Commit Graph

8480 Commits

Author SHA1 Message Date
Jiri Kosina 4b7bd36470 Merge branch 'master' into for-next
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outdated.
2010-12-22 18:57:02 +01:00
Linus Torvalds 74280817e5 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: hda - Fix conflict of Mic Boot controls
  ALSA: HDA: Enable subwoofer on Asus G73Jw
  ALSA: HDA: Fix auto-mute on Lenovo Edge 14
  ASoC: Fix bias power down of non-DAPM codec
  ASoC: WM8580: Fix R8 initial value
  ASoC: fix deemphasis control in wm8904/55/60 codecs
2010-12-17 09:27:30 -08:00
Takashi Iwai 5aad6c5f77 Merge branch 'fix/asoc' into for-linus 2010-12-17 15:28:37 +01:00
Takashi Iwai 8cd1fd2526 Merge branch 'fix/hda' into for-linus 2010-12-17 15:28:33 +01:00
Takashi Iwai 53e8c3239b 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.

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>
2010-12-17 15:23:41 +01:00
David Henningsson ac61240793 ALSA: HDA: Enable subwoofer on Asus G73Jw
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>
2010-12-15 09:45:36 +01:00
David Henningsson fe67b24010 ALSA: HDA: Fix auto-mute on Lenovo Edge 14
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>
2010-12-15 08:17:30 +01:00
Linus Torvalds f9ae3e125c Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ALSA: HDA: Quirk for Dell Vostro 320 to make microphone work
  ALSA: hda - Reset sample sizes and max bitrates when reading ELD
  ALSA: hda - Always allow basic audio irrespective of ELD info
  ALSA: hda - Do not wrongly restrict min_channels based on ELD
  ASoC: Correct WM8962 interrupt mask register read
  ASoC: WM8580: Debug BCLK and sample size
  ASoC: Fix resource leak if soc_register_ac97_dai_link failed
  ASoC: Hold client_mutex while calling snd_soc_instantiate_cards()
  ASoC: Fix swap of left and right channels for WM8993/4 speaker boost gain
  ASoC: Fix off by one error in WM8994 EQ register bank size
  ALSA: hda: Use position_fix=1 for Acer Aspire 5538 to enable capture on internal mic
  ALSA: hda - Enable jack sense for Thinkpad Edge 13
  ALSA: hda - Fix ThinkPad T410[s] docking station line-out
  ALSA: hda: Use model=lg quirk for LG P1 Express to enable playback and capture
2010-12-14 13:32:40 -08:00
Jarkko Nikula 862af8adbe 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>
2010-12-13 16:47:48 +00:00
Seungwhan Youn a096862809 ASoC: WM8580: Fix R8 initial value
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>
Cc: stable@kernel.org
2010-12-09 10:55:56 +00:00
Dmitry Artamonow 3f343f8512 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
2010-12-09 10:55:37 +00:00
Takashi Iwai d70ab7f7ee Merge branch 'fix/asoc' into for-linus 2010-12-09 08:24:32 +01:00
Takashi Iwai 58936b29c4 Merge branch 'fix/hda' into for-linus 2010-12-09 08:24:25 +01:00
David Henningsson 8a96b1e020 ALSA: HDA: Quirk for Dell Vostro 320 to make microphone work
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>
2010-12-09 08:23:31 +01:00
Anssi Hannula 0bbaee3a58 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.

Signed-off-by: Anssi Hannula <anssi.hannula@iki.fi>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-12-08 08:36:20 +01:00
Anssi Hannula 3dc8642903 ALSA: hda - Always allow basic audio irrespective of ELD info
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>
2010-12-07 20:13:22 +01:00
Anssi Hannula 4b0dbdb17f ALSA: hda - Do not wrongly restrict min_channels based on ELD
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>
2010-12-07 20:12:58 +01:00
Mark Brown 2a7b1a0020 ASoC: Correct WM8962 interrupt mask register read
Fix mismerge from the out of tree BSP where this support was developed.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-07 15:49:42 +00:00
Jassi Brar 6b464321d2 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>
2010-12-07 15:49:18 +00:00
Axel Lin 681e369247 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>
2010-12-07 14:51:15 +00:00
Axel Lin 1dcb4f38e5 ASoC: Hold client_mutex while calling snd_soc_instantiate_cards()
As the comments of snd_soc_instantiate_cards() said,
snd_soc_instantiate_cards() must be called with client_mutex.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-12-06 12:53:43 +00:00
Uk Kim ed8cc471d7 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
2010-12-06 12:43:13 +00:00
Daniel T Chen dd5a089edf 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>
2010-12-06 10:34:09 +01:00
Manoj Iyer ef61d4e6d8 ALSA: hda - Enable jack sense for Thinkpad Edge 13
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>
2010-12-04 08:58:36 +01:00
Linus Torvalds 9cd6315357 Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6:
  ASoC: omap: N810: Don't select CONFIG_OMAP_MUX but make it as dependency
  ALSA: hda: Use "alienware" model quirk for another SSID
  ASoC: WM8731: Fix incorrect mask for bypass path disable
  s6105-ipcam: fix compilation
  s6000-pcm: fix compilation
  s6000-i2s: fix compilation
  ASoC: Fix missing spin_unlock_irqrestore
  ALSA: Fix SNDCTL_DSP_RESET ioctl for OSS emulation
  ASoC: Add missing dev_set_drvdata in p1022_ds_probe
  ASoC: Add missing dev_set_drvdata in mpc8610_hpcd_probe
  ASoC: Remove unneeded !! operations while checking return value of nuc900_checkready
  ASoC: Fix compile error for nuc900-pcm.c
  ASoC: Fix prototype for nuc900_ac97_probe and nuc900_ac97_remove
  ASoC: Fix compile error for nuc900-ac97.c
  ALSA: hda: Use BIOS auto-parsing instead of existing model quirk for MEDION MD2
2010-12-03 09:20:41 -08:00
John Baboval 3a253445e3 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>
2010-12-03 12:27:11 +01:00
Daniel T Chen 77c4d5cdb8 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>
2010-12-03 11:28:01 +01:00
Takashi Iwai 2ff38c9fba Merge branch 'fix/asoc' into for-linus 2010-12-02 17:33:53 +01:00
Takashi Iwai af745bd724 Merge branch 'for-2.6.37' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc 2010-12-02 17:31:18 +01:00
Takashi Iwai c949e3d685 Merge branch 'fix/hda' into for-linus 2010-12-02 17:14:50 +01:00
Jarkko Nikula 0ffd22b694 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>
2010-12-02 11:42:30 +00:00
Daniel T Chen 0defe09ca7 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>
2010-12-02 08:06:00 +01:00
Dimitris Papastamos 2062ea522b ASoC: WM8731: Fix incorrect mask for bypass path disable
According to the datasheet the bypass path enable/disable is
bit 3 therefore we need 0x8 and not 0x4.

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>
2010-12-01 11:26:40 +00:00
Daniel Glöckner b76fb39d49 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>
2010-11-30 11:26:15 +00:00
Daniel Glöckner 9e4ea718d3 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>
2010-11-30 11:26:05 +00:00
Daniel Glöckner b1d36b1c35 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>
2010-11-30 11:25:53 +00:00
Axel Lin 3f1af9d26f 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>
2010-11-30 11:09:24 +00:00
Takashi Iwai 60686aa008 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>
2010-11-30 08:18:58 +01:00
Axel Lin 39a545559f 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>
2010-11-29 18:28:53 +00:00
Axel Lin 67bd489aa3 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>
2010-11-29 18:28:53 +00:00
Axel Lin 3f90e5028a 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>
2010-11-29 18:28:53 +00:00
Axel Lin a7a9820bae 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>
2010-11-29 18:28:53 +00:00
Axel Lin e3edefbd4a 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>
2010-11-29 18:28:53 +00:00
Axel Lin 5a8f1d4701 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>
2010-11-29 18:28:52 +00:00
Daniel T Chen ac70eb1305 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>
2010-11-29 07:41:09 +01:00
Linus Torvalds 0f639a3c5c Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (30 commits)
  ALSA: hda - Use ALC_INIT_DEFAULT for really default initialization
  ASoC: Fix resource reclaim for osk5912
  ASoC: tlv320aic3x - fix variable may be used uninitialized warning
  ASoC: davinci-vcif - fix a memory leak
  ASoC: phycore-ac97: fix resource leak
  ASoC: imx-ssi: fix resource leak
  ASoC: simone: fix resource leak in simone_init error path
  ASoC: sam9g20_wm8731: fix resource leak in at91sam9g20ek_init error path
  ASoC: snd-soc-afeb9260: remove unneeded platform_device_del in error path
  ASoC: pcm030-audio-fabric: fix resource leak in pcm030_fabric_init error path
  ASoC: efika-audio-fabric: fix resource leak in efika_fabric_init error path
  ASoC: Call snd_soc_unregister_dais instead of snd_soc_unregister_dai in sh4_soc_dai_remove
  ASoC: fix SND_PXA2XX_LIB Kconfig warning
  ALSA: hda - Fix ALC660-VD/ALC861-VD capture/playback mixers
  ALSA: HDA: Add an extra DAC for Realtek ALC887-VD
  ASoC: nuc900-ac97: fix a memory leak
  ASoC: Return proper error for omap3pandora_soc_init
  ASoC: wm8961 - clear WM8961_MCLKDIV bit for freq <= 16500000
  ASoC: wm8961 - clear WM8961_DACSLOPE bit for normal mode
  ALSA: hda - Fix Acer 7730G support
  ...
2010-11-27 12:42:15 -08:00
Takashi Iwai 37fa84d8e6 Merge branch 'fix/asoc' into for-linus 2010-11-26 17:17:42 +01:00
Takashi Iwai 5a8cfb4e8a 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.

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>
2010-11-26 17:11:18 +01:00
Axel Lin 25436180ee 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>
2010-11-25 21:15:49 +00:00
Axel Lin fe99b55994 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>
2010-11-25 21:15:35 +00:00