| Commit message (Collapse) | Author | Age | Files | Lines |
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[ Upstream commit a0c426fe143328760c9fd565cd203a37a7b4fde8 ]
We will get DAI ID from "reg" property if it has on DT, otherwise get
it by counting port/endpoint.
But in below case, we need to get DAI ID = 0 via port reg = <0>, but
current implementation returns ID = 1, because it can't judge ID = 0 was
from "non reg" or "reg = <0>".
Thus, it will count port/endpoint number as "non reg" case.
of_graph_parse_endpoint() implementation itself is not a problem,
but because asoc_simple_card_get_dai_id() need to count port/endpoint
number when "non reg" case, it need to know ID = 0 was from
"non reg" or "reg = <0>".
This patch fix this issue.
port {
reg = <0>;
xxxx: endpoint@0 {
};
=> xxxx: endpoint@1 {
};
};
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit b2e9e1c8810ee05c95f4d55800b8afae70ab01b4 ]
Duende Classic was produced by Solid State Logic in 2006, as a
first model of Duende DSP series. The following model, Duende Mini
was produced in 2008. They are designed to receive isochronous
packets for PCM frames via IEEE 1394 bus, perform signal processing by
downloaded program, then transfer isochronous packets for converted
PCM frames.
These two models includes the same embedded board, consists of several
ICs below:
- Texus Instruments Inc, TSB41AB3 for physical layer of IEEE 1394 bus
- WaveFront semiconductor, DICE II STD ASIC for link/protocol layer
- Altera MAX 3000A CPLD for programs
- Analog devices, SHARC ADSP-21363 for signal processing (4 chips)
This commit adds support for the two models to ALSA dice driver. Like
support for the other devices, packet streaming is just available.
Userspace applications should be developed if full features became
available; e.g. program uploader and parameter controller.
$ ./hinawa-config-rom-printer /dev/fw1
{ 'bus-info': { 'adj': False,
'bmc': False,
'chip_ID': 349771402425,
'cmc': True,
'cyc_clk_acc': 255,
'generation': 1,
'imc': True,
'isc': True,
'link_spd': 2,
'max_ROM': 1,
'max_rec': 512,
'name': '1394',
'node_vendor_ID': 20674,
'pmc': False},
'root-directory': [ ['VENDOR', 20674],
['DESCRIPTOR', 'Solid State Logic'],
['MODEL', 112],
['DESCRIPTOR', 'Duende board'],
[ 'NODE_CAPABILITIES',
{ 'addressing': {'64': True, 'fix': True, 'prv': True},
'misc': {'int': False, 'ms': False, 'spt': True},
'state': { 'atn': False,
'ded': False,
'drq': True,
'elo': False,
'init': False,
'lst': True,
'off': False},
'testing': {'bas': False, 'ext': False}}],
[ 'UNIT',
[ ['SPECIFIER_ID', 20674],
['VERSION', 1],
['MODEL', 112],
['DESCRIPTOR', 'Duende board']]]]}
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 11907e9d3533648615db08140e3045b829d2c141 ]
The of_find_device_by_node() takes a reference to the underlying device
structure, we should release that reference.
Signed-off-by: Wen Yang <yellowriver2010@hotmil.com>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit d9c0b2afe820fa3b3f8258a659daee2cc71ca3ef ]
BE dai links only have internal PCM's and their substream ops may
not be set. Suspending these PCM's will result in their
ops->trigger() being invoked and cause a kernel oops.
So skip suspending PCM's if their ops are NULL.
[ NOTE: this change is required now for following the recent PCM core
change to get rid of snd_pcm_suspend() call. Since DPCM BE takes
the runtime carried from FE while keeping NULL ops, it can hit this
bug. See details at:
https://github.com/thesofproject/linux/pull/582
-- tiwai ]
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 8fa857da9744f513036df1c43ab57f338941ae7d ]
The of_find_device_by_node() takes a reference to the underlying device
structure, we should release that reference.
Detected by coccinelle with the following warnings:
./sound/soc/fsl/imx-sgtl5000.c:169:1-7: ERROR: missing put_device;
call of_find_device_by_node on line 105, but without a corresponding
object release within this function.
./sound/soc/fsl/imx-sgtl5000.c:177:1-7: ERROR: missing put_device;
call of_find_device_by_node on line 105, but without a corresponding
object release within this function.
Signed-off-by: Wen Yang <yellowriver2010@hotmail.com>
Cc: Timur Tabi <timur@kernel.org>
Cc: Nicolin Chen <nicoleotsuka@gmail.com>
Cc: Xiubo Li <Xiubo.Lee@gmail.com>
Cc: Fabio Estevam <festevam@gmail.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Cc: Shawn Guo <shawnguo@kernel.org>
Cc: Sascha Hauer <s.hauer@pengutronix.de>
Cc: Pengutronix Kernel Team <kernel@pengutronix.de>
Cc: NXP Linux Team <linux-imx@nxp.com>
Cc: alsa-devel@alsa-project.org
Cc: linuxppc-dev@lists.ozlabs.org
Cc: linux-arm-kernel@lists.infradead.org
Cc: linux-kernel@vger.kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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[ Upstream commit 70b773219a32c7b8f3e53e041bc023ad99fd81f4 ]
Although qcom_snd_parse_of() tries to manage the of-node refcount,
there are still a few places that lead to the unblanced refcount in
the error code path. Namely,
- for_each_child_of_node() needs to unreference the iterator node if
aborting the loop in the middle,
- cpu, codec and platform node objects have to be unreferenced at each
iteration,
- platform and codec node objects have to be referred before jumping
to the error handling code that unreference them unconditionally.
This patch tries to address these by moving the assignment of platform
and codec node objects to the beginning of the loop and adding the
of_node_put() calls adequately.
Fixes: c25e295cd77b ("ASoC: qcom: Add support to parse common audio device nodes")
Cc: Patrick Lai <plai@codeaurora.org>
Cc: Banajit Goswami <bgoswami@codeaurora.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Sasha Levin <sashal@kernel.org>
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commit e2a829b3da01b9b32c4d0291d042b8a6e2a98ca3 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 6ac371aa1a74240fb910c98aa3484d5ece8473d3 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit a806ef1cf3bbc0baadc6cdeb11f12b5dd27e91c2 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit e1037354a0a75acdea2b27043c0a371ed85cf262 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c7531e31c8a440b5fe6bd62664def5bcb6262f96 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2733ccebf4a937a0858e7d05a4a003b89715033f upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 667a8f73753908c4d0171e52b71774f9be5d6713 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit da484d00f020af3dd7cfcc6c4b69a7f856832883 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 136824efaab2c095fc911048f7c7ddeda258c965 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c8a9afa632f0fd45731d3353525faf1fdb362c89 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 10f5b1b85ed10a80d45bc2db450e65bd792efaad upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 113ce08109f8e3b091399e7cc32486df1cff48e7 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit ca0214ee2802dd47239a4e39fb21c5b00ef61b22 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c709f14f0616482b67f9fbcb965e1493a03ff30b upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2b1d9c8f87235f593826b9cf46ec10247741fff9 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit b5a236c175b0d984552a5f7c9d35141024c2b261 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 98081ca62cbac31fb0f7efaf90b2e7384ce22257 upstream.
Currently we deal with single codec and suspend codec callbacks for
all S3, S4 and runtime PM handling. But it turned out that we want
distinguish the call patterns sometimes, e.g. for applying some init
sequence only at probing and restoring from hibernate.
This patch slightly modifies the common PM callbacks for HD-audio
codec and stores the currently processed PM event in power_state of
the codec's device.power field, which is currently unused. The codec
callback can take a look at this event value and judges which purpose
it's being called.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 31d2350d602511efc9ef626b848fe521233b0387 upstream.
ac97_of_get_child_device() take the refcount of the node explicitly
via of_node_get(), but this leads to an unbalance. The
for_each_child_of_node() loop itself takes the refcount for each
iteration node, hence you don't need to take the extra refcount
again.
Fixes: 2225a3e6af78 ("ALSA: ac97: add codecs devicetree binding")
Reviewed-by: Robert Jarzmik <robert.jarzmik@free.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 744c67ffeb06f2d2493f4049ba0bd19698ce0adf upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2d012c65a9ca26a0ef87ea0a42f1653dd37155f5 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 721f1e6c1fd137e7e2053d8e103b666faaa2d50c upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 05bd7fcdd06b19a10f069af1bea3ad9abac038d7 upstream.
The ADCs are sleeping when the SLEEP bit is set and running when it's
cleared, so the bit should be inverted.
Tested on pcm1863.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit fcf4daabf08079e6d09958a2992e7446ef8d0438 upstream.
According to DS, the gain is between -12 dB and 40 dB, with a 0.5 dB step.
Tested on pcm1863.
Signed-off-by: Codrin Ciubotariu <codrin.ciubotariu@microchip.com>
Acked-by: Andrew F. Davis <afd@ti.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: stable@vger.kernel.org
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cc29ea007347f39f4c5a4d27b0b555955a0277f9 upstream.
The ESAI_xCR_xWA is xCR's bit, not the xCCR's bit, driver set it to
wrong register, correct it.
Fixes 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver")
Cc: <stable@vger.kernel.org>
Signed-off-by: Shengjiu Wang <shengjiu.wang@nxp.com>
Reviewed-by: Fabio Estevam <festevam@gmail.com>
Ackedy-by: Nicolin Chen <nicoleotsuka@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cbc05fd6708c1744ee6a61cb4c461ff956d30524 upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c0ca5eced22215c1e03e3ad479f8fab0bbb30772 upstream.
Dell Precision 5820 with ALC3234 codec (which is equivalent with
ALC255) shows click noises at (runtime) PM resume on the headphone.
The biggest source of the noise comes from the cleared headphone pin
control at resume, which is done via the standard shutup procedure.
Although we have an override of the standard shutup callback to
replace with NOP, this would skip other needed stuff (e.g. the pull
down of headset power). So, instead, this "fixes" the behavior of
alc_fixup_no_shutup() by introducing spec->no_shutup_pins flag.
When this flag is set, Realtek codec won't call the standard
snd_hda_shutup_pins() & co. Now alc_fixup_no_shutup() just sets this
flag instead of overriding spec->shutup callback itself. This allows
us to apply the similar fix for other entries easily if needed in
future.
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 8bb37a2a4d7c02affef554f5dc05f6d2e39c31f9 upstream.
The ASUS UX362FA with ALC294 cannot detect the headset MIC and outputs
through the internal speaker and the headphone. This issue can be fixed
by the quirk in the commit 4e0511067 ALSA: hda/realtek: Enable audio
jacks of ASUS UX533FD with ALC294.
Besides, ASUS UX362FA and UX533FD have the same audio initial pin config
values. So, this patch replaces SND_PCI_QUIRK of UX533FD with a new
SND_HDA_PIN_QUIRK which benefits both UX362FA and UX533FD.
Fixes: 4e051106730d ("ALSA: hda/realtek: Enable audio jacks of ASUS UX533FD with ALC294")
Signed-off-by: Jian-Hong Pan <jian-hong@endlessm.com>
Signed-off-by: Ming Shuo Chiu <chiu@endlessm.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 167897f4b32c2bc18b3b6183029a33fb420a114e upstream.
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>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cfc35f9c128cea8fce6a5513b1de50d36f3b209f upstream.
I set 10 seconds for the timeout of the i915 audio component binding
with a hope that recent machines are fast enough to handle all probe
tasks in that period, but I was too optimistic. The binding may take
longer than that, and this caused a problem on the machine with both
audio and graphics driver modules loaded in parallel, as Paul Menzel
experienced. This problem haven't hit so often just because the KMS
driver is loaded in initrd on most machines.
As a simple workaround, extend the timeout to 60 seconds.
Fixes: f9b54e1961c7 ("ALSA: hda/i915: Allow delayed i915 audio component binding")
Reported-by: Paul Menzel <pmenzel+alsa-devel@molgen.mpg.de>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit f97a0944a72b26a2bece72516294e112a890f98a upstream.
In data blocks of common isochronous packet for MOTU devices, PCM
frames are multiplexed in a shape of '24 bit * 4 Audio Pack', described
in IEC 61883-6. The frames are not aligned to quadlet.
For capture PCM substream, ALSA firewire-motu driver constructs PCM
frames by reading data blocks byte-by-byte. However this operation
includes bug for lower byte of the PCM sample. This brings invalid
content of the PCM samples.
This commit fixes the bug.
Reported-by: Peter Sjöberg <autopeter@gmail.com>
Cc: <stable@vger.kernel.org> # v4.12+
Fixes: 4641c9394010 ("ALSA: firewire-motu: add MOTU specific protocol layer")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Liquid Saffire 56
commit 7dc661bd8d3261053b69e4e2d0050cd1ee540fc1 upstream.
ALSA bebob driver has an entry for Focusrite Saffire Pro 10 I/O. The
entry matches vendor_id in root directory and model_id in unit
directory of configuration ROM for IEEE 1394 bus.
On the other hand, configuration ROM of Focusrite Liquid Saffire 56
has the same vendor_id and model_id. This device is an application of
TCAT Dice (TCD2220 a.k.a Dice Jr.) however ALSA bebob driver can be
bound to it randomly instead of ALSA dice driver. At present, drivers
in ALSA firewire stack can not handle this situation appropriately.
This commit uses more identical mod_alias for Focusrite Saffire Pro 10
I/O in ALSA bebob driver.
$ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 042a829d bus_info_length 4, crc_length 42, crc 33437
404 31333934 bus_name "1394"
408 f0649222 irmc 1, cmc 1, isc 1, bmc 1, pmc 0, cyc_clk_acc 100,
max_rec 9 (1024), max_rom 2, gen 2, spd 2 (S400)
40c 00130e01 company_id 00130e |
410 000606e0 device_id 01000606e0 | EUI-64 00130e01000606e0
root directory
-----------------------------------------------------------------
414 0009d31c directory_length 9, crc 54044
418 04000014 hardware version
41c 0c0083c0 node capabilities per IEEE 1394
420 0300130e vendor
424 81000012 --> descriptor leaf at 46c
428 17000006 model
42c 81000016 --> descriptor leaf at 484
430 130120c2 version
434 d1000002 --> unit directory at 43c
438 d4000006 --> dependent info directory at 450
unit directory at 43c
-----------------------------------------------------------------
43c 0004707c directory_length 4, crc 28796
440 1200a02d specifier id: 1394 TA
444 13010001 version: AV/C
448 17000006 model
44c 81000013 --> descriptor leaf at 498
dependent info directory at 450
-----------------------------------------------------------------
450 000637c7 directory_length 6, crc 14279
454 120007f5 specifier id
458 13000001 version
45c 3affffc7 (immediate value)
460 3b100000 (immediate value)
464 3cffffc7 (immediate value)
468 3d600000 (immediate value)
descriptor leaf at 46c
-----------------------------------------------------------------
46c 00056f3b leaf_length 5, crc 28475
470 00000000 textual descriptor
474 00000000 minimal ASCII
478 466f6375 "Focu"
47c 73726974 "srit"
480 65000000 "e"
descriptor leaf at 484
-----------------------------------------------------------------
484 0004a165 leaf_length 4, crc 41317
488 00000000 textual descriptor
48c 00000000 minimal ASCII
490 50726f31 "Pro1"
494 30494f00 "0IO"
descriptor leaf at 498
-----------------------------------------------------------------
498 0004a165 leaf_length 4, crc 41317
49c 00000000 textual descriptor
4a0 00000000 minimal ASCII
4a4 50726f31 "Pro1"
4a8 30494f00 "0IO"
$ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom
ROM header and bus information block
-----------------------------------------------------------------
400 040442e4 bus_info_length 4, crc_length 4, crc 17124
404 31333934 bus_name "1394"
408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255,
max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400)
40c 00130e04 company_id 00130e |
410 018001e9 device_id 04018001e9 | EUI-64 00130e04018001e9
root directory
-----------------------------------------------------------------
414 00065612 directory_length 6, crc 22034
418 0300130e vendor
41c 8100000a --> descriptor leaf at 444
420 17000006 model
424 8100000e --> descriptor leaf at 45c
428 0c0087c0 node capabilities per IEEE 1394
42c d1000001 --> unit directory at 430
unit directory at 430
-----------------------------------------------------------------
430 000418a0 directory_length 4, crc 6304
434 1200130e specifier id
438 13000001 version
43c 17000006 model
440 8100000f --> descriptor leaf at 47c
descriptor leaf at 444
-----------------------------------------------------------------
444 00056f3b leaf_length 5, crc 28475
448 00000000 textual descriptor
44c 00000000 minimal ASCII
450 466f6375 "Focu"
454 73726974 "srit"
458 65000000 "e"
descriptor leaf at 45c
-----------------------------------------------------------------
45c 000762c6 leaf_length 7, crc 25286
460 00000000 textual descriptor
464 00000000 minimal ASCII
468 4c495155 "LIQU"
46c 49445f53 "ID_S"
470 41464649 "AFFI"
474 52455f35 "RE_5"
478 36000000 "6"
descriptor leaf at 47c
-----------------------------------------------------------------
47c 000762c6 leaf_length 7, crc 25286
480 00000000 textual descriptor
484 00000000 minimal ASCII
488 4c495155 "LIQU"
48c 49445f53 "ID_S"
490 41464649 "AFFI"
494 52455f35 "RE_5"
498 36000000 "6"
Cc: <stable@vger.kernel.org> # v3.16+
Fixes: 25784ec2d034 ("ALSA: bebob: Add support for Focusrite Saffire/SaffirePro series")
Signed-off-by: Takashi Sakamoto <o-takashi@sakamocchi.jp>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"Here are a few last-minute fixes for 5.0.
The most significant one is the OF-node refcount fix for ASoC
simple-card, which could be triggered on many boards. Another fix for
ASoC core is for the error handling in topology, while others are
device-specific fixes for Samsung and HD-audio"
* tag 'sound-5.0' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ASoC: simple-card: fixup refcount_t underflow
ASoC: topology: free created components in tplg load error
ALSA: hda/realtek: Disable PC beep in passthrough on alc285
ALSA: hda/realtek - Headset microphone and internal speaker support for System76 oryp5
ASoC: samsung: i2s: Fix prescaler setting for the secondary DAI
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.0
A few small fixes, a driver fix for Samsung, a fix for refcounting of
of_nodes in the simple-card driver that triggered on a lot of systems
and a fix for topology error handling.
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commit da215354eb55c ("ASoC: simple-card: merge simple-scu-card")
merged simple-card and simple-scu-card. Then it had refcount
underflow bug. This patch fixup it.
We will get below error without this patch.
OF: ERROR: Bad of_node_put() on /sound
CPU: 3 PID: 237 Comm: kworker/3:1 Not tainted 5.0.0-rc6+ #1514
Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT)
Workqueue: events deferred_probe_work_func
Call trace:
dump_backtrace+0x0/0x150
show_stack+0x24/0x30
dump_stack+0xb0/0xec
of_node_release+0xd0/0xd8
kobject_put+0x74/0xe8
of_node_put+0x24/0x30
__of_get_next_child+0x50/0x70
of_get_next_child+0x40/0x68
asoc_simple_card_probe+0x604/0x730
platform_drv_probe+0x58/0xa8
...
Reported-by: Vicente Bergas <vicencb@gmail.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Topology resources are no longer needed if any element failed to load.
Signed-off-by: Bard liao <yung-chuan.liao@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Make sure i2s->rclk_srcrate is properly initialized also during
playback through the secondary DAI.
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Acked-by: Krzysztof Kozlowski <krzk@kernel.org>
Signed-off-by: Mark Brown <broonie@kernel.org>
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It is reported that there's a constant background "hum/whitenoise"
in the headset on the Lenovo X1 machines with the codec alc285, and it
is confirmed that if we run the command below, the noise will stop.
sudo hda-verb /dev/snd/hwC0D0 0x1d SET_PIN_WIDGET_CONTROL 0x0
Then I consulted this issue with Kailang, he told me the pin 0x1d on
this codec is used for PC beep in, the noise probably comes from this
pin and we can also disable the PC beep in passthrough, then the PC
beep in will not affect other sound playback.
Fixes: c4cfcf6f4297 ("ALSA: hda/realtek - fix the pop noise on headphone for lenovo laptops")
Bugzilla: https://bugzilla.redhat.com/show_bug.cgi?id=1660581
Cc: <stable@vger.kernel.org>
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Hui Wang <hui.wang@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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System76 oryp5
On the System76 Oryx Pro (oryp5), there is a headset microphone input
attached to 0x19 that does not have a jack detect. In order to get it
working, the pin configuration needs to be set correctly, and the
ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC fixup needs to be applied. This is
similar to the MIC_NO_PRESENCE fixups for some Dell laptops, except we
have a separate microphone jack that is already configured correctly.
Since the ALC1220 does not have a fixup similar to
ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, I have exposed the fixup from the
ALC269 in a way that it can be accessed from the
alc1220_fixup_system76_oryp5 function. In addition, the
alc1220_fixup_clevo_p950 needs to be applied to gain speaker output.
Signed-off-by: Jeremy Soller <jeremy@system76.com>
Cc: <stable@vger.kernel.org>
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:
"It's a bit of surprising that we've got more changes than hoped at
this late stage, but they all don't look too scary but small fixes.
One change in ALSA core side is again the PCM regression fix that was
partially addressed for OSS, but now the all relevant change is
reverted instead. Also, a few ASoC core fixes for UAF and OOB are
included, while the rest are usual random device-specific fixes"
* tag 'sound-5.0-rc7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: pcm: Revert capture stream behavior change in blocking mode
ALSA: usb-audio: Fix implicit fb endpoint setup by quirk
ALSA: hda - Add quirk for HP EliteBook 840 G5
ASoC: samsung: Prevent clk_get_rate() calls in atomic context
ASoC: rsnd: ssiu: correct shift bit for ssiu9
ASoC: rsnd: fixup rsnd_ssi_master_clk_start() user count check
ASoC: dapm: fix out-of-bounds accesses to DAPM lookup tables
ASoC: topology: fix oops/use-after-free case with dai driver
ASoC: rsnd: fixup MIX kctrl registration
ASoC: core: Allow soc_find_component lookups to match parent of_node
ASoC: rt5682: Correct the setting while select ASRC clk for AD/DA filter
ASoC: MAINTAINERS: fsl: Change Fabio's email address
ASoC: hdmi-codec: fix oops on re-probe
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In the commit 62ba568f7aef ("ALSA: pcm: Return 0 when size <
start_threshold in capture"), we changed the behavior of
__snd_pcm_lib_xfer() to return immediately with 0 when a capture
stream has a high start_threshold. This was intended to be a
correction of the behavior consistency and looked harmless, but this
was the culprit of the recent breakage reported by syzkaller, which
was fixed by the commit e190161f96b8 ("ALSA: pcm: Fix tight loop of
OSS capture stream").
At the time for the OSS fix, I didn't touch the behavior for ALSA
native API, as assuming that this behavior actually is good. But this
turned out to be also broken actually for a similar deployment,
e.g. one thread goes to a write loop in blocking mode while another
thread controls the start/stop of the stream manually.
Overall, the original commit is harmful, and it brings less merit to
keep that behavior. Let's revert it.
Fixes: 62ba568f7aef ("ALSA: pcm: Return 0 when size < start_threshold in capture")
Fixes: e190161f96b8 ("ALSA: pcm: Fix tight loop of OSS capture stream")
Cc: <stable@vger.kernel.org>
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 v5.0
A selection of driver specific fixes here, along with a few core fixes:
- A fixup for some MFD devices that were broken by the previous fixes
for deferred probe.
- A fix for potential out of bounds array accesses when ordering DAPM
power/up down sequences.
- Avoid use after free issue when unloading and reloading drivers using
topologies.
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This patch moves clk_get_rate() call from trigger() to hw_params()
callback to avoid calling sleeping clk API from atomic context
and prevent deadlock as indicated below.
Before this change clk_get_rate() was being called with same
spinlock held as the one passed to the clk API when registering
clocks exposed by the I2S driver.
[ 82.109780] BUG: sleeping function called from invalid context at kernel/locking/mutex.c:908
[ 82.117009] in_atomic(): 1, irqs_disabled(): 128, pid: 1554, name: speaker-test
[ 82.124235] 3 locks held by speaker-test/1554:
[ 82.128653] #0: cc8c5328 (snd_pcm_link_rwlock){...-}, at: snd_pcm_stream_lock_irq+0x20/0x38
[ 82.137058] #1: ec9eda17 (&(&substream->self_group.lock)->rlock){..-.}, at: snd_pcm_ioctl+0x900/0x1268
[ 82.146417] #2: 6ac279bf (&(&pri_dai->spinlock)->rlock){..-.}, at: i2s_trigger+0x64/0x6d4
[ 82.154650] irq event stamp: 8144
[ 82.157949] hardirqs last enabled at (8143): [<c0a0f574>] _raw_read_unlock_irq+0x24/0x5c
[ 82.166089] hardirqs last disabled at (8144): [<c0a0f6a8>] _raw_read_lock_irq+0x18/0x58
[ 82.174063] softirqs last enabled at (8004): [<c01024e4>] __do_softirq+0x3a4/0x66c
[ 82.181688] softirqs last disabled at (7997): [<c012d730>] irq_exit+0x140/0x168
[ 82.188964] Preemption disabled at:
[ 82.188967] [<00000000>] (null)
[ 82.195728] CPU: 6 PID: 1554 Comm: speaker-test Not tainted 5.0.0-rc5-00192-ga6e6caca8f03 #191
[ 82.204302] Hardware name: SAMSUNG EXYNOS (Flattened Device Tree)
[ 82.210376] [<c0111a54>] (unwind_backtrace) from [<c010d8f4>] (show_stack+0x10/0x14)
[ 82.218084] [<c010d8f4>] (show_stack) from [<c09ef004>] (dump_stack+0x90/0xc8)
[ 82.225278] [<c09ef004>] (dump_stack) from [<c0152980>] (___might_sleep+0x22c/0x2c8)
[ 82.232990] [<c0152980>] (___might_sleep) from [<c0a0a2e4>] (__mutex_lock+0x28/0xa3c)
[ 82.240788] [<c0a0a2e4>] (__mutex_lock) from [<c0a0ad80>] (mutex_lock_nested+0x1c/0x24)
[ 82.248763] [<c0a0ad80>] (mutex_lock_nested) from [<c04923dc>] (clk_prepare_lock+0x78/0xec)
[ 82.257079] [<c04923dc>] (clk_prepare_lock) from [<c049538c>] (clk_core_get_rate+0xc/0x5c)
[ 82.265309] [<c049538c>] (clk_core_get_rate) from [<c0766b18>] (i2s_trigger+0x490/0x6d4)
[ 82.273369] [<c0766b18>] (i2s_trigger) from [<c074fec4>] (soc_pcm_trigger+0x100/0x140)
[ 82.281254] [<c074fec4>] (soc_pcm_trigger) from [<c07378a0>] (snd_pcm_do_start+0x2c/0x30)
[ 82.289400] [<c07378a0>] (snd_pcm_do_start) from [<c07376cc>] (snd_pcm_action_single+0x38/0x78)
[ 82.298065] [<c07376cc>] (snd_pcm_action_single) from [<c073a450>] (snd_pcm_ioctl+0x910/0x1268)
[ 82.306734] [<c073a450>] (snd_pcm_ioctl) from [<c0292344>] (do_vfs_ioctl+0x90/0x9ec)
[ 82.314443] [<c0292344>] (do_vfs_ioctl) from [<c0292cd4>] (ksys_ioctl+0x34/0x60)
[ 82.321808] [<c0292cd4>] (ksys_ioctl) from [<c0101000>] (ret_fast_syscall+0x0/0x28)
[ 82.329431] Exception stack(0xeb875fa8 to 0xeb875ff0)
[ 82.334459] 5fa0: 00033c18 b6e31000 00000004 00004142 00033d80 00033d80
[ 82.342605] 5fc0: 00033c18 b6e31000 00008000 00000036 00008000 00000000 beea38a8 00008000
[ 82.350748] 5fe0: b6e3142c beea384c b6da9a30 b6c9212c
[ 82.355789]
[ 82.357245] ======================================================
[ 82.363397] WARNING: possible circular locking dependency detected
[ 82.369551] 5.0.0-rc5-00192-ga6e6caca8f03 #191 Tainted: G W
[ 82.376395] ------------------------------------------------------
[ 82.382548] speaker-test/1554 is trying to acquire lock:
[ 82.387834] 6d2007f4 (prepare_lock){+.+.}, at: clk_prepare_lock+0x78/0xec
[ 82.394593]
[ 82.394593] but task is already holding lock:
[ 82.400398] 6ac279bf (&(&pri_dai->spinlock)->rlock){..-.}, at: i2s_trigger+0x64/0x6d4
[ 82.408197]
[ 82.408197] which lock already depends on the new lock.
[ 82.416343]
[ 82.416343] the existing dependency chain (in reverse order) is:
[ 82.423795]
[ 82.423795] -> #1 (&(&pri_dai->spinlock)->rlock){..-.}:
[ 82.430472] clk_mux_set_parent+0x34/0xb8
[ 82.434975] clk_core_set_parent_nolock+0x1c4/0x52c
[ 82.440347] clk_set_parent+0x38/0x6c
[ 82.444509] of_clk_set_defaults+0xc8/0x308
[ 82.449186] of_clk_add_provider+0x84/0xd0
[ 82.453779] samsung_i2s_probe+0x408/0x5f8
[ 82.458376] platform_drv_probe+0x48/0x98
[ 82.462879] really_probe+0x224/0x3f4
[ 82.467037] driver_probe_device+0x70/0x1c4
[ 82.471716] bus_for_each_drv+0x44/0x8c
[ 82.476049] __device_attach+0xa0/0x138
[ 82.480382] bus_probe_device+0x88/0x90
[ 82.484715] deferred_probe_work_func+0x6c/0xbc
[ 82.489741] process_one_work+0x200/0x740
[ 82.494246] worker_thread+0x2c/0x4c8
[ 82.498408] kthread+0x128/0x164
[ 82.502131] ret_from_fork+0x14/0x20
[ 82.506204] (null)
[ 82.508976]
[ 82.508976] -> #0 (prepare_lock){+.+.}:
[ 82.514264] __mutex_lock+0x60/0xa3c
[ 82.518336] mutex_lock_nested+0x1c/0x24
[ 82.522756] clk_prepare_lock+0x78/0xec
[ 82.527088] clk_core_get_rate+0xc/0x5c
[ 82.531421] i2s_trigger+0x490/0x6d4
[ 82.535494] soc_pcm_trigger+0x100/0x140
[ 82.539913] snd_pcm_do_start+0x2c/0x30
[ 82.544246] snd_pcm_action_single+0x38/0x78
[ 82.549012] snd_pcm_ioctl+0x910/0x1268
[ 82.553345] do_vfs_ioctl+0x90/0x9ec
[ 82.557417] ksys_ioctl+0x34/0x60
[ 82.561229] ret_fast_syscall+0x0/0x28
[ 82.565477] 0xbeea384c
[ 82.568421]
[ 82.568421] other info that might help us debug this:
[ 82.568421]
[ 82.576394] Possible unsafe locking scenario:
[ 82.576394]
[ 82.582285] CPU0 CPU1
[ 82.586792] ---- ----
[ 82.591297] lock(&(&pri_dai->spinlock)->rlock);
[ 82.595977] lock(prepare_lock);
[ 82.601782] lock(&(&pri_dai->spinlock)->rlock);
[ 82.608975] lock(prepare_lock);
[ 82.612268]
[ 82.612268] *** DEADLOCK ***
Fixes: 647d04f8e07a ("ASoC: samsung: i2s: Ensure the RCLK rate is properly determined")
Reported-by: Krzysztof Kozłowski <krzk@kernel.org>
Signed-off-by: Sylwester Nawrocki <s.nawrocki@samsung.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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Currently "0xf << 36" is used to
clear SSIU-9 internal buffer state, which overflows 32-bit value
according to user reference manual, it is always bit4 ~ bit7
of SSI_SYS_STATUS[1,3,5,7] registers indicate
SSIU-9's buffer state, so "0xf << 4" should be used.
This patch fix incorrect shifting issue in SSIU-9 case
Fixes: commit b7169ddea2f2 ("ASoC: rsnd: remove RSND_REG_ from rsnd_reg")
Signed-off-by: Jiada Wang <jiada_wang@mentor.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set
under non-atomic") added new rsnd_ssi_prepare() and moved
rsnd_ssi_master_clk_start() to .prepare.
But, ssi user count (= ssi->usrcnt) is incremented at .init
(= rsnd_ssi_init()).
Because of these timing exchange, ssi->usrcnt check at
rsnd_ssi_master_clk_start() should be adjusted.
Otherwise, 2nd master clock setup will be no check.
This patch fixup this issue.
Fixes: commit 4d230d1271064 ("ASoC: rsnd: fixup not to call clk_get/set under non-atomic")
Reported-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Reported-by: Valentine Barshak <valentine.barshak@cogentembedded.com>
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Tested-by: Yusuke Goda <yusuke.goda.sx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
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