commit e03fa055bc upstream.
Sjoerd Simons reports that, without using position_fix=1, recording
experiences overruns. Work around that by applying the LPIB quirk
for his hardware.
Reported-and-tested-by: Sjoerd Simons <sjoerd@debian.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit d81a12bc29 upstream.
The load_mixer_volumes() function, which can be triggered by
unprivileged users via the SOUND_MIXER_SETLEVELS ioctl, is vulnerable to
a buffer overflow. Because the provided "name" argument isn't
guaranteed to be NULL terminated at the expected 32 bytes, it's possible
to overflow past the end of the last element in the mixer_vols array.
Further exploitation can result in an arbitrary kernel write (via
subsequent calls to load_mixer_volumes()) leading to privilege
escalation, or arbitrary kernel reads via get_mixer_levels(). In
addition, the strcmp() may leak bytes beyond the mixer_vols array.
Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 6027277e77 upstream.
Add a quirk entry for Thinkpad Edge 11 as well as other TP Edge models.
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 3f343f8512 upstream.
Deemphasis control's .get callback should update control's value instead
of returning it - return value of callback function is used for indicating
error or success of operation.
Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit a096862809 upstream.
Acc to WM8580 manual, the default value for R8 is 0x10, not 0x1c.
Signed-off-by: Seungwhan Youn <sw.youn@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit ed8cc471d7 upstream.
SPKOUTL_BOOST start from third bit, SPKOUTLR_BOOST start from 0 bit.
Signed-off-by: Uk Kim <w0806.kim@samsung.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 77c4d5cdb8 upstream.
BugLink: https://launchpad.net/bugs/595482
The original reporter states that audible playback from the internal
speaker is inaudible despite the hardware being properly detected. To
work around this symptom, he uses the model=lg quirk to properly enable
both playback, capture, and jack sense. Another user corroborates this
workaround on separate hardware. Add this PCI SSID to the quirk table
to enable it for further LG P1 Expresses.
Reported-and-tested-by: Philip Peitsch <philip.peitsch@gmail.com>
Tested-by: nikhov
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit dd5a089edf upstream.
BugLink: https://launchpad.net/bugs/685161
The reporter of the bug states that he must use position_fix=1 to enable
capture for the internal microphone, so set it for his machine's PCI
SSID. Verified using 2.6.35 and the 2010-12-04 alsa-driver build.
Reported-and-tested-by: Ralph Wabel <rwabel@gmx.net>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 3dc8642903 upstream.
Commit bbbe33900d added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.
However, according to CEA-861-D no SAD is needed for basic audio
(32/44.1/48kHz stereo 16-bit audio), which is instead indicated with a
basic audio flag in the CEA EDID Extension.
The flag is not present in ELD. However, as all audio capable sinks are
required to support basic audio, we can assume it to be always
available.
Fix allowed audio formats with sinks that have SADs (Short Audio
Descriptors) which do not completely overlap with the basic audio
formats (there are no reports of affected devices so far) by always
assuming that basic audio is supported.
Reported-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 4b0dbdb17f upstream.
Commit bbbe33900d added functionality to restrict PCM parameters
based on ELD info (derived from EDID data) of the audio sink.
However, it wrongly assumes that the bits 0-2 of the first byte of
CEA Short Audio Descriptors mean a supported number of channels. In
reality, they mean the maximum number of channels (as per CEA-861-D
7.5.2). This means that the channel count can only be used to restrict
max_channels, not min_channels.
Restricting min_channels causes us to deny opening the device in stereo
mode if the sink only has SADs that declare larger numbers of channels
(like Primare SP32 AV Processor does).
Fix that by not restricting min_channels based on ELD information.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Reported-by: Jean-Yves Avenard <jyavenard@gmail.com>
Tested-by: Jean-Yves Avenard <jyavenard@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 8a96b1e020 upstream.
BugLink: http://launchpad.net/497546
Confirmed that the ideapad model works better than the current
quirk for Dell Vostro 320.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 0defe09ca7 upstream.
BugLink: https://launchpad.net/bugs/683695
The original reporter states that headphone jacks do not appear to
work. Upon inspecting his codec dump, and upon further testing, it is
confirmed that the "alienware" model quirk is correct.
Reported-and-tested-by: Cody Thierauf
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 60686aa008 upstream.
In OSS emulation, SNDCTL_DSP_RESET ioctl needs the reset of the internal
buffer state in addition to drop of the running streams. Otherwise the
succeeding access becomes inconsistent.
Tested-by: Amit Nagal <helloin.amit@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit cc1c452e50 upstream.
The patch enables ALC887-VD to use the DAC at nid 0x26,
which makes it possible to use this DAC for e g Headphone
volume.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 7167594a3d upstream.
The mixer nids passed to alc_auto_create_input_ctls are wrong: 0x15 is
a pin, and 0x09 is the ADC on both ALC660-VD/ALC861-VD. Thus with
current code, input playback volume/switches and input source mixer
controls are not created, and recording doesn't work. Select correct
mixers, 0x0b (input playback mixer) and 0x22 (capture source mixer).
Reference: https://qa.mandriva.com/show_bug.cgi?id=61159
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 5a8cfb4e8a upstream.
When SKU assid gives no valid bits for 0x38, the driver didn't take
any action, so far. This resulted in the missing initialization for
external amps, etc, thus the silent output in the end.
Especially users hit this problem on ALC888 newly since 2.6.35,
where the driver doesn't force to use ALC_INIT_DEFAULT any more.
This patch sets the default initialization scheme to use
ALC_INIT_DEFAULT when no valid bits are set for SKU assid.
Reference:
https://bugzilla.redhat.com/show_bug.cgi?id=657388
Reported-and-tested-by: Kyle McMartin <kyle@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit a0e90acc65 upstream.
BugLink: https://launchpad.net/bugs/677830
The original reporter states that the subwoofer does not mute when
inserting headphones. We need an entry for his machine's SSID in the
subwoofer pin fixup list, so add it there (verified using hda_analyzer).
Reported-and-tested-by: i-NoD
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 6cb3b707f9 upstream.
By adding the subwoofer as a speaker pin, it is treated correctly when auto-muting.
BugLink: https://launchpad.net/bugs/611803
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 0613a59456 upstream.
BugLink: https://launchpad.net/bugs/669279
The original reporter states: "The Master mixer does not change the
volume from the headphone output (which is affected by the headphone
mixer). Instead it only seems to control the on-board speaker volume.
This confuses PulseAudio greatly as the Master channel is merged into
the volume mix."
Fix this symptom by applying the hp_only quirk for the reporter's SSID.
The fix is applicable to all stable kernels.
Reported-and-tested-by: Ben Gamari <bgamari@gmail.com>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 01e0f1378c upstream.
ALC887-VD is like ALC888-VD. It can not be initialized as ALC882.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 2f7dceeda4 upstream.
MCLKDIV bit of Register 04h Clocking1:
0 : Divide by 1
1 : Divide by 2
Thus in the case of freq <= 16500000, we should clear MCLKDIV bit.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 08b1a38465 upstream.
DACSLOPE bit of Register 06h ADC and DAC Control 2:
0: Normal mode
1: Sloping stop-band mode
Thus in the case of normal mode, we should clear DACSLOPE bit.
Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 6d212d8e86 upstream.
Not all bits can be read back from POWER1 so avoid corruption when using
a read/modify/write cycle by marking it non-volatile - the only thing we
read back from it is the chip revision which has diagnostic value only.
We can re-add later but that's a more invasive change than is suitable
for a bugfix.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit ac70eb1305 upstream.
BugLink: https://launchpad.net/bugs/682199
A 2.6.35 (Ubuntu Maverick) user, burningphantom1, reported a regression
in audio: playback was inaudible through both speakers and headphones.
In commit 272a527c04 of sound-2.6.git, a new model was added with this
machine's PCI SSID. Fortunately, it is now sufficient to use the auto
model for BIOS auto-parsing instead of the existing quirk.
Playback, capture, and jack sense were verified working for both
2.6.35 and the alsa-driver snapshot from 2010-11-27 when model=auto is
used.
Reported-and-tested-by: burningphantom1
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 673f7a8984 upstream.
BugLink: https://launchpad.net/bugs/677652
The original reporter states that, in 2.6.35, headphones do not appear
to work, nor does inserting them mute the A52J's onboard speakers. Upon
inspecting the codec dump, it appears that the newly committed hp-laptop
quirk will suffice to enable this basic functionality. Testing was done
with an alsa-driver build from 2010-11-21.
Reported-and-tested-by: Joan Creus
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
[Not upstream as .37 fixes this differently in a much more complete way
that is not able to be backported easily.]
(Ported on top of 2.6.36)
BugLink: http://launchpad.net/bugs/628961
BugLink: http://launchpad.net/bugs/605047
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Diego Elio Pettenò <flameeyes@gmail.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 14d34f166c upstream.
Creative HD-audio controller chips require some workarounds:
- Additional delay before RIRB response
- Set the initial RIRB counter to 0xc0
The latter seems to be done in general in Windows driver, so we may
use this value later for all types if it's confirmed to work better.
Reported-by: Wai Yew CHAY <wychay@ctl.creative.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 24b55c69b6 upstream.
The dig_out_nid field must take a digital-converter widget, but the current
ca0110 parser passed the pin wrongly instead.
Reported-by: Wai Yew CHAY <wychay@ctl.creative.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 62b7e5e09b upstream.
Creative IBG controllers require the playback stream-tags to be started
from 1, instead of capture+1. Otherwise the stream stalls.
Reported-by: Wai Yew CHAY <wychay@ctl.creative.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
commit 0e7adbe263 upstream.
The sticky PCM stream assignment introduced in 2.6.36 kernel seems
causing problems on AD codecs. At some time later, the streaming no
longer works by unknown reason. A simple workaround is to disable
sticky-assignment for these codecs.
Tested-by: Vasily Khoruzhick <anarsoul@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Greg Kroah-Hartman <gregkh@suse.de>
When a driver module is unloaded and the last still open file is a raw
MIDI device, the card and its devices will be actually freed in the
snd_card_file_remove() call when that file is closed. Afterwards, rmidi
and rmidi->card point into freed memory, so the module pointer is likely
to be garbage.
(This was introduced by commit 9a1b64caac82aa02cb74587ffc798e6f42c6170a.)
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Reported-by: Krzysztof Foltman <wdev@foltman.com>
Cc: 2.6.30-2.6.35 <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: https://bugs.launchpad.net/bugs/653420
Add another HP DV6 notebook (103c:363e) to use STAC_HP_DV5.
Signed-off-by: Luke Yelavich <luke.yelavich@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
We shouldn't return directly here because we're still holding the
&soundcard_mutex.
This bug goes all the way back to the start of git. It's strange that
no one has complained about it as a runtime bug.
CC: stable@kernel.org
Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'fix/misc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
ALSA: i2c/other/ak4xx-adda: Fix a compile warning with CONFIG_PROCFS=n
ALSA: prevent heap corruption in snd_ctl_new()
The snd_ctl_new() function in sound/core/control.c allocates space for a
snd_kcontrol struct by performing arithmetic operations on a
user-provided size without checking for integer overflow. If a user
provides a large enough size, an overflow will occur, the allocated
chunk will be too small, and a second user-influenced value will be
written repeatedly past the bounds of this chunk. This code is
reachable by unprivileged users who have permission to open
a /dev/snd/controlC* device (on many distros, this is group "audio") via
the SNDRV_CTL_IOCTL_ELEM_ADD and SNDRV_CTL_IOCTL_ELEM_REPLACE ioctls.
Signed-off-by: Dan Rosenberg <drosenberg@vsecurity.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SNDRV_HDSP_IOCTL_GET_CONFIG_INFO and
SNDRV_HDSP_IOCTL_GET_CONFIG_INFO ioctls in hdspm.c and hdsp.c allow
unprivileged users to read uninitialized kernel stack memory, because
several fields of the hdsp{m}_config_info structs declared on the stack
are not altered or zeroed before being copied back to the user. This
patch takes care of it.
Signed-off-by: Dan Rosenberg <dan.j.rosenberg@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The SPDIF in audio widget must be searched through the list as the widget
that contains the given pin as the connection source. The current code
was implemented in a reverse way.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Make sure we stay within the cache boundaries when updating the
register cache.
Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
On the HT-Omega Claro halo card, the ADC data must be captured from the
second I2S input. Using the default first input, which isn't connected
to anything, would result in silence.
Signed-off-by: Erik J. Staab <ejs@insightbb.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The clkdev API doesn't use .name and .id members of struct clk for clock
lookup. Instead clocks should be added to a lookup list. Without this patch
audio om the Migo-R board fails silently.
Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
The PCM proc files may open a race against substream close, which can
end up with an Oops. Use the open_mutex to protect for it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
