Commit graph

4517 commits

Author SHA1 Message Date
Clemens Ladisch
84f9df159d ALSA: ymfpci: fix PCM open error handling
The installation of the minimum period size constraint in the PCM open
callbacks was not checked for errors.  Add this check, and move the call
to the beginning of the function to avoid having to do any cleanups in
the error case.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-20 08:56:34 +02:00
Takashi Iwai
0308110615 Merge branch 'fix/misc' into topic/misc 2011-09-16 08:29:04 +02:00
David Henningsson
2e1210bc3d ALSA: HDA: Cirrus - fix "Surround Speaker" volume control name
This patch fixes "Surround Speaker Playback Volume" being cut off.
(Commit b4dabfc452 was probably meant to fix this, but it fixed
only the "Switch" name, not the "Volume" name.)

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 13:45:12 +02:00
Clemens Ladisch
dba8b46992 ALSA: mpu401: clean up interrupt specification
The semantics of snd_mpu401_uart_new()'s interrupt parameters are
somewhat counterintuitive:  To prevent the function from allocating its
own interrupt, either the irq number must be invalid, or the irq_flags
parameter must be zero.  At the same time, the irq parameter being
invalid specifies that the mpu401 code has to work without an interrupt
allocated by the caller.  This implies that, if there is an interrupt
and it is allocated by the caller, the irq parameter must be set to
a valid-looking number which then isn't actually used.

With the removal of IRQF_DISABLED, zero becomes a valid irq_flags value,
which forces us to handle the parameters differently.

This patch introduces a new flag MPU401_INFO_IRQ_HOOK for when the
device interrupt is handled by the caller, and makes the allocation of
the interrupt to depend only on the irq parameter.  As suggested by
Takashi, the irq_flags parameter was dropped because, when used, it had
the constant value IRQF_DISABLED.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-14 11:00:51 +02:00
Takashi Iwai
99e14c9d41 ALSA: hda - Terminate the recursive connection search properly
The recursive search of widget connections in snd_hda_get_conn_index()
must be terminated at the pin and the audio-out widgets.  Otherwise
you'll get "too deep connection" warnings unnecessarily.

Reported-by: Francis Moreau <francis.moro@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-13 10:33:16 +02:00
Raymond Yau
89f3325a6e ALSA: ymfpci: add "Playback" to FM Legacy Volume control
YDSXGR_LEGACYOUTVOL is a Playback Volume control for OPL3 FM Synth.

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-09-12 10:30:28 +02:00
David Henningsson
468c545885 ALSA: hda: Conexant: Allow different output types to share DAC
Headphones has stopped working for the original reported (a regression
compared to 2.6.38). This is because Speaker and Headphones share the
same DAC, in which case no Headphones volume control was created.
This patch fixes so that both Speaker and Headphones volume
controls are created in such scenario.

BugLink: http://bugs.launchpad.net/bugs/817943
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-25 15:08:03 +02:00
Takashi Iwai
7675535958 ALSA: hda/conexant - Enable ADC-switching for auto-mic mode, too
The ADC-switching can work also in the auto-mic mode, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-24 10:53:10 +02:00
Maarten Lankhorst
391e69143d ALSA: ctxfi: Bump playback substreams to 256
There are references in the code to 256 sources, so I tested it with 256 aplays,
of which the first and last with real data and the rest playing /dev/zero .

Also increase amount of page tables, so the default aplay size works.

Signed-off-by: Maarten Lankhorst <m.b.lankhorst@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-24 10:39:44 +02:00
Takashi Iwai
1f015f5fdc ALSA: hda - Fix double-headphone/speaker paths for Cxt auto-parser
When multiple headphones or speakers are assigned but no individual
DACs are available, the driver should take the first HP/SPK DAC instead
of another primary output.  The patch adds a bit-flag to dac field of
struct pin_dac_pair indicating that it's a slave DAC.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-23 14:57:08 +02:00
Takashi Iwai
3c715a9884 ALSA: hda - Update jack-sense info even when no automute is set
The internal states, jack_present and line_jack_present should be
updated upon unsolicited events even if no automute is set.
Otherwise the wrong state is referred when the automute behavior is
changed by the mixer control.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-23 12:41:09 +02:00
Takashi Iwai
675c1aa3c4 ALSA: hda - Fix output-path initialization for Realtek auto-parser
When the headphone or speaker output has no own DAC, initialize the path
using the primary DAC.  Otherwise the path won't be set properly and
can result in the silence.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-23 12:36:28 +02:00
Takashi Iwai
1b004d03d8 ALSA: hda - Fix error check from snd_hda_get_conn_index() in patch_cirrus.c
snd_hda_get_conn_index() returns a negative value while the current code
stores it in an unsigned int.  It must be stored in a signed integer.

Reported-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-20 09:24:54 +02:00
Takashi Iwai
b6acf013bd ALSA: hda - Don't spew too many ELD errors
Currently HD-audio driver shows the all error ELD byte as an error
in the kernel message.  This is annoying when the video driver doesn't
set the correct ELD from the beginning. e.g. radeon sends a zero-byte
data, but we still check ELD with the fixed 128 byte as a workaround
for some broken devices, it spews 128-times errors.

For avoiding this, the driver aborts reading when the first byte is
invalid.  In such a case, the whole data is certainly invalid.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-20 09:23:10 +02:00
Takashi Iwai
b55ac2a116 Merge branch 'fix/misc' into topic/misc 2011-08-19 08:30:38 +02:00
Takashi Iwai
3fe45aeaf2 ALSA: hda - Add "PCM" volume to vmaster slave list
The new parser may use "PCM" volume, but it was missing the vmaster
slave list, thus "Master" volume didn't control it.

Reference: https://bugzilla.kernel.org/show_bug.cgi?id=41342

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-18 15:13:17 +02:00
Takashi Iwai
c503ad466d ALSA: hda - Fix duplicated capture-volume creation for ALC268 models
Fix the duplicated creation of capture-mixer elements for some static
ALC268 configurations.  The capture mixers must be put to cap_mixer field
instead of mixers array.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-16 14:23:20 +02:00
Takashi Iwai
d877681d2e ALSA: hdspm - Simplify with snd_pcm_hw_constraint_pow2()
Refactoring the code using snd_pcm_hw_constraint_pow2() helper function.

Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-15 13:37:03 +02:00
Takashi Iwai
3fa9e3d230 ALSA: hdspm - Add missing KNOT flag for AES32 rate restriction
AES32 supports the non-standard 128kHZ, and this is enabled only when
SNDRV_PCM_RATE_KNOT is set in hw.rates field.

Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-15 13:36:49 +02:00
Takashi Iwai
52e6fb4812 ALSA: hdspm - Correct max buffer size limit
Some modesl can support up to 8192 frames per period.

Tested-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-15 13:36:18 +02:00
Clemens Ladisch
dc3fcd1655 ALSA: virtuoso: fix Essence ST(X) S/PDIF input
On the Xonar Essence ST/STX, the connector J14 has been confirmed to be
a digital input, so enable it in the driver.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-15 13:35:14 +02:00
Daniel T Chen
eade7b281c ALSA: ac97: Add HP Compaq dc5100 SFF(PT003AW) to Headphone Jack Sense whitelist
BugLink: https://bugs.launchpad.net/bugs/826081

The original reporter needs 'Headphone Jack Sense' enabled to have
audible audio, so add his PCI SSID to the whitelist.

Reported-and-tested-by: Muhammad Khurram Khan
Cc: <stable@kernel.org>
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-15 10:26:37 +02:00
Adrian Knoth
2e61027079 ALSA: hdspm - Enable 32 samples/period on RME RayDAT/AIO
Newer RME cards like RayDAT and AIO support 32 samples per period. This
value is encoded as {1,1,1} in the HDSP_LatencyMask bits in the control
register.

Since {1,1,1} is also the representation for 8192 samples/period on
older RME cards, we have to special case 32 samples and 32768 bytes
according to the actual card.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-15 10:25:39 +02:00
Adrian Knoth
7cb155ff3e ALSA: hdspm - Introduce hdspm_get_latency() to harmonize latency calculation
Currently, hdspm_decode_latency is called several times, violating the
DRY principle. Given that we need to distinguish between old and new
cards when decoding the latency bits in the control register, introduce
hdspm_get_latency() to provide the required functionality.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-15 10:25:32 +02:00
Adrian Knoth
1ad5972f71 ALSA: hdspm - Reorder period sizes according to their bit representation
On newer RME cards like RayDAT and AIO, the 8192 samples per period size
are no longer supported. Instead, setting all three bits of
HDSP_LatencyMask to one ({1,1,1}) now corresponds to 32 samples per
period.

To make this more obvious to future developers, let's reorder the array
according to their bit representation, starting at 64 ({0,0,0}) up to
4096 ({1,1,0}) and finally 32 ({1,1,1}).

Note that this patch doesn't change semantics.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-15 10:25:23 +02:00
Adrian Knoth
1b6fa108b3 ALSA: hdspm - Set period_bytes_min to 32 * 4 for new RME cards
On newer RME cards like RayDAT and AIO, the lower bound is 32 samples
per period in contrast to 64 samples as seen on older cards.

We hence lower period_bytes_min to 32 * 4. Four bytes per sample.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-15 10:24:42 +02:00
Adrian Knoth
135d1535f4 ALSA: hdspm - Allow for 8192 period size on RME MADI and AES cards
Older RME cards like MADI and AES support period sizes of 8192 samples.
The original hdspm driver already featured this value, apparently, it
was lost during the rewrite.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-15 10:24:35 +02:00
Julia Lawall
a5a3973da8 ALSA: azt3328 - adjust error handling code to include debugging code
snd_azf3328_dbgcallenter is called at the very beginning of the function,
so it could be useful to call snd_azf3328_dbgcallleave at all exit points.

Signed-off-by: Julia Lawall <julia@diku.dk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-10 11:52:23 +02:00
Wang Shaoyan
96b6359779 ALSA: hda - Add CONFIG_SND_HDA_POWER_SAVE to stac_vrefout_set()
In commit 45eebda7, it add new function stac_vrefout_set, but it
is only used in code between CONFIG_SND_HDA_POWER_SAVE macro, so
add the macro to avoid such warning:

  sound/pci/hda/patch_sigmatel.c:676:12: warning: 'stac_vrefout_set' defined but not used

Signed-off-by: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-10 11:22:08 +02:00
Takashi Iwai
0a2d31b62d Merge branch 'fix/kconfig' into for-linus 2011-08-08 14:30:29 +02:00
Takashi Iwai
df944f6678 ALSA: Fix dependency of CONFIG_SND_TEA575X
CONFIG_SND_TEA575X is enabled by RADIO_SF16FMR2, but the latter one is
no PCI device.  Since tea575x-tuner itself is independent from the board
bus type, the config should be moved out of SND_PCI dependency.

Reported-by: Randy Dunlap <rdunlap@xenotime.net>
Acked-by: Randy Dunlap <rdunlap@xenotime.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-07 17:54:17 +02:00
Thomas Meyer
67ada8367c ALSA: asihpi - use kzalloc()
Use kzalloc rather than kmalloc followed by memset with 0

 This considers some simple cases that are common and easy to validate
 Note in particular that there are no ...s in the rule, so all of the
 matched code has to be contiguous

 The semantic patch that makes this output is available
 in scripts/coccinelle/api/alloc/kzalloc-simple.cocci.

 More information about semantic patching is available at
 http://coccinelle.lip6.fr/

Signed-off-by: Thomas Meyer <thomas@m3y3r.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-07 17:32:52 +02:00
Wang Shaoyan
81c0a78b64 ALSA: hda - Fix a complile warning in patch_via.c
sound/pci/hda/patch_via.c:2087: warning: 'dac' may be used uninitialized in this function

Signed-off-by: Wang Shaoyan <wangshaoyan.pt@taobao.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-05 12:51:01 +02:00
Takashi Iwai
3d56c8e6b0 ALSA: hdspm - Fix uninitialized compile warnings
Put the exception checks for io_type switch() for possible mistakes in
future.  Also this shuts up annoying compile warnings.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-05 12:30:12 +02:00
Takashi Iwai
c3540b81ee ALSA: hda - Use auto-parser for ASUS UX50, Eee PC P901, S101 and P1005
It works fine with auto-parser and now the digital mic workaround was
implemented in auto-parser fixup, let's drop the static model quirks for
these models.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-04 15:21:19 +02:00
Takashi Iwai
adabb3ec8b ALSA: hda - Fix digital-mic mono recording on ASUS Eee PC
The digital-mic unit on ASUS Eee PC gives PDM signals instead of the
normal stereo PCM, thus you can't record a mono stream from the stereo
stream as is; the summed stereo signal results in almost zero level, and
you'll hear only soft noise.

As a workaround, use ALC269-specific COEF to manipulate the dmic route
for mono, like used for ALC271x.  This is implemented as a fix-up, thus
it works only with model=auto or without REALTEK_QUIRKS Kconfig.

Reported-and-tested-by: Pavel Roskin <proski@gnu.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-04 15:21:13 +02:00
Linus Torvalds
0d7e92da50 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: asihpi - Clarify adapter index validity check
  ALSA: asihpi - Don't leak firmware if mem alloc fails
  ALSA: rtctimer.c needs module.h
  ASoC: Fix txx9aclc.c build
  ALSA: hdspm - Add firmware revision 0xcc for RME MADI
  ALSA: hdspm - Fix reported external sample rate on RME MADI and MADIface
  ALSA: hdspm - Provide MADI speed mode selector on RME MADI and MADIface
  ALSA: sound/core/pcm_compat.c: adjust array index
2011-08-02 20:48:22 -10:00
Eliot Blennerhassett
08f984c7f7 ALSA: asihpi - Clarify adapter index validity check
Avoids assigning possibly invalid address to pa, even if it
is never dereferenced.
Correct error response to reflect request object/function ids.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-02 09:26:31 +02:00
Jesper Juhl
dc889f1864 ALSA: asihpi - Don't leak firmware if mem alloc fails
We leak the memory allocated to 'firmware' when we fail to
release_firmware() after a kmalloc() failure in hpi_dsp_code_open().
This patch should take care of the leak.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-08-01 10:57:06 +02:00
Linus Torvalds
664a41b8a9 Merge branch 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6
* 'v4l_for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mchehab/linux-2.6: (430 commits)
  [media] ir-mce_kbd-decoder: include module.h for its facilities
  [media] ov5642: include module.h for its facilities
  [media] em28xx: Fix DVB-C maxsize for em2884
  [media] tda18271c2dd: Fix saw filter configuration for DVB-C @6MHz
  [media] v4l: mt9v032: Fix Bayer pattern
  [media] V4L: mt9m111: rewrite set_pixfmt
  [media] V4L: mt9m111: fix missing return value check mt9m111_reg_clear
  [media] V4L: initial driver for ov5642 CMOS sensor
  [media] V4L: sh_mobile_ceu_camera: fix Oops when USERPTR mapping fails
  [media] V4L: soc-camera: remove soc-camera bus and devices on it
  [media] V4L: soc-camera: un-export the soc-camera bus
  [media] V4L: sh_mobile_csi2: switch away from using the soc-camera bus notifier
  [media] V4L: add media bus configuration subdev operations
  [media] V4L: soc-camera: group struct field initialisations together
  [media] V4L: soc-camera: remove now unused soc-camera specific PM hooks
  [media] V4L: pxa-camera: switch to using standard PM hooks
  [media] NetUP Dual DVB-T/C CI RF: force card hardware revision by module param
  [media] Don't OOPS if videobuf_dvb_get_frontend return NULL
  [media] NetUP Dual DVB-T/C CI RF: load firmware according card revision
  [media] omap3isp: Support configurable HS/VS polarities
  ...

Fix up conflicts:
 - arch/arm/mach-omap2/board-rx51-peripherals.c:
     cleanup regulator supply definitions in mach-omap2
   vs
     OMAP3: RX-51: define vdds_csib regulator supply
 - drivers/staging/tm6000/tm6000-alsa.c (trivial)
2011-07-30 00:08:53 -07:00
Adrian Knoth
5f8b4d53d7 ALSA: hdspm - Add firmware revision 0xcc for RME MADI
Apparently, there are multiple old firmware revisions in the wild for
the PCI RME MADI cards. Just add them to the list of supported devices
and treat them like their modern counterparts.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-29 07:41:30 +02:00
Adrian Knoth
d12c51d829 ALSA: hdspm - Fix reported external sample rate on RME MADI and MADIface
In slave mode, the card can only detect the base frequency (32..48kHz)
on the MADI link (exception: 96k frames), so the real external sample
rate is this base frequency multiplied by 1, 2 or 4 depending on the
speed mode.

This patch enables 64..192kHz sample rates in clock slave mode, which
failed before due to an alleged sample rate mismatch between the MADI
link (e.g., 48kHz) and the application in DS/QS mode (e.g., 96kHz,
192kHz).

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-29 07:41:04 +02:00
Adrian Knoth
700d1ef33f ALSA: hdspm - Provide MADI speed mode selector on RME MADI and MADIface
When running in slave mode (no clock master), there is no way to
determine the real wirespeed on the MADI link (single/double/quad
speed). Like physical gear, simply provide the user with a tristate
switch to select the appropriate format.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-29 07:40:51 +02:00
Linus Torvalds
6f56c21866 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:
  sound: oss: rename local change_bits to avoid powerpc bitsops.h definition
  ALSA: hda - Fix duplicated DAC assignments for Realtek
  ALSA: asihpi - off by one in asihpi_hpi_ioctl()
  ALSA: hda - Fix Oops with Realtek quirks with NULL adc_nids
  ALSA: asihpi - bug fix pa use before init.
  ALSA: hda - Add support for vref-out based mute LED control on IDT codecs
2011-07-28 05:49:31 -07:00
Ondrej Zary
9e8fa0e644 [media] radio-sf16fmr2: convert to generic TEA575x interface
Convert radio-sf16fmr2 to use generic TEA575x implementation. Most of the
driver code goes away as SF16-FMR2 is basically just a TEA5757 tuner
connected to ISA bus.
The card can optionally be equipped with PT2254A volume control (equivalent
of TC9154AP) - the volume setting is completely reworked (with balance control
added) and tested.

Signed-off-by: Ondrej Zary <linux@rainbow-software.org>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mauro Carvalho Chehab <mchehab@redhat.com>
2011-07-27 17:52:23 -03:00
Linus Torvalds
7562343716 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: (22 commits)
  ALSA: hda - Cirrus Logic CS421x support
  ALSA: Make pcm.h self-contained
  ALSA: hda - Allow codec-specific set_power_state ops
  ALSA: hda - Add post_suspend patch ops
  ALSA: hda - Make CONFIG_SND_HDA_POWER_SAVE depending on CONFIG_PM
  ALSA: hda - Make sure mute led reflects master mute state
  ALSA: hda - Fix invalid mute led state on resume of IDT codecs
  ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver"
  ALSA: hda - Add support of the 4 internal speakers on certain HP laptops
  ALSA: Make snd_pcm_debug_name usable outside pcm_lib
  ALSA: hda - Fix DAC filling for multi-connection pins in Realtek parser
  ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.
  ASoC: SAMSUNG: Add I2S0 internal dma driver
  ASoC: SAMSUNG: Modify I2S driver to support idma
  ASoC: davinci: add missing break statement
  ASoC: davinci: fix codec start and stop functions
  ASoC: dapm - add DAPM macro for external enum widgets
  ASoC: Acknowledge WM8962 interrupts before acting on them
  ASoC: sgtl5000: guide user when regulator support is needed
  ASoC: sgtl5000: refactor registering internal ldo
  ...
2011-07-27 09:25:15 -07:00
Takashi Iwai
c48a8fb0d3 ALSA: hda - Fix duplicated DAC assignments for Realtek
Copying hp_pins and speaker_pins from line_out_pins may confuse the
parser, and it can lead to duplicated initializations for the same pin
with a wrong DAC assignment.  The problem appears in 3.0 kernel code.

Cc: <stable@kernel.org> (for 3.0)
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-27 16:44:40 +02:00
Dan Carpenter
ae6ff61e43 ALSA: asihpi - off by one in asihpi_hpi_ioctl()
"adapter" is used as an array index in the adapters[] array so
the off by one would make us read past the end.

1c073b6797 "ALSA: asihpi - Remove spurious adapter index check"
reverted Dan Rosenberg's check that would have prevented the
overflow here.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-27 16:06:16 +02:00
Takashi Iwai
60a6a8425a ALSA: hda - Fix Oops with Realtek quirks with NULL adc_nids
Somce quirk models don't set adc_nids but let the parser filling it.
But the recent code has unnecessary NULL-checks of spec->input_mux,
and it resulted in NULL dereferences.
This patch fixes that regression.

Reported-and-tested-by: Oliver Neukum <oneukum@suse.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-27 16:05:30 +02:00
Eliot Blennerhassett
767cd365b2 ALSA: asihpi - bug fix pa use before init.
Fixes bug introduced by 1c073b67.
Also declare pa local to block in which it is used.

Signed-off-by: Eliot Blennerhassett <eblennerhassett@audioscience.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-07-27 10:08:26 +02:00