Takashi Iwai [Thu, 13 Jan 2011 14:41:45 +0000 (15:41 +0100)]
ALSA: hda - Add fixup-call in init callback
In some cases, the fix-up is required in the init callback to be called
both at the first initialization and at the resume. The new action type
ALC_FIXUP_ACT_INIT is used for this case.
Takashi Iwai [Thu, 13 Jan 2011 13:22:32 +0000 (14:22 +0100)]
ALSA: hda - Reorganize fixup structure for Realtek
Instead of keeping various data types in a single record, put the
type field and keep a single value in each entry, but allows chaining
multiple fixup entries. This allows more flexible data management
(see ALC275_FIXUP_SONY_HWEQ for example).
Takashi Iwai [Tue, 11 Jan 2011 17:11:04 +0000 (18:11 +0100)]
ALSA: hda - Add static_hdmi_pcm option to HDMI codec parser
The dynamic PCM restriction based on ELD information may lead to the
problem in some cases, e.g. when the receiver is turned off. Then it
may send a TV HDMI default such as channels = 2. Since it's still
plugged, the driver doesn't know whether it's the right configuration
for future use. Now, when an app opens the device at this moment,
then turn on the receiver, the app still sends channels=2.
The right solution is to implement some kind of notification and
automatic re-open mechanism. But, this is a goal far ahead.
This patch provides a workaround for such a case by providing a new
module option static_hdmi_pcm for snd-hda-codec-hdmi module. When
this is set to true, the driver doesn't change PCM parameters per
ELD information. For users who need the static configuration like
the scenario above, set this to true.
The parameter can be changed dynamically via sysfs, too.
Takashi Iwai [Mon, 10 Jan 2011 14:45:23 +0000 (15:45 +0100)]
ALSA: hda - Add support for multiple headphone/speaker controls for Realtek
So far, Realtek auto-parser assumed that the multiple pins are only for
line-outs, and assigned the channel names like Front, Surround, etc for
the multiple outputs. But, there are devices that have multiple
headphones, and these can be better controlled with the corresponding
control-name like "Headphone" with indicies.
Takashi Iwai [Mon, 10 Jan 2011 13:47:35 +0000 (14:47 +0100)]
ALSA: hda - Fix multi-headphone handling for Realtek codecs
When multiple headphone pins are defined without line-out pins, the
driver takes them as primary outputs. But it forgot to set line_out_type
to HP by assuming there is some rest of HP pins. This results in some
mis-handling of these pins for Realtek codec parser. It takes as if
these are pure line-out jacks.
Reported-and-tested-by: Fernando Lemos <fernandotcl@gmail.com> Cc: <stable@kernel.org> Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: HDA: Add Lenovo vendor quirk for Conexant 205xx
BugLink: http://bugs.launchpad.net/bugs/689036
Many new Lenovos need the ideapad quirk. Also, since the
auto parser for this chip is far from optimal, the regression
risk is low (although not zero).
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: HDA: Fix volume control indices for Mics (Realtek)
If more than one mic is present with different locations,
e g "Front Mic" and "Rear Mic", they can use the same index (0),
since their names are different.
Previous behavior was to have "Front Mic" as index 1, causing it
to be ignored by e g PulseAudio.
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
ALSA: HDA: Rename "Mic Boost" to "Mic Boost Volume"
BugLink: http://bugs.launchpad.net/bugs/697240
If the "Volume" suffix is not given, alsa-lib gets confused and
loses the dB information at the simple element level.
Boosts generally affects both playback and capture, as they are
applied early in the chain. Hence no "Playback" or "Capture" in
the suffix.
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
BugLink: http://bugs.launchpad.net/bugs/696493
According to datasheet (and real-world testing), IDT 92HD88B can
have internal mics at NID 0x11 and 0x20, so enable them accordingly.
Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Daniel T Chen [Tue, 28 Dec 2010 22:20:02 +0000 (17:20 -0500)]
ALSA: hda: Use LPIB quirk for Dell Inspiron m101z/1120
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>
Dan Rosenberg [Sat, 25 Dec 2010 21:23:40 +0000 (16:23 -0500)]
sound: Prevent buffer overflow in OSS load_mixer_volumes
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.
Takashi Iwai [Thu, 23 Dec 2010 15:35:34 +0000 (16:35 +0100)]
ALSA: hda - Fix GPIO2-fixup for Sony laptops
The fix-up entries by the commit 2785591a9760c677a7ee6f541e751c23086f5bfd
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.
Takashi Iwai [Thu, 23 Dec 2010 09:17:52 +0000 (10:17 +0100)]
ALSA: hda - Try to find an empty control index when it's occupied
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.
Takashi Iwai [Thu, 23 Dec 2010 09:16:05 +0000 (10:16 +0100)]
ALSA: hda - Fix conflict of d-mic capture volume controls
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.
Jesper Juhl [Mon, 20 Dec 2010 23:03:17 +0000 (00:03 +0100)]
ALSA: pcm: remember to always call va_end() on stuff that we va_start()
The Coverity checker spotted that we do not always remember to call
va_end() on 'args' in failure paths in snd_pcm_hw_rule_add().
Here's a patch to fix that up (compile tested only) - it also removes
some annoying trailing whitespace that caught my eye while I was in the
area..
Takashi Iwai [Fri, 17 Dec 2010 14:23:41 +0000 (15:23 +0100)]
ALSA: hda - Fix conflict of Mic Boot controls
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.
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>
Jarkko Nikula [Fri, 10 Dec 2010 18:53:55 +0000 (20:53 +0200)]
ASoC: Fix bias power down of non-DAPM codec
Currently bias of non-DAPM codec will be powered down (standby/off) whenever
there is a stream stop. This is wrong in simultaneous playback/capture since
the bias is put down immediately after stopping the first stream.
Fix this by using the codec->active count when figuring out the needed bias
level after stream stop.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
ASoC: fix deemphasis control in wm8904/55/60 codecs
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> Cc: stable@kernel.org
ALSA: HDA: Remove unconnected PCM devices for Intel HDMI
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>
Anssi Hannula [Tue, 7 Dec 2010 19:19:23 +0000 (21:19 +0200)]
ALSA: hda - Reset sample sizes and max bitrates when reading ELD
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.
Anssi Hannula [Tue, 7 Dec 2010 18:56:19 +0000 (20:56 +0200)]
ALSA: hda - Always allow basic audio irrespective of ELD info
Commit bbbe33900d1f3c 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.
Anssi Hannula [Tue, 7 Dec 2010 16:41:35 +0000 (18:41 +0200)]
ALSA: hda - Do not wrongly restrict min_channels based on ELD
Commit bbbe33900d1f3c 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.
Jassi Brar [Tue, 7 Dec 2010 10:23:07 +0000 (19:23 +0900)]
ASoC: WM8580: Debug BCLK and sample size
In case of SNDRV_PCM_FORMAT_S32_LE, we need to set WM8580_AIF_LENGTH_32,
rather than WM8580_AIF_LENGTH_24.
Also, the BCLK has to be 64fs, for sample size of 20, 24 and 32 bits.
Signed-off-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Tue, 7 Dec 2010 12:56:30 +0000 (20:56 +0800)]
ASoC: Fix resource leak if soc_register_ac97_dai_link failed
Properly free the resources in the case of soc_register_ac97_dai_link failure.
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>
Uk Kim [Sun, 5 Dec 2010 08:26:07 +0000 (17:26 +0900)]
ASoC: Fix swap of left and right channels for WM8993/4 speaker boost gain
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> Cc: stable@kernel.org
Uk Kim [Sun, 5 Dec 2010 08:32:16 +0000 (17:32 +0900)]
ASoC: Fix off by one error in WM8994 EQ register bank size
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> Cc: stable@kernel.org
Daniel T Chen [Sun, 5 Dec 2010 13:43:14 +0000 (08:43 -0500)]
ALSA: hda: Use position_fix=1 for Acer Aspire 5538 to enable capture on internal mic
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>
Anssi Hannula [Sun, 5 Dec 2010 00:34:15 +0000 (02:34 +0200)]
ALSA: hda - use generic hdmi parser for ATI R6xx codec
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.
Takashi Iwai [Fri, 3 Dec 2010 11:29:14 +0000 (12:29 +0100)]
ALSA: hda - Clean up cxt5066 port-D handling & co
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.
John Baboval [Thu, 2 Dec 2010 16:21:31 +0000 (11:21 -0500)]
ALSA: hda - Fix ThinkPad T410[s] docking station line-out
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>
Daniel T Chen [Fri, 3 Dec 2010 03:45:45 +0000 (22:45 -0500)]
ALSA: hda: Use model=lg quirk for LG P1 Express to enable playback and capture
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>
Jarkko Nikula [Wed, 1 Dec 2010 09:01:20 +0000 (11:01 +0200)]
ASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency
Not all omap boards use kernel based pin multiplexing so
CONFIG_SND_OMAP_SOC_N810 should not select it by default as it can make
harm to other boards in multi-board kernels.
Therefore put CONFIG_OMAP_MUX as a dependency to N810 ASoC machine driver.
Thanks to Tony Lindgren <tony@atomide.com> for noticing.
Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Cc: Tony Lindgren <tony@atomide.com> Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Tony Lindgren <tony@atomide.com> Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Daniel T Chen [Thu, 2 Dec 2010 00:16:07 +0000 (19:16 -0500)]
ALSA: hda: Use "alienware" model quirk for another SSID
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>
Daniel Glöckner [Tue, 30 Nov 2010 00:00:18 +0000 (01:00 +0100)]
s6105-ipcam: fix compilation
When the s6105-ipcam ASoC driver had been converted to the
multi-component API, a single reference to a former structure
element remained, blocking successful compilation.
Signed-off-by: Daniel Glöckner <daniel-gl@gmx.net> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel Glöckner [Tue, 30 Nov 2010 00:00:17 +0000 (01:00 +0100)]
s6000-pcm: fix compilation
s6000_soc_platform has lost its forward declaration and there no
longer is a name element in it, so use a string constant when
calling request_irq.
Signed-off-by: Daniel Glöckner <daniel-gl@gmx.net> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Daniel Glöckner [Tue, 30 Nov 2010 00:00:16 +0000 (01:00 +0100)]
s6000-i2s: fix compilation
A semicolon was missing.
Signed-off-by: Daniel Glöckner <daniel-gl@gmx.net> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Mon, 29 Nov 2010 09:42:47 +0000 (17:42 +0800)]
ASoC: Fix missing spin_unlock_irqrestore
In nuc900_dma_hw_params(), if snd_pcm_lib_malloc_pages failed
it returns without calling spin_unlock_irqrestore().
Since snd_pcm_lib_malloc_pages() does not touch struct nuc900_audio,
we don't need to hold the lock while calling snd_pcm_lib_malloc_pages().
Fix it by moving spin_lock_irqsave() down to after snd_pcm_lib_malloc_pages().
In nuc900_dma_prepare(), spin_unlock_irqrestore() is missing in the error path.
Fix it by removing the return in default case.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Wan ZongShun <mcuos.com@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Takashi Iwai [Tue, 30 Nov 2010 07:14:21 +0000 (08:14 +0100)]
ALSA: Fix SNDCTL_DSP_RESET ioctl for OSS emulation
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> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
Axel Lin [Mon, 29 Nov 2010 06:55:58 +0000 (14:55 +0800)]
ASoC: Add missing dev_set_drvdata in p1022_ds_probe
Otherwise, calling dev_get_drvdata in p1022_ds_remove returns NULL.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Mon, 29 Nov 2010 06:54:58 +0000 (14:54 +0800)]
ASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe
Otherwise, calling dev_get_drvdata in mpc8610_hpcd_remove returns NULL.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Timur Tabi <timur@freescale.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Mon, 29 Nov 2010 09:43:39 +0000 (17:43 +0800)]
ASoC: Remove unneeded !! operations while checking return value of nuc900_checkready
I think this unneededd !! operations just reduce the readability.
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>
Axel Lin [Mon, 29 Nov 2010 09:40:53 +0000 (17:40 +0800)]
ASoC: Fix compile error for nuc900-pcm.c
This patch fixes below error:
CC sound/soc/nuc900/nuc900-pcm.o
sound/soc/nuc900/nuc900-pcm.c: In function 'nuc900_dma_open':
sound/soc/nuc900/nuc900-pcm.c:267: error: 'nuc900_ac97_data' undeclared (first use in this function)
sound/soc/nuc900/nuc900-pcm.c:267: error: (Each undeclared identifier is reported only once
sound/soc/nuc900/nuc900-pcm.c:267: error: for each function it appears in.)
sound/soc/nuc900/nuc900-pcm.c: At top level:
sound/soc/nuc900/nuc900-pcm.c:337: error: expected ',' or ';' before 'static'
sound/soc/nuc900/nuc900-pcm.c:354: error: 'nuc900_soc_platform_probe' undeclared here (not in a function)
make[3]: *** [sound/soc/nuc900/nuc900-pcm.o] Error 1
make[2]: *** [sound/soc/nuc900] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
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>
Axel Lin [Mon, 29 Nov 2010 09:40:05 +0000 (17:40 +0800)]
ASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove
This patch fixes below compile warning:
CC sound/soc/nuc900/nuc900-ac97.o
sound/soc/nuc900/nuc900-ac97.c:300: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:301: warning: initialization from incompatible pointer type
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>
Axel Lin [Mon, 29 Nov 2010 09:39:10 +0000 (17:39 +0800)]
ASoC: Fix compile error for nuc900-ac97.c
Fix below compile error by add a missing ';'.
CC sound/soc/nuc900/nuc900-ac97.o
sound/soc/nuc900/nuc900-ac97.c:300: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:301: warning: initialization from incompatible pointer type
sound/soc/nuc900/nuc900-ac97.c:318: error: expected ',' or ';' before 'static'
sound/soc/nuc900/nuc900-ac97.c:405: error: 'nuc900_ac97_drvprobe' undeclared here (not in a function)
make[3]: *** [sound/soc/nuc900/nuc900-ac97.o] Error 1
make[2]: *** [sound/soc/nuc900] Error 2
make[1]: *** [sound/soc] Error 2
make: *** [sound] Error 2
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>
Daniel T Chen [Sat, 27 Nov 2010 18:58:04 +0000 (13:58 -0500)]
ALSA: hda: Use BIOS auto-parsing instead of existing model quirk for MEDION MD2
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>
Takashi Iwai [Fri, 26 Nov 2010 16:11:18 +0000 (17:11 +0100)]
ALSA: hda - Use ALC_INIT_DEFAULT for really default initialization
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.
Axel Lin [Wed, 24 Nov 2010 14:24:01 +0000 (22:24 +0800)]
ASoC: Fix resource reclaim for osk5912
In current implementation, there are resources leak in the error path.
This patch properly reclaims the allocated resources in the error path.
Also adds a missing clk_put in osk_soc_exit.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Wed, 24 Nov 2010 14:40:59 +0000 (22:40 +0800)]
ASoC: tlv320aic3x - fix variable may be used uninitialized warning
If aic3x_read failed , val is used uninitialized.
Fix it by initializing val to 0.
This patch fixes below compile warning:
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_get_gpio':
sound/soc/codecs/tlv320aic3x.c:1183: warning: 'val' may be used uninitialized in this function
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_headset_detected':
sound/soc/codecs/tlv320aic3x.c:1211: warning: 'val' may be used uninitialized in this function
sound/soc/codecs/tlv320aic3x.c: In function 'aic3x_button_pressed':
sound/soc/codecs/tlv320aic3x.c:1219: warning: 'val' may be used uninitialized in this function
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>
Axel Lin [Thu, 25 Nov 2010 03:33:14 +0000 (11:33 +0800)]
ASoC: davinci-vcif - fix a memory leak
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>
Axel Lin [Thu, 25 Nov 2010 07:14:03 +0000 (15:14 +0800)]
ASoC: phycore-ac97: fix resource leak
Fix imx_phycore_init() error path and imx_phycore_exit() to properly free
allocated resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Thu, 25 Nov 2010 07:13:09 +0000 (15:13 +0800)]
ASoC: imx-ssi: fix resource leak
Fix imx_ssi_probe() error path and imx_ssi_remove() to properly free
allocated resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Axel Lin [Thu, 25 Nov 2010 07:12:30 +0000 (15:12 +0800)]
ASoC: simone: fix resource leak in simone_init error path
Fix the error path to properly free allocated resources.
Signed-off-by: Axel Lin <axel.lin@gmail.com> Acked-by: Mika Westerberg <mika.westerberg@iki.fi> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
projects / karo-tx-linux.git / log