- much improved implementation due to clean codec hierarchy
- preparation for potential per-codec spinlock change
NOTE: additionally removes a chip->pcm[codec_type] NULL ptr check
(due to it requiring access to external chip struct),
however I believe this to be ok since this condition should not occur
and most drivers don't check against that either.
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- use a separate variable for the frequency part, don't always "or" it
- use a "clever"(?) macro to shorten the code
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
- correct samples to be POSIX shell compatible
- add logging of jiffies value in _pointer()
- several comments
- cleanup
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The assignment to the local variable 'channel' in
snd_ca0106_pcm_pointer_capture() is a little crazy. Order of assignment is
undefined. This fixes it.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Jiri Kosina <jkosina@suse.cz>
The fix-up entries by the commit 2785591a97
ALSA: hda - Add fix-up for Sony VAIO with ALC275 codecs
weren't applied in the right position. They had to be before the quirk
entry matching to all Sony devices.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a mixer control element was already created with the given name,
try to find another index for avoiding conflicts, instead of breaking
with an error. This makes the driver more robust.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When the d-mics are assigned to the same purpose of another analog mic
pins, the driver doesn't compute the index properly, resulting in an
error with "existing control". This patch fixes it.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Conflicts:
MAINTAINERS
arch/arm/mach-omap2/pm24xx.c
drivers/scsi/bfa/bfa_fcpim.c
Needed to update to apply fixes for which the old branch was too
outdated.
ALC275 doesn't require the ALC269 (and its variants) specific init
sequences. Add the check of codec id.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set GPIO2 for some Sony VAIO with ALC275 to fix speaker output.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Change non-standard mic control names to standard control names
to clean up the namespace.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Usually external microphones are just labelled "Mic", so rename
"Ext Mic" and "External Mic" to "Mic" to clear up the namespace.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
"Int Mic" and "Internal Mic" both mean the same thing, so rename
the former to the latter in order to clean up the namespace a little.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/580006
SKU turns off auto-mute for these machines, so ignore the SKU.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Due to the recent change for multiple mics assignment, we need to handle
the index of each Mic Boost control respectively. Otherwise the driver
gets the control element conflicts, and gives the unsable state.
Reference: kernel bug 25002
https://bugzilla.kernel.org/show_bug.cgi?id=25002
Reported-and-tested-by: Adam Williamson <awilliam@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Multiple quirk functions were using the exact same code to verify if the Mic
jack was plugged and mute the Mic accordingly
Signed-off-by: Anisse Astier <anisse@astier.eu>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Set default association/sequence right on pin 0x17 in order for
the automatic parser to recognize the subwoofer correctly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/690530
The SKU value of this machine dictates that auto-mute should be
disabled. Since the SKU value is similar to the PCI SSID, the most
likely conclusion is that the SKU value should be ignored.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Mute speakers when a line-out jack is plugged as well as headphone jacks
with the new Conexant codec parser in the auto mode.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
flush_scheduled_work() is deprecated and scheduled to be removed.
* cancel[_delayed]_work() + flush_scheduled_work() ->
cancel[_delayed]_work_sync().
* wm8350, wm8753 and soc-core use custom code to cancel a delayed
work, execute it immediately if it was pending and wait for its
completion. This is equivalent to flush_delayed_work_sync(). Use
it instead.
Signed-off-by: Tejun Heo <tj@kernel.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Rev. E of the M-Audio Delta 66 is partially supported (commit
ef2cd2ccad), but the layout of the GPIO
pins was still unclear. This patch adds the GPIO definitions so that
communication to the CS8247 & 2x AK4524 works correctly.
ALSA bug#3327 has more details; users cap & jhunt report there that the
GPIO wiring is similar to the Digigram VX442 (chip select: pin 4 =
CS8427, pin 5 = AK4524 #0, pin 6 = AK4524 #1). There has been a lot of
conflicting information in the bug, but given these definitions, my
Delta 66E works; I tested analog in&out at 44.1kHz & 96kHz, analog gain
settings, S/PDIF clock sync, and S/PDIF in&out at 44.1kHz.
Signed-off-by: Brian Bloniarz <brian.bloniarz@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/497546
Confirmed that the ideapad model works better than the current
quirk for Dell Vostro 320.
Cc: stable@kernel.org (2.6.35+)
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
create fixup function for the mario model and override amp capabilities
for NID 0x2
Signed-off-by: Todd Broch <tbroch@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Facilitate fixup for realtek codecs via modelname lookup of fixup
data. Fallback to quirk based lookup in absence of model definition.
Signed-off-by: Todd Broch <tbroch@chromium.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some newer chips have more than one HDMI output, but usually not
all of them are exposed as physical jacks. Removing the unused
PCM devices (as indicated by BIOS in the pin config default) will
reduce user confusion as they currently have to choose between
several HDMI devices, some of them not working anyway.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When a new HDMI/DP device is plugged in, hdmi_update_short_audio_desc()
is called for every SAD (Short Audio Descriptor) in the ELD data. For
LPCM coding type SAD defines the supported sample sizes. For several
other coding types (such as AC-3), a maximum bitrate is defined.
The maximum bitrate and sample size fields are not always cleared.
Therefore, if a device is unplugged and a different one is plugged in,
and the coding types of some SAD positions differ between the devices,
the old max_bitrate or sample_bits values will persist if the new SADs
do not define those values.
The leftover max_bitrate and sample_bits do not cause any issues other
than wrongly showing up in eld#X.Y procfs file and kernel log.
Fix that by always clearing sample_bits and max_bitrate when reading
SADs.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Reformat and update the comments that describe the hardware connections
on the various models.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of the hardcoded "CMI8788", show the actual chip name.
Note: This is neither what the chip is (it's always the same),
nor what the chip is actually called.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
To help with debugging, add the registers of the model-specific
codecs to the controller and AC97 register dump in the proc file.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The "Front Panel" switch on the Xonar D1/DX actually switches only the
output direction, so mark it appropriately.
The front panel microphone is controlled by the FMIC2MIC bit of the
CM9780. It was unconditionally enabled on the D1/DX and never set on
the ST(X); add a control for it. Selecting the front panel microphone
as source does not actually disable the microphone jack, but this is
bug-compatible with the Windows driver, and users rely on it.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The GPIO bit that enables analog output on the Xonar HDAV1.3 also
disables the HDMI audio output, so we better add a switch for it.
Hopefully, this is sufficient to make the HDMI output work.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Initialize the configuration of some unknown GPIO output bits (that
might not be used at all) to be the same as in the Windows driver, just
to be sure.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Switch to the generic hdmi parser for codec id 1002:aa01 (ATI R6xx
HDMI), as the codec appears to work fine with it.
Note that the codec is still limited to stereo output only, despite it
reportedly being multichannel capable. Some as of yet unknown quirks
will be needed to get that working.
Testing was done on 2.6.36 by John Ettedgui.
Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Tested-by: John Ettedgui <john.ettedgui@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Added a quirk to cxt5066_cfg_tbl to enable jack sense for ThinkPad Edge 13.
Reference: http://launchpad.net/bugs/685015
Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Instead of hard-coded magic numbers, properly define and use macros
for improve the readability. Also, dell_automute is handled samely
as thinkpad, since it also sets port_d_mode, too.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
On the docking station for the Lenovo T410 and T410s, the line-out
doesn't work. The trouble seems to be that it generates a plug event,
but then doesn't report that the jack is connected. So automute mutes
the jack when you plug something into it. The following patch (next
message) fixes it.
Signed-off-by: John Baboval <john.baboval at virtualcomputer.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the RME HDSP RPM IO box. Changes have been made in the identification of the IO box and the neccessary controls have been added.
Signed-off-by: Florian Faber <faberman@linuxproaudio.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
The refactoring commit d433a67831
ALSA: hda - Optimize the check of ALC269 codec variants
introduced a wrong check for ALC269-vb type. This patch corrects it.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Add a generic callback function for fixup elements. This can be used
to do some unusual things like overriding the AMP cache, etc.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALC887-VD is like ALC888-VD. It can not be initialized as ALC882.
Signed-off-by: Kailang Yang <kailang@realtek.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
. Fix PulseAudio "ALSA driver bug" issue
(if we have two alternated areas within a 64k DMA buffer, then max
period size should obviously be 32k only).
Back references:
http://pulseaudio.org/wiki/AlsaIssueshttp://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction
When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.
PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.
Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Allow disabling period wakeup interrupts for all PCM streams.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
Cc: <stable@kernel.org> [2.6.35+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
. Fix PulseAudio "ALSA driver bug" issue
(if we have two alternated areas within a 64k DMA buffer, then max
period size should obviously be 32k only).
Back references:
http://pulseaudio.org/wiki/AlsaIssueshttp://fedoraproject.org/wiki/Features/GlitchFreeAudio
. In stop timer function, need to supply ACK in the timer control byte.
. Minor log output correction
When I did my first PA testing recently, the period size bug resulted
in quite precisely observeable half-period-based playback distortion.
PA-based operation is quite a bit more underrun-prone (despite its
zero-copy optimizations etc.) than raw ALSA with this rather spartan
sound hardware implementation on my puny Athlon.
Note that even with this patch, azt3328 still doesn't work for both
cases yet, PA tsched=0 and tsched
(on tsched=0 it will playback tiny fragments of periods, leading to tiny
stuttering sounds with some pauses in between, whereas with
timer-scheduled operation playback works fine - minus some quite increased
underrun trouble on PA vs. ALSA, that is).
Signed-off-by: Andreas Mohr <andi@lisas.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/669092
ALC887 does not have any volume control ability on the mixer NIDs,
so put the volume controls on the dac NIDs instead. Without this
patch, ALC887 users cannot use alsamixer at all.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
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>
Cc: <stable@kernel.org> [2.6.32+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
In each function, the value apcm is stored in the private_data field of
runtime. At the same time the function ct_atc_pcm_free_substream is stored
in the private_free field of the same structure. ct_atc_pcm_free_substream
dereferences and ultimately frees the value in the private_data field. But
each function can exit in an error case with apcm having been freed, in
which case a subsequent call to the private_free function would perform a
dereference after free. On the other hand, if the private_free field is
not initialized, it is NULL, and not invoked (see snd_pcm_detach_substream
in sound/core/pcm.c). To avoid the introduction of a dangling pointer, the
initializations of the private_data and private_free fields are moved to
the end of the function, past any possible free of apcm. This is safe
because the previous calls to snd_pcm_hw_constraint_integer and
snd_pcm_hw_constraint_minmax, which take runtime as an argument, do not
refer to either of these fields.
In each function, there is one error case where apcm needs to be freed, and
a call to kfree is added.
The sematic match that finds this problem is as follows:
(http://coccinelle.lip6.fr/)
// <smpl>
@@
expression e,e1,e2,e3;
identifier f,free1,free2;
expression a;
@@
*e->f = a
... when != e->f = e1
when any
if (...) {
... when != free1(...,e,...)
when != e->f = e2
* kfree(a)
... when != free2(...,e,...)
when != e->f = e3
}
// </smpl>
Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
If the platform already provides a definition for these accessors
do not redefine them. The warning was caught on MIPS.
Signed-off-by: Florian Fainelli <florian@openwrt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://launchpad.net/bugs/673075
According to the datasheet of 92HD87B, there is a digital mic
at nid 0x11, so enable it in order to be able to use the mic.
Cc: stable@kernel.org
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the TempoTec/MediaTek HiFier Serenade sound card.
The PCI ID was already there, but the driver handled it like the
Fantasia model, which resulted in a dummy recording device. As
a stereo output-only card, this model is to be handled exactly
like the HG2PCI.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Sort the PCI IDs so that they make logical sense. Also move the card
name comments into this list because the model symbols should be (more)
self-explanationary.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Add support for the Kuroutoshikou CMI8787-HG2PCI sound card.
[replaced non-latin letters in the patch by tiwai]
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
The snd-hifier driver contains more duplicated code than model-specific
code, so it does not make sense for it to be a separate driver.
Handling the two-channel output restriction can be easily done in the
generic driver.
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
This patch add support for the MacBookAir3,1 and MacBookAir3,2 to the alsa
sound system.
Signed-off-by: Edgar (gimli) Hucek <gimli@dark-green.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
I noticed that sound/pci/asihpi/hpicmn.c::hpi_alloc_control_cache() does
not check the return value from kmalloc(), which may fail.
If kmalloc() fails we'll dereference a null pointer and things will go bad
fast.
There are two memory allocations in that function and there's also the
problem that the first may succeed and the second may fail and nothing is
done about that either which will also go wrong down the line.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Acked-by: Eliot Blennerhassett <linux@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
converts a 1 bit signed bitfield to an unsigned.
Reported-by: Dr. David Alan Gilbert <linux@treblig.org>
Signed-off-by: Tim Blechmann <tim@klingt.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
When reading through sound/pci/cs46xx/dsp_spos.c I noticed a couple of
things in cs46xx_dsp_spos_create().
It seems to me that we don't always free the various memory buffers we
allocate and we also do some work (structure member assignment) early,
that is completely pointless if some of the memory allocations fail and
we end up just aborting the whole thing.
I don't have hardware to test, so the patch below is compile tested only,
but it makes the following changes:
- Make sure we always free all allocated memory on failures.
- Don't do pointless work assigning to structure members before we know
all memory allocations, that may abort progress, have completed
successfully.
- Remove some trailing whitespace.
Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Tested-by: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Some HP laptops have lower amplifier levels for speakers in comparison
with headphone outputs. This patch changes the BTL amp level for these
machines to balance both the speaker and headphone output levels.
Signed-off-by: Vitaliy Kulikov <Vitaliy.Kulikov@idt.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (365 commits)
ALSA: hda - Disable sticky PCM stream assignment for AD codecs
ALSA: usb - Creative USB X-Fi volume knob support
ALSA: ca0106: Use card specific dac id for mute controls.
ALSA: ca0106: Allow different sound cards to use different SPI channel mappings.
ALSA: ca0106: Create a nice spot for mapping channels to dacs.
ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence.
ALSA: ca0106: Pull out dac powering routine into separate function.
ALSA: ca0106 - add Sound Blaster 5.1vx info.
ASoC: tlv320dac33: Use usleep_range for delays
ALSA: usb-audio: add Novation Launchpad support
ALSA: hda - Add workarounds for CT-IBG controllers
ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs
ASoC: tpa6130a2: Error handling for broken chip
ASoC: max98088: Staticise m98088_eq_band
ASoC: soc-core: Fix codec->name memory leak
ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066
ALSA: hda - Add some workarounds for Creative IBG
ALSA: hda - Fix wrong SPDIF NID assignment for CA0110
ALSA: hda - Fix codec rename rules for ALC662-compatible codecs
ALSA: hda - Add alc_init_jacks() call to other codecs
...
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>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
