Connor McAdams [Mon, 3 Aug 2020 00:29:25 +0000 (20:29 -0400)]
ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
When the ZxR headphone gain control was added, the ca0132_switch_get
function was not updated, which meant that the changes to the control
state were not saved when entering/exiting alsamixer.
Huacai Chen [Sun, 2 Aug 2020 09:26:40 +0000 (17:26 +0800)]
ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
There are several Loongson-3 based laptops produced by CZC or Lemote,
they use alc269/alc662 codecs and need specific pin-tables, this patch
add their pin-tables.
Kailang Yang [Wed, 29 Jul 2020 07:09:27 +0000 (15:09 +0800)]
ALSA: hda/realtek - Fixed HP right speaker no sound
HP NB right speaker had no sound output.
This platform was connected to I2S Amp for speaker out.(None Realtek I2S Amp IC)
EC need to check codec GPIO1 pin to initial I2S Amp.
ALSA: hda: fix NULL pointer dereference during suspend
When the ASoC card registration fails and the codec component driver
never probes, the codec device is not initialized and therefore
memory for codec->wcaps is not allocated. This results in a NULL pointer
dereference when the codec driver suspend callback is invoked during
system suspend. Fix this by returning without performing any actions
during codec suspend/resume if the card was not registered successfully.
ALSA: hda/hdmi: Fix keep_power assignment for non-component devices
It's been reported that, when neither nouveau nor Nvidia graphics
driver is used, the screen starts flickering. And, after comparing
between the working case (stable 4.4.x) and the broken case, it turned
out that the problem comes from the audio component binding. The
Nvidia and AMD audio binding code clears the bus->keep_power flag
whenever snd_hdac_acomp_init() succeeds. But this doesn't mean that
the component is actually bound, but it merely indicates that it's
ready for binding. So, when both nouveau and Nvidia are blacklisted
or not ready, the driver keeps running without the audio component but
also with bus->keep_power = false. This made the driver runtime PM
kicked in and powering down when unused, which results in flickering
in the graphics side, as it seems.
For fixing the bug, this patch moves the bus->keep_power flag change
into generic_acomp_notifier_set() that is the function called from the
master_bind callback of component ops; i.e. it's guaranteed that the
binding succeeded.
ALSA: hda: Workaround for spurious wakeups on some Intel platforms
We've received a regression report on Intel HD-audio controller that
wakes up immediately after S3 suspend. The bisection leads to the
commit ac791d7801a4 ("ALSA: hda: Skip controller resume if not
needed"). This commit replaces the system-suspend to use
pm_runtime_force_suspend() instead of the direct call of
__azx_runtime_suspend(). However, by some really mysterious reason,
pm_runtime_force_suspend() causes a spurious wakeup (although it calls
the same __azx_runtime_suspend() internally).
As an ugly workaround for now, revert the behavior to call
__azx_runtime_suspend() and __azx_runtime_resume() for those old Intel
platforms that may exhibit such a problem, while keeping the new
standard pm_runtime_force_suspend() and pm_runtime_force_resume()
pair for the remaining chips.
Armas Spann [Fri, 24 Jul 2020 14:08:37 +0000 (16:08 +0200)]
ALSA: hda/realtek: typo_fix: enable headset mic of ASUS ROG Zephyrus G14(GA401) series with ALC289
This patch fixes a small typo I accidently submitted with the initial patch. The board should be named GA401 not G401.
Fixes: 1584199bbbda ("ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289") Signed-off-by: Armas Spann <zappel@retarded.farm> Cc: <stable@vger.kernel.org> Link: https://lore.kernel.org/r/20200724140837.302763-1-zappel@retarded.farm Signed-off-by: Takashi Iwai <tiwai@suse.de>
Armas Spann [Fri, 24 Jul 2020 14:06:16 +0000 (16:06 +0200)]
ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G15(GA502) series with ALC289
This patch adds support for headset mic to the ASUS ROG Zephyrus
G15(GA502) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.
Follow the recent inclusive terminology guidelines and replace the
word "slave" in vmaster API. I chose the word "follower" at this time
since it seems fitting for the purpose.
Note that the word "master" is kept in API, since it refers rather to
audio master volume control.
Also, while we're at it, a typo in comments is corrected, too.
Rationale:
Reduces attack surface on kernel devs opening the links for MITM
as HTTPS traffic is much harder to manipulate.
Deterministic algorithm:
For each file:
If not .svg:
For each line:
If doesn't contain `\bxmlns\b`:
For each link, `\bhttp://[^# \t\r\n]*(?:\w|/)`:
If neither `\bgnu\.org/license`, nor `\bmozilla\.org/MPL\b`:
If both the HTTP and HTTPS versions
return 200 OK and serve the same content:
Replace HTTP with HTTPS.
ALSA: hda/realtek: Fixed ALC298 sound bug by adding quirk for Samsung Notebook Pen S
Fixed no headphone sound bug on laptop Samsung Notebook Pen S
(950SBE-951SBE), by using existing patch in Linus' tree, commit 4232ed22997a (ALSA: hda/realtek: Add quirk for Samsung Notebook).
This laptop uses the same ALC298 but different subsystem id 0x144dc812.
I added SND_PCI_QUIRK at sound/pci/hda/patch_realtek.c
Merge tag 'asoc-fix-v5.8-rc5' of https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Fixes for v5.8
An awful lot of mostly small fixes here, mainly for x86 based platforms
and the CODEC drivers mainly used on them. For the most part this is
either minor device specific stuff which seems to come from detailed
testing or robustness against errors which comes from people having done
some fuzzing runs aginst the topology code.
ALSA: info: Drop WARN_ON() from buffer NULL sanity check
snd_info_get_line() has a sanity check of NULL buffer -- both buffer
itself being NULL and buffer->buffer being NULL. Basically both
checks are valid and necessary, but the problem is that it's with
snd_BUG_ON() macro that triggers WARN_ON(). The latter condition
(NULL buffer->buffer) can be met arbitrarily by user since the buffer
is allocated at the first write, so it means that user can trigger
WARN_ON() at will.
This patch addresses it by simply moving buffer->buffer NULL check out
of snd_BUG_ON() so that spurious WARNING is no longer triggered.
Revert "ALSA: pcm: Use SG-buffer only when direct DMA is available"
This reverts commit 46920b9f6ad8 ("ALSA: pcm: Use SG-buffer only when
direct DMA is available") also the modification commit 05239164fb11
("ALSA: pcm: Fix build error on m68k and others").
Poking the DMA internal helper is a layer violation, so we should
avoid that. Meanwhile the actual bug has been addressed by the
Kconfig fix in commit 16ccc9808396 ("dma-pool: decouple DMA_REMAP from
DMA_COHERENT_POOL"), so we can live without this hack.
snd_byt_cht_es8316_mc_probe() misses to call put_device() in an error
path. Add the missed function call to fix it.
Fixes: f8d5e4ca1533 ("ASoC: Intel: bytcht_es8316: Add quirk for inverted jack detect") Signed-off-by: Jing Xiangfeng <jingxiangfeng@huawei.com> Reviewed-by: Hans de Goede <hdegoede@redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Link: https://lore.kernel.org/r/20200714080918.148196-1-jingxiangfeng@huawei.com Signed-off-by: Mark Brown <broonie@kernel.org>
Jon Hunter [Tue, 14 Jul 2020 16:08:41 +0000 (17:08 +0100)]
ALSA: hda/tegra: Disable sync-write operation
Commit 655d880d4442 ("ALSA: hda: Enable sync-write operation as default
for all controllers") enabled sync-write for all controllers and this is
causing audio playback on the Tegra186 HDA device to fail. For now,
disable sync-write support for Tegra to fix this.
Jian-Hong Pan [Mon, 13 Jul 2020 06:04:22 +0000 (14:04 +0800)]
ALSA: hda/realtek: Enable headset mic of Acer TravelMate B311R-31 with ALC256
The Acer TravelMate B311R-31 laptop's audio (1025:1430) with ALC256
cannot detect the headset microphone until
ALC256_FIXUP_ACER_MIC_NO_PRESENCE quirk maps the NID 0x19 as the headset
mic pin.
Armas Spann [Sat, 11 Jul 2020 11:05:57 +0000 (13:05 +0200)]
ALSA: hda/realtek: enable headset mic of ASUS ROG Zephyrus G14(G401) series with ALC289
This patch adds support for headset mic to the ASUS ROG Zephyrus
G14(GA401) notebook series by adding the corresponding
vendor/pci_device id, as well as adding a new fixup for the used
realtek ALC289. The fixup stets the correct pin to get the headset mic
correctly recognized on audio-jack.
ALSA: usb-audio: Fix race against the error recovery URB submission
USB MIDI driver has an error recovery mechanism to resubmit the URB in
the delayed timer handler, and this may race with the standard start /
stop operations. Although both start and stop operations themselves
don't race with each other due to the umidi->mutex protection, but
this isn't applied to the timer handler.
For fixing this potential race, the following changes are applied:
- Since the timer handler can't use the mutex, we apply the
umidi->disc_lock protection at each input stream URB submission;
this also needs to change the GFP flag to GFP_ATOMIC
- Add a check of the URB refcount and skip if already submitted
- Move the timer cancel call at disconnection to the beginning of the
procedure; this assures the in-flight timer handler is gone properly
before killing all pending URBs
ALSA: line6: Sync the pending work cancel at disconnection
Recently syzkaller reported a UAF in LINE6 driver, and it's likely
because we call cancel_delayed_work() at the disconnect callback
instead of cancel_delayed_work_sync(). Let's use the correct one
instead.
ALSA: line6: Perform sanity check for each URB creation
LINE6 drivers create stream URBs with a fixed pipe without checking
its validity, and this may lead to a kernel WARNING at the submission
when a malformed USB descriptor is passed.
For avoiding the kernel warning, perform the similar sanity checks for
each pipe type at creating a URB.
ALSA: xen: Remove superfluous fall through comments
The "fall through" comments found in switch-cases in ALSA xen driver
are all superfluous. The kernel coding style allows the multiple
cases in a row. Let's remove them.
Replace the existing /* fall through */ comments and its variants with
the new pseudo-keyword macro fallthrough[1]. Also, remove unnecessary
fall-through markings when it is the case.
Mark Hills [Wed, 8 Jul 2020 10:18:48 +0000 (11:18 +0100)]
ALSA: echoaudio: Address bugs in the interrupt handling
Distorted audio appears occasionally, affecting either playback or
capture and requiring the affected substream to be closed by all
applications and re-opened.
The best way I have found to reproduce the bug is to use dmix in
combination with Chromium, which opens the audio device multiple times
in threads. Anecdotally, the problems appear to have increased with
faster CPUs. I ruled out 32-bit counter wrapping; it often happens
much earlier.
Since applying this patch I have not had problems, where previously
they would occur several times a day.
The patch targets the following issues:
* Check for progress using the counter from the hardware, not after it
has been truncated to the buffer.
This is a clean way to address a possible bug where if a whole
ringbuffer advances between interrupts, it goes unnoticed.
* Move last_period state from chip to pipe
This more logically belongs as part of pipe, and code is reasier to
read if it is "counter position last time a period elapsed".
Now the code has no references to period count. A period is just
when the regular counter crosses a threshold. This increases
readability and reduces scope for bugs.
* Treat period notification and buffer advance independently:
This helps to clarify what is the responsibility of the interrupt
handler, and what is pcm_pointer().
Removing shared state between these operations means race conditions
are fixed without introducing locks. Synchronisation is only around
the read of pipe->dma_counter. There may be cache line contention
around "struct audiopipe" but I did not have cause to profile this.
Pay attention to be robust where dma_counter wrapping is not a
multiple of period_size or buffer_size.
This is a revised patch based on feedback from Takashi and Giuliano.
Mark Hills [Wed, 8 Jul 2020 10:18:45 +0000 (11:18 +0100)]
ALSA: echoaudio: Race conditions around "opencount"
Use of atomics does not make these statements robust:
atomic_inc(&chip->opencount);
if (atomic_read(&chip->opencount) > 1 && chip->rate_set)
chip->can_set_rate=0;
and
if (atomic_read(&chip->opencount)) {
if (chip->opencount) {
changed = -EAGAIN;
} else {
changed = set_digital_mode(chip, dmode);
It would be necessary to atomically increment or decrement the value
and use the returned result. And yet we still need to prevent other
threads making use of "can_set_rate" while we set it.
However in all but one case the atomic is misleading as they are already
running with "mode_mutex" held.
Decisions are made on mode setting are often intrinsically connected
to "opencount" because some operations are not permitted unless
there is sole ownership.
So instead simplify this, and use "mode_mutex" as a lock for all reference
counting and mode setting.
Mark Hills [Wed, 8 Jul 2020 10:18:44 +0000 (11:18 +0100)]
ALSA: echoaudio: Remove redundant check
This check is always false, as it's not the responsibilty of the
device-specific code to make this check. It is already checked
in snd_echo_digital_mode_put.
I do not have a Mona interface to test this change.
This patch is in preparation for follow-up patch to modify the
behavior of "opencount".
Mark Brown [Wed, 8 Jul 2020 14:02:04 +0000 (15:02 +0100)]
Merge series "ASoC: topology: fix error handling flow" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
While experimenting and introducing errors in Baytrail topology files
until I got them right, I encountered multiple kernel oopses and
memory leaks. This is a first batch to harden the code, but we should
probably think of a tool to fuzz the topology...
Pierre-Louis Bossart (5):
ASoC: topology: fix kernel oops on route addition error
ASoC: topology: fix tlvs in error handling for widget_dmixer
ASoC: topology: use break on errors, not continue
ASoC: topology: factor kfree(se) in error handling
ASoC: topology: add more logs when topology load fails.
Mark Brown [Wed, 8 Jul 2020 14:02:03 +0000 (15:02 +0100)]
Merge series "ASoC: codecs: add MAX98373 Soundwire driver" from Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>:
V2 with a number of cleanups:
split between I2C and SoundWire modes, as done for rt5682, and updated
Kconfigs.
removed useless initializations common to both modes
removed idle_bias on
fixed register classified as volatile in error
fixed SPDX comments
Pierre-Louis Bossart (2):
ASoC: codecs: max98373: split I2C and common parts
ASoC: Intel: sof-sdw: add MAX98373 I2C dependencies
Ryan Lee (2):
ASoC: codecs: max98373: Removed superfluous volume control from chip
default
ASoC: codecs: max98373: add SoundWire support
randerwang (2):
ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend
ASoC: Intel: sdw_max98373: add card_late_probe support
ASoC: SOF: core: fix null-ptr-deref bug during device removal
The DSP should be notified for device removal only if the
probe was successful. Fixes the following KASAN bug:
BUG: KASAN: null-ptr-deref in sof_ipc_tx_message+0x80/0x160 [snd_sof]
ASoc: codecs: max98373: remove Idle_bias_on to let codec suspend
Idle_bias_on is used to decide bias on/off in standby state by dapm.
When Idle_bias_on is set to one, dapm will keep max98373 active at
idle time. Max98373 is doing nothing in this state, so remove
idle_bias_on setting to let max98373 get suspended when it is idle.
Signed-off-by: randerwang <rander.wang@linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ryan Lee <ryans.lee@maximintegrated.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200707205740.114927-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Ryan Lee [Tue, 7 Jul 2020 20:57:36 +0000 (15:57 -0500)]
ASoC: codecs: max98373: Removed superfluous volume control from chip default
Volume control in probe function is not necessary.
Signed-off-by: Ryan Lee <ryans.lee@maximintegrated.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Rander Wang <rander.wang@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200707205740.114927-3-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ASoC: topology: fix kernel oops on route addition error
When errors happens while loading graph components, the kernel oopses
while trying to remove all topology components. This can be
root-caused to a list pointing to memory that was already freed on
error.
remove_route() is already called on errors and will perform the
required cleanups so there's no need to free the route memory in
soc_tplg_dapm_graph_elems_load() if the route was added to the
list. We do however want to free the routes allocated but not added to
the list.
Fixes: 3991d4f2a4965 ('ASoC: topology: modify dapm route loading routine and add dapm route unloading') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Link: https://lore.kernel.org/r/20200707203749.113883-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
Daniel Baluta [Tue, 7 Jul 2020 21:04:39 +0000 (16:04 -0500)]
ASoC: SOF: imx: add min/max channels for SAI/ESAI on i.MX8/i.MX8M
This is identical with change for Intel platforms done with
commit 459cf4085260 ("ASoC: SOF: Intel: add min/max channels for SSP on Baytrail/Broadwell")
and fixes a regression on i.MX8/i.MX8M:
[ 25.705750] esai-Codec: ASoC: no backend playback stream
[ 27.923378] esai-Codec: ASoC: no users playback at close - state
This is root-caused to the introduction of the DAI capability checks
with snd_soc_dai_stream_valid(). Its use in soc-pcm.c makes it a
requirement for all DAIs to report at least a non-zero min_channels
field.
Fixes: f27f69de3f74ae ("ASoC: soc-pcm: dpcm: Only allow playback/capture if supported") Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com> Link: https://lore.kernel.org/r/20200707210439.115300-4-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
When SOF is used, the normal links are converted into DPCM ones. This
generates an error
[ 58.276668] bdw-rt5677 bdw-rt5677: CPU DAI spi-RT5677AA:00 for rtd
Wake on Voice does not support playback
[ 58.276676] bdw-rt5677 bdw-rt5677: ASoC: can't create pcm Wake on
Voice :-22
ASoC: soc-dai: set dai_link dpcm_ flags with a helper
Add a helper to walk through all the DAIs and set dpcm_playback and
dpcm_capture flags based on the DAIs capabilities, and use this helper
to avoid setting these flags arbitrarily in generic cards.
The commit referenced in the Fixes tag did not introduce the
configuration issue but will prevent the card from probing when
detecting invalid configurations.
Fixes: 96094a591312c ('ASoC: soc-pcm: dpcm: fix playback/capture checks') Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com> Reviewed-by: Kai Vehmanen <kai.vehmanen@linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com> Link: https://lore.kernel.org/r/20200707210439.115300-2-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown <broonie@kernel.org>
ALSA: usb/line6: remove 'defined but not used' warning
Fix W=1 warning. Variables are declared in a header file included from
multiple C files, replace by #defines as suggested by Takashi
sound/usb/line6/driver.h:70:18: warning: ‘SYSEX_EXTRA_SIZE’ defined
but not used [-Wunused-const-variable=]
70 | static const int SYSEX_EXTRA_SIZE = sizeof(line6_midi_id) + 4;
| ^~~~~~~~~~~~~~~~
sound/usb/line6/driver.h:69:18: warning: ‘SYSEX_DATA_OFS’ defined but
not used [-Wunused-const-variable=]
69 | static const int SYSEX_DATA_OFS = sizeof(line6_midi_id) + 3;
| ^~~~~~~~~~~~~~
The commit 46920b9f6ad8 ("ALSA: pcm: Use SG-buffer only when direct
DMA is available") introduced a check of the DMA type and this caused
a build error on m68k (and possibly some others) due to the lack of
dma_is_direct() definition. Since the check is needed only for
CONFIG_SND_DMA_SGBUF enablement (i.e. solely x86), use #ifdef instead
of IS_ENABLED() for avoiding such a build error.
Maxime Ripard [Tue, 7 Jul 2020 07:42:37 +0000 (09:42 +0200)]
ASoC: core: Remove only the registered component in devm functions
The ASoC devm_ functions that register a component
(devm_snd_soc_register_component and devm_snd_dmaengine_pcm_register) will
clean their component by running snd_soc_unregister_component.
snd_soc_unregister_component will then remove all the components for the
device that was used to register the component in the first place.
However, some drivers register several components (such as a DAI and a
dmaengine PCM) on the same device, and if the dmaengine PCM is registered
first, then the DAI will be cleaned up first and
snd_dmaengine_pcm_unregister will be called next.
snd_dmaengine_pcm_unregister will then lookup the dmaengine PCM component
on the device, and if there's one unregister that component and release its
dmaengine channels. That doesn't happen in practice though since the first
call to snd_soc_unregister_component removed all the components, so we
never get the chance to release the dmaengine channels.
In order to fix this, instead of removing all the components for a given
device, we can simply remove the component that was registered in the first
place. We should have the same number of component registration than we
have components, so it should work just fine.
External HDMI receivers have analog circuitry that needs to be powered-on
when exiting standby, and a mechanism to detect PCM v. IEC61937 data.
These two steps take time and up to 2-3 seconds of audio may be muted
when starting playback.
Intel hardware (Haswell and beyond) can keep the link active
with a 'silent stream', so that the receiver does not go through those
two steps when valid audio is transmitted. This mechanism relies
on an setting the channel_id as 0xf, sending info packet and preventing
the codec from going to D3, which will increase the platform
static power consumption. The info packet assumes a basic 2ch stereo,
and the silent stream is enabled when connecting a monitor.
In case of format changes the detection of PCM v. IEC61937 needs to
be re-run. In this case there is no way to avoid the 2-3s mute.
The silent stream is enabled with a Kconfig option, as well as a kernel
parameter should there be a need to override the build time default.
This approach is used based on the power_save capability as an example,
but in the future, it may be used with a kcontrol,
depending on UCM support for HDaudio legacy.
sound/drivers/vx/vx_core.c: In function ‘snd_vx_threaded_irq_handler’:
sound/drivers/vx/vx_core.c:515:3: warning: suggest braces around empty
body in an ‘if’ statement [-Wempty-body]
515 | ; /* so far, nothing to do yet */
| ^
ALSA: pci/via82xx: remove 'set but not used' warnings
Fix W=1 warnings. Mark variables as __always_unused.
sound/pci/via82xx.c: In function ‘snd_via82xx_codec_wait’:
sound/pci/via82xx.c:547:6: warning: variable ‘err’ set but not used
[-Wunused-but-set-variable]
547 | int err;
| ^~~
sound/pci/via82xx_modem.c: In function ‘snd_via82xx_codec_wait’:
sound/pci/via82xx_modem.c:401:6: warning: variable ‘err’ set but not
used [-Wunused-but-set-variable]
401 | int err;
| ^~~
ALSA: pci/es1938: remove 'set but not used' warning
Fix W=1 warning. Mark the 'audiostatus' variable as __always_unused.
sound/pci/es1938.c: In function ‘snd_es1938_interrupt’:
sound/pci/es1938.c:1622:24: warning: variable ‘audiostatus’ set but
not used [-Wunused-but-set-variable]
1622 | unsigned char status, audiostatus;
| ^~~~~~~~~~~
Fix W=1 warning. Mark the 'req' variable as __always_unused.
sound/xen/xen_snd_front.c: In function ‘xen_snd_front_stream_close’:
sound/xen/xen_snd_front.c:117:21: warning: variable ‘req’ set but not
used [-Wunused-but-set-variable]
117 | struct xensnd_req *req;
| ^~~
sound/pci/rme9652/hdspm.c: In function ‘hdspm_autosync_ref’:
sound/pci/rme9652/hdspm.c:3030:16: warning: comparison of unsigned
expression >= 0 is always true [-Wtype-limits]
3030 | if ((syncref >= HDSPM_AES32_AUTOSYNC_FROM_WORD) &&
| ^~
ALSA: pci/korg1212: remove 'set but not used' warnings
Fix W=1 warnings. Two variables are only used for debug logs, mark
with __maybe_unused:
sound/pci/korg1212/korg1212.c: In function ‘snd_korg1212_create’:
sound/pci/korg1212/korg1212.c:2152:36: warning: variable ‘iomem2_size’
set but not used [-Wunused-but-set-variable]
2152 | unsigned ioport_size, iomem_size, iomem2_size;
| ^~~~~~~~~~~
sound/pci/korg1212/korg1212.c:2152:11: warning: variable ‘ioport_size’
set but not used [-Wunused-but-set-variable]
2152 | unsigned ioport_size, iomem_size, iomem2_size;
| ^~~~~~~~~~~
ALSA: pci/emu10k1: remove 'set but not used' warning
Fix W=1 warning. The loopsize variable is only used in compiled-out
code, so mark with __maybe_unused.
sound/pci/emu10k1/emu10k1_patch.c: In function
‘snd_emu10k1_sample_new’:
sound/pci/emu10k1/emu10k1_patch.c:30:22: warning: variable ‘loopsize’
set but not used [-Wunused-but-set-variable]
30 | int truesize, size, loopsize, blocksize;
| ^~~~~~~~
ASoC: pci/emu10k1: remove "set but not used' warnings
Fix W=1 warnings. Mark variables used for reads as __always_unused.
sound/pci/emu10k1/emu10k1_main.c: In function
‘snd_emu10k1_cardbus_init’:
sound/pci/emu10k1/emu10k1_main.c:626:15: warning: variable ‘value’ set
but not used [-Wunused-but-set-variable]
626 | unsigned int value;
| ^~~~~
sound/pci/emu10k1/emu10k1_main.c: In function
‘snd_emu1010_load_firmware_entry’:
sound/pci/emu10k1/emu10k1_main.c:656:15: warning: variable
‘write_post’ set but not used [-Wunused-but-set-variable]
656 | unsigned int write_post;
| ^~~~~~~~~~
ALSA: pci/aw2-saa7146: remove 'set but not used' warning
Fix W=1 warning by nothing variable as always unused.
sound/pci/aw2/aw2-saa7146.c: In function ‘snd_aw2_saa7146_interrupt’:
sound/pci/aw2/aw2-saa7146.c:333:15: warning: variable ‘iicsta’ set but
not used [-Wunused-but-set-variable]
333 | unsigned int iicsta;
| ^~~~~~
ALSA: pci/echoaudio: remove 'set but not used' warning
Fix W=1 warning. One variable is only used in a conditionally-compiled
block, mark as __maybe_unused
sound/pci/echoaudio/echoaudio.c: In function ‘snd_echo_probe’:
sound/pci/echoaudio/echoaudio.c:1958:6: warning: variable ‘i’ set but
not used [-Wunused-but-set-variable]
1958 | int i, err;
| ^
ALSA: pci/asihpi: remove 'set but not used' warning
Fix W=1 warnings, mark variables as __always_unused
sound/pci/asihpi/asihpi.c: In function ‘snd_asihpi_tuner_band_get’:
sound/pci/asihpi/asihpi.c:1907:6: warning: variable ‘num_bands’ set
but not used [-Wunused-but-set-variable]
1907 | u32 num_bands;
| ^~~~~~~~~
sound/pci/asihpi/asihpi.c: In function ‘snd_asihpi_tuner_band_put’:
sound/pci/asihpi/asihpi.c:1934:6: warning: variable ‘num_bands’ set
but not used [-Wunused-but-set-variable]
1934 | u32 num_bands;
| ^~~~~~~~~
ALSA: pci/asihpi: remove 'set but not used' warnings
Fix W=1 warnings by removing 2 unnecessary initializations and
removing a variable that's not used.
sound/pci/asihpi/asihpi.c: In function ‘snd_asihpi_tuner_band_get’:
sound/pci/asihpi/asihpi.c:1907:6: warning: variable ‘num_bands’ set
but not used [-Wunused-but-set-variable]
1907 | u32 num_bands = 0;
| ^~~~~~~~~
sound/pci/asihpi/asihpi.c: In function ‘snd_asihpi_tuner_band_put’:
sound/pci/asihpi/asihpi.c:1934:6: warning: variable ‘num_bands’ set
but not used [-Wunused-but-set-variable]
1934 | u32 num_bands = 0;
| ^~~~~~~~~
sound/pci/asihpi/asihpi.c: In function ‘snd_asihpi_mux_info’:
sound/pci/asihpi/asihpi.c:2164:6: warning: variable ‘err’ set but not
used [-Wunused-but-set-variable]
2164 | int err;
| ^~~
Fix W=1 warning by noting variable as __always_unused.
sound/isa/gus/gus_uart.c: In function ‘snd_gf1_interrupt_midi_in’:
sound/isa/gus/gus_uart.c:16:22: warning: variable ‘data’ set but not
used [-Wunused-but-set-variable]
16 | unsigned char stat, data, byte;
Vinod Koul [Mon, 29 Jun 2020 13:47:37 +0000 (19:17 +0530)]
ALSA: compress: fix partial_drain completion state
On partial_drain completion we should be in SNDRV_PCM_STATE_RUNNING
state, so set that for partially draining streams in
snd_compr_drain_notify() and use a flag for partially draining streams
While at it, add locks for stream state change in
snd_compr_drain_notify() as well.
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