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commit d17344b3547669f5b6ee4fda993d03737a141bd6 upstream.
There was a race condition when the system suspends while hda_power_work
is running in the work queue. If system suspend (snd_hda_suspend)
happens after the work queue releases power_lock but before it calls
hda_call_codec_suspend, codec_suspend runs with power_on=0, causing the
codec to power up for register reads, and hanging when it calls
cancel_delayed_work_sync from the running work queue.
The call chain from the work queue will look like this:
hda_power_work <<- power_on = 1, unlock, then power_on cleard by suspend
hda_call_codec_suspend
hda_set_power_state
snd_hda_codec_read
codec_exec_verb
snd_hda_power_up
snd_hda_power_save
__snd_hda_power_up
cancel_delayed_work_sync <<-- cancelling executing wq
Fix this by waiting for the work queue to finish before starting suspend
if suspend is not happening on the work queue.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 90accc58a6946e7245993da6079f88d8c29cb731 upstream.
DMA Position in Buffer (DPIB) should be used for
ring buffer management, while LPIB register provides
information on the number of samples transfered on
the link. The difference between the two pieces of
information corresponds to hardware/DMA buffering.
This patch reports this difference in runtime->delay, and
removes the use of the COMBO mode on recent Intel hardware.
Credits to Takashi Iwai for an initial patch.
[rebased to for-next branch and replaced snd_printk() with
snd_printdd() by tiwai]
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit d279fae8a41690ec1b20c07be8c6f42f8af27a17 upstream.
A new PCI id 0x0d0c for Haswell HDA Controller.
[root@SKBM04SDP ~]# lspci |grep Audio
00:03.0 Audio device: Intel Corporation Device 0d0c (rev 02)
00:1b.0 Audio device: Intel Corporation Lynx Point HD Audio Controller
Signed-off-by: Wang Xingchao <xingchao.wang@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c05fce586d4da2dfe0309bef3795a8586e967bc3 upstream.
Added support for Xbox Communicator to USB quirks.
Signed-off-by: Marko Friedemann <mfr@bmx-chemnitz.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 2b58fd5b3193fd3af3d15114d95706087d25a7fe upstream.
Playback Designs' USB devices have some hardware limitations on their
USB interface. In particular:
- They need a 20ms delay after each class compliant request as the
hardware ACKs the USB packets before the device is actually ready
for the next command. Sending data immediately will result in buffer
overflows in the hardware.
- The devices send bogus feedback data at the start of each stream
which confuse the feedback format auto-detection.
This patch introduces a new quirks hook that is called after each
control packet and which adds a delay for all devices that match
Playback Designs' USB VID for now.
In addition, it adds a counter to snd_usb_endpoint to drop received
packets on the floor. Another new quirks function that is called once
an endpoint is started initializes that counter for these devices on
their sync endpoint.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Andreas Koch <andreas@akdesigninc.com>
Supported-by: Demian Martin <demianm_1@yahoo.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit c10514394ef9e8de93a4ad8c8904d71dcd82c122 upstream.
While going through Ubuntu bugs, I discovered this patch being
posted and a confirmation that the patch works as expected.
Finding out how the hw volume really works would be preferrable
to just disabling the broken one, but this would be better than
nothing.
Credit: sndfnsdfin (qawsnews)
BugLink: https://bugs.launchpad.net/bugs/559939
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 4b527b6516ab1f0af8aaedd02dbf71ee2c1180f4 upstream.
When the input gain for the internal mic is set to its maximum level,
the background noise becomes so high - and any relevant signal clipped -
that the setting becomes unusable. It is better to limit the amplification.
BugLink: https://bugs.launchpad.net/bugs/1052460
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 9f720bb9409ea5923361fbd3fdbc505ca36cf012 upstream.
In commit af741c1 ("ALSA: hda/realtek - Call alc_auto_parse_customize_define()
always after fixup"), alc_auto_parse_customize_define was moved after
detection of ALC271X.
The problem is that detection of ALC271X relies on spec->cdefine.platform_type,
and it's set on alc_auto_parse_customize_define.
Move the alc_auto_parse_customize_define and its required fixup setup
before the block doing the ALC271X and other codec setup.
BugLink: https://bugs.launchpad.net/bugs/1006690
Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
Reviewed-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit cf55e904516947597d75fd3844acc24891a95772 upstream.
If headphone jack can't detect plug presence, and we have the jack in
the jack table, snd_hda_jack_detect will return the plug as always
present (as it'll be considered as a phantom jack). The problem is that
when this happens, line out pins will always be disabled, resulting in
no sound if there are no headphones connected.
This was reported as a no sound problem after suspend on
http://bugs.launchpad.net/bugs/1052499, since the bug doesn't manifests
on first initialization before the phantom jack is added, but on resume
we reexecute the initialization code, and via_hp_automute starts
reporting HP always present with the jack now on the table.
BugLink: https://bugs.launchpad.net/bugs/1052499
Signed-off-by: Herton Ronaldo Krzesinski <herton.krzesinski@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit d4f1e48bd11e3df6a26811f7a1f06c4225d92f7d upstream.
When the loopback timer handler is running, calling del_timer() (for STOP
trigger) will not wait for the handler to complete before deactivating the
timer. The timer gets rescheduled in the handler as usual. Then a subsequent
START trigger will try to start the timer using add_timer() with a timer pending
leading to a kernel panic.
Serialize the calls to add_timer() and del_timer() using a spin lock to avoid
this.
Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah@linux.intel.com>
Signed-off-by: Vinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit e4db0952e542090c605fd41d31d761f1b4624f4a upstream.
The Lenovo IdeaPad U310 has an internal mic where the right channel
is phase inverted.
Signed-off-by: Felix Kaechele <felix@fetzig.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit 689185b78ba6fbe0042f662a468b5565909dff7a upstream.
Help UIs associate it with the matching gain control.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit ae60503741991a36ed6b2a8f53b249b2a72af52b upstream.
The volume update bits were being set on all but one input and one output.
Signed-off-by: Charles Keepax <ckeepax@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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commit eb4d5fc1f0ce89e3d5b072c594a1e213a0e05881 upstream.
In some circumstances we may need to flush volume updates to the device
after switching to class W mode. Do this unconditionally to ensure that
these situations are handled.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>
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Also fix the calls to next_packet_size() for the pause case. This was
missed in 245baf983 ("ALSA: snd-usb: fix calls to next_packet_size").
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reviewed-by: Takashi Iwai <tiwai@suse.de>
Reported-and-tested-by: Christian Tefzer <ctrefzer@gmx.de>
Cc: stable@kernel.org
[ Taking directly because Takashi is on vacation - Linus ]
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
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Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for 3.6
A bigger set of updates than I'm entirely comfortable with - things
backed up a bit due to travel. As ever the majority of these are small,
focused updates for specific drivers though there are a couple of core
changes. There's been good exposure in -next.
The AT91 patch fixes a build break.
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Signed-off-by: Bo Shen <voice.shen@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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ASUS X53S also suffers from the same issue as in commit c302d6133.
Use POS_FIX_POSBUF for this hardware, too.
Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=47461
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The AK4396 DAC has a linear-scale attentuator, but
sound/pci/ice1712/prodigy_hifi.c used a log scale instead, which is
not quite right. This patch restores the correct scale, borrowing
from the ak4396 code in sound/pci/oxygen/oxygen.c.
Signed-off-by: Matteo Frigo <athena@fftw.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit c20c5a841cbe47f5b7812b57bd25397497e5fbc0 changed some chipsets to
default to POS_FIX_COMBO so they now use POS_FIX_LPIB instead of
POS_FIX_POSBUF. Since then I've been getting artifacts on playback, including
repeated sounds on my Asus laptop.
My hardware is Cougar Point which the commit log of
c20c5a841cbe47f5b7812b57bd25397497e5fbc0 mentions as tested so POS_FIX_COMBO
probably works in general but apparently it doesn't on Asus K53E therefore the
need for the quirk.
Signed-off-by: Catalin Iacob <iacobcatalin@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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O_RDONLY is zero so the original test (f->f_flags & O_RDONLY) is always
false and it will never do compress capture. The test for O_WRONLY is
also slightly off. The original test would consider "->flags =
(O_WRONLY | O_RDWR)" as write only instead of rejecting it as invalid.
I've also removed the pr_err() because that could flood dmesg.
Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_hda_codec_reset() calls restore_pincfgs() where the codec is
powered up again, which eventually tries to resume and initialize via
the callbacks of the codec. However, it's the place just after codec
free callback, thus no codec callbacks should be called after that.
On a codec like CS4206, it results in Oops due to the access in init
callback.
This patch fixes the issue by clearing the codec callbacks properly
after freeing codec.
Reported-by: Daniel J Blueman <daniel@quora.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The I2S controllers are programmed with an "attention" level of 4 DWORDs.
This must match the configuration passed to the DMA driver, so that when
they burst in data, they don't overflow the available FIFO space. Also,
the burst size is relevant to the destination for playback, and source
for capture, not vice-versa as originally written.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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The recent fix for the missing fine delayed time adjustment gives
strange error messages at each start of the playback stream, such as
delay: estimated 0, actual 352
delay: estimated 353, actual 705
These come from the sanity check in retire_playback_urb(). Before the
stream is activated via start_endpoints(), a few silent packets have
been already sent. And at this point the delay account is still in
the state as if the new packets are just queued, so the driver gets
confused and spews the bogus error messages.
For fixing the issue, we just need to check whether the received
packet is valid, whether it's zero sized or not.
Reported-by: Markus Trippelsdorf <markus@trippelsdorf.de>
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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The pause and resume operations indicate that the stream can be
un-paused/resumed from the exact location they were paused/suspended.
This is not true for this driver, the pause and suspend triggers share
the same code path with stop, they flush all pending DMA transfers.
This drops all pending samples. The pause_release/resume triggers are
the same as start, except that prepare won't be called beforehand,
nothing will be enqueued to the DMA engine and nothing will happen (no
audio). Removing the pause flag will let apps know that it isn't
supported. Removing the resume flag will cause user space to call
prepare and start instead of resume, so audio will continue playing when
the system wakes up.
Before removing the pause and resume flags, I tested this on an exynos
5250, using 'aplay -i'. Pause/un-pause leads to silence followed by a
write error. Suspend/resume testing led to the same result. Removing
the two flags fixes suspend/resume (since snd_pcm_prepare is called
again). And leads to a proper reporting of pause not supported.
Signed-off-by: Dylan Reid <dgreid@chromium.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@vger.kernel.org
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With the commit [2faa3bf: ALSA: hda - Rewrite the mute-LED hook with
vmaster hook in patch_sigmatel.c], the former Master volume control
was converted to PCM. This was supposed to be covered by the vmaster
control. But due to the lack of "PCM" slave definition, this didn't
happen properly. The patch fixes the missing entry.
Reported-by: Andrew Shadura <bugzilla@tut.by>
Cc: <stable@vger.kernel.org> [v3.4+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Playing a mono track on a mc13783 codec results in incorrect playback rate.
Remove mono support so that a mono track can be played correctly.
Signed-off-by: Fabio Estevam <fabio.estevam@freescale.com>
Tested-by: Gaëtan Carlier <gcembed@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Signed-off-by: Heather Lomond <hlomond@opensource.wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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The if condition
if (!buf && !buf->area)
checks if the buf pointer is NULL and then dereferences it again to
check if the buffer area is NULL, resulting in possible NULL
dereference.
Signed-off-by: Prasad Joshi <prasadjoshi.linux@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Commit 68e67f40b ("ALSA: snd-usb: move calls to usb_set_interface")
saved us some unnecessary calls to snd_usb_set_interface() but ignored
the fact that there is at least one device out there which operates on
two endpoint in different interfaces simultaniously.
Take care for this by catching the case where data and sync endpoints
are located on different interfaces and calling snd_usb_set_interface()
between the start of the two endpoints.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-by: Robert M. Albrecht <linux@romal.de>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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In order to support devices with implicit feedback streaming models,
packet sizes are now stored with each individual urb, and the PCM
handling code which fills the buffers purely relies on the size fields
now.
However, calling snd_usb_audio_next_packet_size() for all possible
packets in an URB at once, prior to letting the PCM code do its job
does in fact not lead to the same behaviour than what the old code did:
The PCM code will break its loop once a period boundary is reached,
consequently using up less packets that it really could.
As snd_usb_audio_next_packet_size() implements a feedback mechanism to
the endpoints phase accumulator, the number of calls to that function
matters, and when called too often, the data rate runs out of bounds.
Fix this by making the next_packet function public, and call it from the
PCM code as before if the packet data sizes are not defined.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Parts of commit 294c4fb8 ("ALSA: usb: refine delay information with USB
frame counter") were unfortunately lost during the refactoring of the
snd-usb driver in 3.5.
This patch adds them back, restoring the correct delay information
behaviour.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_usb_endpoint_free() frees the structure that contains its argument.
Signed-off-by: Pavel Roskin <proski@gnu.org>
Cc: stable@vger.kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in
PCM capture stream") fixed a scheduling-while-atomic bug that happened
when snd_usb_endpoint_start was called from the trigger callback, which
is an atmic context. However, the patch breaks the idea of the endpoints
reference counting, which is the reason why the driver has been
refactored lately.
Revert that commit and let snd_usb_endpoint_start() take care of the URB
cancellation again. As this function is called from both atomic and
non-atomic context, add a flag to denote whether the function may sleep.
Signed-off-by: Daniel Mack <zonque@gmail.com>
Cc: stable@kernel.org [3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Both the schematics and practical testing show that the HP detect GPIO
is high when the headphones are plugged in. Hence, the snd_soc_jack_gpio
should not specify to invert the signal.
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Andrey Danin <danindrey@mail.ru>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: <stable@vger.kernel.org> # v3.4 v3.5
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These codecs seem reporting EPSS but require longer delay for the
proper D3 transition. For example, D3_STOP_CLOCK_OK bit won't be set
correctly even after D3.
In this patch, codec->epss flag is overridden for avoid the
misbehavior.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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EPSS parameter should be static, so we can read it once and remember.
This also allows more easily to override the wrong EPSS capability
reported from a codec by changing the flag in the codec
initialization step.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means
that any DAPM context being updated will have the bias level automatically
set, including the card. We can't safely do this as the card callbacks are
called for each device context and so the management of the card bias is
more complex. Several multi-component cards rely on this behaviour.
Skip updates during the asynchronous run entirely. We should really do them
in the synchronous section but it's not 100% clear which values to pick as
the different DAPM contexts may have different bias levels.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated)
ensures that we update non-CODEC DAPM contexts but means that if a
CODEC has no set_bias_level() operation it'll not be updated. Fix
that.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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This fixes an issue with a machine where there were no speakers,
but GPIO0 had to be data=1 for the headphone to be functioning.
I'm not sure if we need a more advanced patch to solve all possible cases,
but if so, this patch would still provide a minor optimisation.
BugLink: https://bugs.launchpad.net/bugs/1040077
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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snd_card_als100_probe() does not set pcm field in struct snd_sb.
As a result, PCM is not suspended and applications don't know that they need
to resume the playback.
Tested with Labway A381-F20 card (ALS120).
Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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It was forgotten to initialize ret to the result of calling
snd_soc_dai_set_sysclk, unlike at the other calls in the same function.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
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ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Acked-by: Jarkko Nikula <jarkko.nikula@bitmer.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Remove unnecessary calls to devm_kfree and replace iounmap by devm_iounmap
(and use resource_size for the third argument). These changes make it
possible to remove the error-handling code at the end of
ux500_msp_i2s_init_msp, and all of the gotos become direct returns.
In the case of the second call to devm_kzalloc, the return variable ret was
not initialized. Here it is changed to a direct return of -ENOMEM.
A simplified version of the semantic match that finds the second problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
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ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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Initialize ret on the second call to imx_audmux_v2_configure_port so that
the subsequent test checks that result and not the previous one.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
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ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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When the codec turn-on operation is canceled by the immediate
power-on, the driver left the power_transition flag as is.
This caused the persistent avoidance of power-save behavior.
Cc: <stable@vger.kernel.org> [v3.5+]
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Additional updates for 3.6
A batch more bugfixes, all driver-specific and fairly small and
unremarkable in a global context. The biggest batch are for the newly
added Arizona drivers.
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This is a series of fixes for CA0132, especially the missing SPDIF I/O
and the mixer build errors.
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It's possible that these amps are settable somehow, e g through
secret codec verbs, but for now, don't create the controls (as
they won't be working anyway, and cause errors in amixer).
Cc: stable@kernel.org
BugLink: https://bugs.launchpad.net/bugs/1038651
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
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Initialize ret before returning on failure, as done elsewhere in the
function.
A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)
// <smpl>
(
if@p1 (\(ret < 0\|ret != 0\))
{ ... return ret; }
|
ret@p1 = 0
)
... when != ret = e1
when != &ret
*if(...)
{
... when != ret = e2
when forall
return ret;
}
// </smpl>
Signed-off-by: Julia Lawall <Julia.Lawall@lip6.fr>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
|
