecs, this is not needed, but it is
harmless. After a codec is runtime resumed, if it is not used by any
apps, it will be runtime suspended soon and furthermore we don't run
suspend frequently, this change will not add much power consumption.
Fixes: cc72da7d4d06 ("ALSA: hda - Use standard runtime PM for codec power-save control")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The commit 3baffc4a84d7 (ALSA: hda/intel: Refactoring PM code) changed
the behaviour of azx_resume(), it triggers the jackpoll_work after
applying this commit.
This change introduced a new issue, all codecs are runtime active
after S3, and will not call runtime_suspend() automatically.
The root cause is the jackpoll_work calls snd_hda_power_up/down_pm,
and it calls up_pm before snd_hdac_enter_pm is called, while calls
the down_pm in the middle of enter_pm and leave_pm is called. This
makes the dev->power.usage_count unbalanced after S3.
To fix it, let azx_resume() don't trigger jackpoll_work as before
it did.
Fixes: 3baffc4a84d7 ("ALSA: hda/intel: Refactoring PM code")
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The function snd_opl3_drum_switch declaration in the header file
has the order of the two arguments on_off and vel swapped when
compared to the definition arguments of vel and on_off. Fix this
by swapping them around to match the definition.
This error predates the git history, so no idea when this error
was introduced.
Signed-off-by: Colin Ian King <colin.king@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Another machine which does not like the power saving (noise):
https://bugzilla.redhat.com/show_bug.cgi?id=1689623
Also, reorder the Lenovo C50 entry to keep the table sorted.
Reported-by: hs.guimaraes@outlook.com
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Current ALSA firewire-motu driver uses the value of 'model' field
of unit directory in configuration ROM for modalias for MOTU
FireWire models. However, as long as I checked, Pre8 and
828mk3(Hybrid) have the same value for the field (=0x100800).
unit | version | model
--------------- | --------- | ----------
828mkII | 0x000003 | 0x101800
Traveler | 0x000009 | 0x107800
Pre8 | 0x00000f | 0x100800 <-
828mk3(FW) | 0x000015 | 0x106800
AudioExpress | 0x000033 | 0x104800
828mk3(Hybrid) | 0x000035 | 0x100800 <-
When updating firmware for MOTU 8pre FireWire from v1.0.0 to v1.0.3,
I got change of the value from 0x100800 to 0x103800. On the other
hand, the value of 'version' field is fixed to 0x00000f. As a quick
glance, the higher 12 bits of the value of 'version' field represent
firmware version, while the lower 12 bits is unknown.
By induction, the value of 'version' field represents actual model.
This commit changes modalias to match the value of 'version' field,
instead of 'model' field. For degug, long name of added sound card
includes hexadecimal value of 'model' field.
Fixes: 6c5e1ac0e144 ("ALSA: firewire-motu: add support for Motu Traveler")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Cc: <stable@vger.kernel.org> # v4.19+
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In case request_region fails, the fix returns an error code to
avoid NULL pointer dereference.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In case ioremap_nocache fails, the fix releases chip and returns
an error code upstream to avoid NULL pointer dereference.
Signed-off-by: Kangjie Lu <kjlu@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Enable headset mode support for new WYSE NB platform.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This patch will enable WYSE AIO for Headset mode.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The ALC225_FIXUP_HEADSET_JACK fixup can be merged to alc295_fixup_chromebook.
There are no other users for ALC225_FIXUP_HEADSET_JACK other than
the chromebook hardware.
Fixes: 10f5b1b85ed1 ("ALSA: hda/realtek - Fixed Headset Mic JD not stable")
Cc: Kailang Yang <kailang@realtek.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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This is just a port of the ASoC Icelake HDMI codec code to the legacy
HDA driver with some cleanups.
ASoC commit 019033c854a20e10f691f6cc0e897df8817d9521:
"ASoC: Intel: hdac_hdmi: add Icelake support"
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: Bard liao <bard.liao@intel.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Apply the HP_MIC_NO_PRESENCE fixups for the more HP Z2 G4 and
HP Z240 models.
Reported-by: Jeff Burrell <jeff.burrell@hp.com>
Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It will be lose Mic JD state when Chrome OS boot and headset was plugged.
Implement of reset combo jack JD. It will show normally.
Fixes: e854747d7593 ("ALSA: hda/realtek - Enable headset button support for new codec")
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The Acer TravelMate X514-51T with ALC255 cannot detect the headset MIC
until ALC255_FIXUP_ACER_HEADSET_MIC quirk applied. Although, the
internal DMIC uses another module - snd_soc_skl as the driver. We still
need the NID 0x1a in the quirk to enable the headset MIC.
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The #ifdef protection around the PM functions is wrong, leading to
a failed reference in some configurations:
sound/pci/hda/hda_tegra.c: In function 'hda_tegra_runtime_suspend':
sound/pci/hda/hda_tegra.c:273:2: error: implicit declaration of function 'hda_tegra_disable_clocks'; did you mean 'hda_tegra_enable_clocks'? [-Werror=implicit-function-declaration]
Better remove the #ifdefs entirely and rely on the compiler silently
dropping unused functions marked __maybe_unused.
Fixes: 707e0759f2f4 ("ALSA: hda/tegra: implement runtime suspend/resume")
Acked-by: Thierry Reding <treding@nvidia.com>
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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usb_alloc_urb() can fail due to kmalloc failure and push the error
upstream. Further this can cause a NULL pointer dereference in
init_pipe_urbs(). This patch avoids such a scenario.
Signed-off-by: Aditya Pakki <pakki001@umn.edu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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