Commit Graph

10320 Commits (4b0109830842fa645c7f7460dc713cedfe4473f6)

Author SHA1 Message Date
Takashi Iwai 8df2a3129d ALSA: hda - Fix re-routing of HP-independent mode in patch_via.c
Re-route the whole output path when HP-independent mode is changed.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 12:20:13 +02:00
Lydia Wang a00a5fad9d ALSA: hda - Fix creations of playback volume controls in patch_via.c
Fix a issue to create playback volume control if pin has amplifier capability
but not DAC.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 10:24:18 +02:00
Takashi Iwai 8e3679dca2 ALSA: hda - Revisit output_path parsing in patch_via.c
Change the order of the output-path list in a way from the DAC to the
target pin.  Also now the list include the target pin, too.

Together with this format change, simplify the arguments of
parse_output_path() function, and fix the initialization in
via_auto_init_output().

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 09:01:36 +02:00
Takashi Iwai 30f7c5d491 ALSA: hda - Use xxx Boost Volume for VIA
Drop "Capture" prefix from the mic-boost names.
Otherwise some control names can overflow the max name length.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 08:37:41 +02:00
Takashi Iwai efb9f469b6 ALSA: hda - Fix a compile error in patch_ca0132.c for the recent SPDIF change
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:44:51 +02:00
Takashi Iwai 216c7a0f22 Merge branch 'fix/hda' into topic/via-cleanup
Conflicts:
	sound/pci/hda/patch_via.c
2011-06-21 07:34:45 +02:00
Ian Minett 95c6e9cb77 ALSA: hda - Add Creative CA0132 HDA codec support
Create patch_ca0132.c, to add support for devices featuring the
Creative CA0132 HD-audio codec.

This driver implements :-
* 1 playback subdevice to headphone and speaker
* 2 capture subdevices:
   i - Mic-in
   ii- Line-in
* mixer device

Advanced DSP features are not yet included.
Developed and maintained by Creative Labs, Inc.

Signed-off-by: Ian Minett <ian_minett@creativelabs.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:31:25 +02:00
Lydia Wang e905a83acd ALSA: VIA HDA: Create a master amplifier control for VT1718S.
Create a master volume and mute control of playback for VT1718S.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:24:56 +02:00
Lydia Wang ba31a60d0f ALSA: VIA HDA: Mute/unmute mixer conncted to Headphone for VT1718S.
When switch HP independent mode, mute/unmute connctions of mixer  which is
connected to headphone for VT1718S.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:23:19 +02:00
Lydia Wang 42467b32ce ALSA: VIA HDA: Modify initial verbs list for VT1718S.
Remove some invalid initial verbs and correct some wrong initial verbs
for VT1718S codec.

Signed-off-by: Lydia Wang <lydiawang@viatech.com.cn>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:22:57 +02:00
Tony Vroon c933790614 ALSA: hda - Remove ALC268 model override for CPR2000
The "diverse" Quanta ID 0x0763 is overridden to ALC268_ACER.
This keeps headphone automute and microphone input from operating
on at least one laptop from Opti Systems.
Without the override, the BIOS parser does a fine job setting the
card up and everything works.

Tested-By: Peter Schneider <e.at.chi.kaen@googlemail.com>
Signed-off-by: Tony Vroon <tony@linx.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-21 07:18:36 +02:00
Dan Carpenter c20974090e ASoC: adau1701: signedness bug in adau1701_write()
"ret" is supposed to be signed here.  The current code will only
return -EIO on error, instead of a more appropriate error code such
as -EAGAIN etc.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Acked-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-20 18:46:33 +01:00
David Henningsson 6f2e810ad5 ALSA: HDA: Remove quirk for an HP device
The reporter, who is running kernel 2.6.38, reports that
he needs to set model=auto for the headphone output to work
correctly.

BugLink: http://bugs.launchpad.net/bugs/761022
Cc: stable@kernel.org (v2.6.38+)
Reported-by: Jo
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:32:50 +02:00
Takashi Iwai ada509ec00 ALSA: hda - Simplify analog-low-current mode check for VIA codecs
Use the existing aa-loop list for simplifying the check for analog
low-current mode.  Also fix the stream count test for playback streams.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:16 +02:00
Takashi Iwai 47be05ce0a ALSA: hda - Remove NID_MAPPING hacks in patch_via.c
There is no longer virtual kmixer element for NID mapping.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:16 +02:00
Takashi Iwai c619160787 ALSA: hda - Remove unused defines and struct fields in patch_via.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:15 +02:00
Takashi Iwai 6aadf41d6b ALSA: hda - Name the primary out as Speaker when needed for VIA codecs
When the primary output is the speaker output, rather name it as
"Speaker".  This will be more intuitive.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:14 +02:00
Takashi Iwai 13af8e77ea ALSA: hda - Create loopback-list dynamically in patch_via.c
Create loopback list dynamically from the parsed input pins for VIA
codecs instead of the fixed arrays.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:13 +02:00
Takashi Iwai e3d7a1431f ALSA: hda - Fix smart51 handling again
Fix the broken detection of smart51 and its handling.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:12 +02:00
Takashi Iwai 370bafbdae ALSA: hda - Create virtual-master control for VIA codecs
Now let's add the missing Master control to VIA codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:12 +02:00
Takashi Iwai 4a918ffeaa ALSA: hda - Initialize unsol events dynamically in patch_via.c
Issue the init verbs of unsolicited events dynamically from the parsed
results for VIA codecs.  Also, consolidate the unsol handlers for HP
and line-out mutes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:11 +02:00
Takashi Iwai 096a885494 ALSA: hda - Initialize input-path dynamically in patch_via.c
Similarly like the previous commit, initialize the input-paths dynamically
from the parsed results instead of the fixed array for VIA codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:10 +02:00
Takashi Iwai 5d41762a21 ALSA: hda - Initialize output path dynamically in patch_via.c
Instead of fixed array for each codec type, initialize the output path
dynamically from the parsed results.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:09 +02:00
Takashi Iwai 0fe0adf82f ALSA: hda - Replace with standard consts in patch_via.c
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:09 +02:00
Takashi Iwai ece8d0431f ALSA: hda - Fix independent-HP handling in patch_via.c
Fix races in handling of HP DAC and independent streams for VIA codecs.
Also, allow the HP output path without front-DAC, and removed
unnecessary activation of HP mixer elements.

This also removes the handling of shared side/HP stream; it's anyway
implemented in a broken way, so we need to re-implement the feature
later...

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:08 +02:00
Takashi Iwai 12daef65fd ALSA: hda - Unify auto-parser in patch_via.c
Now all codecs use the same parser-path, so we can reduce into a single
auto-parser function.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:07 +02:00
Takashi Iwai 7f0df88ce0 ALSA: hda - Return error for invalid setup for VIA
Instead of ignoring the invalid pin configuration, return the error.
This will avoid unexpected crash, anyway.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:06 +02:00
Takashi Iwai d7a99cce55 ALSA: hda - Unify capture-mixer creations in patch_via.c
Create capture-related mixer elements dynamically from the parsed
ADCs and input-pins instead of fixed values for each codec.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:05 +02:00
Takashi Iwai 7eb56e84e6 ALSA: hda - Assign HP-independent PCM to individual stream
Instead of using the secondary substream, create an individual PCM
stream for HP-independent PCM.  Otherwise it's difficult to handle
different channel numbers with multi-channel stream in the sam PCM
stream structure.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:05 +02:00
Takashi Iwai 9af7421091 ALSA: hda - Unify PCM assignments in patch_via.c
Unify PCM streams for all codecs by assigning the NID dynamically.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:04 +02:00
Takashi Iwai 57307bf24a ALSA: hda - Don't create secondary substream when no independent-hp is used
For VIA codecs, we shouldn't create a substream for independent HP mode,
when no individual HP DAC is found.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:03 +02:00
Takashi Iwai f4a7828bc1 ALSA: hda - Re-implement smart51 detection for VIA codecs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:02 +02:00
Takashi Iwai 4a79616d07 ALSA: hda - Unify output-control parsing in patch_via.c
Parse the output-paths more dynamically, i.e. traverse the paths
from each output pin instead of fixed assignment for each codec.
Now all codecs are using the same output parser code.

The smart51 setup doesn't work with this change, and will be fixed
in the next commits.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:01 +02:00
Takashi Iwai 620e2b28b7 ALSA: hda - Unify input-volume creations in patch_via.c
Now storing the analog-mixer widget in spec, we can simplify the rest
parts.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:01 +02:00
Takashi Iwai 64be285b66 ALSA: hda - Auto-mute all LO and speakers in patch_via.c
Muting all line-outs and/or speakers is more common in other drivers,
so we should follow it, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:24:00 +02:00
Takashi Iwai 3e0693e278 ALSA: hda - Change pin-ctl for auto-muting in patch_via.c
Mute the outputs via pin-controls instead of amps for the auto-mute
handling.  This makes our life easier as it avoids conflict of the states
between the mixer elements and the auto-mute toggles.

With this change, we can use vmaster for the master control easily now.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:59 +02:00
Takashi Iwai 82673bc895 ALSA: hda - Generate PCM names dynamically in patch_via.c
This reduces lots of static strings.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:58 +02:00
Takashi Iwai 291c9e33bf ALSA: hda - Refactor ctl array handling in patch_via.c
No functional change.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:57 +02:00
Takashi Iwai e06e5a2974 ALSA: hda - Defer mixer element creation to the right time in patch_via.c
The jack-detect control should be created at the time of build_controls
callback instead of calling snd_hda_add_ctls() at the tree-parsing time.
For that, copy the control to the temporary array like other cases.

Also, fixed typos of vt1708_jack_detect in all places.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:56 +02:00
Takashi Iwai a766d0d763 ALSA: hda - Fill ADCs dynamically for VIA codecs
Instead of giving the fixed ADC list, parse the widgets and fill in
ADCs dynamically.

Also, probe the stereo-mixer input more dynamically, too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:55 +02:00
Takashi Iwai 24088a58d6 ALSA: hda - Add control to suppress the dynamic pin-power for VIA
Currently VIA driver controls the power-state of each pin per jack
detection.  But, it means that the power-state mismatch may occur when
the machine doesn't give the proper jack-detection.

For avoiding this problem, a new control element "Dynamic Power-Control"
is provided so that user can turn on/off the pin-power control.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:23:55 +02:00
Takashi Iwai 5f4b36d64d ALSA: hda - Remove superfluous NID_MAPPING use for smart51 mixer
Just a minor clean up; nid-mapping can be set directly to the smart51
mixer element.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-20 16:07:02 +02:00
Mark Brown 29376bc7e2 ASoC: Fix DAPM sequence run for per-widget I/O methods
Previously we were using the DAPM context rather than a widget as the
argument for update_bits() so we didn't need to care that our list walk
of widgets left us one beyond the end of the list. Now we're using them
for the register update we need to make sure we're pointing at an actual
widget not the list_head.

Fix originally suggested by Liam on IM.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-20 11:27:10 +01:00
Mark Brown ee8c7e9744 ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency
The Sigma code is in drivers/firmware which is only included on a very
small subset of architectures and so ends up breaking the build on
others.  There's a pending patch to make the directory build as standard
but it's not merged yet.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-18 15:31:38 +01:00
David Henningsson b13e552d37 ALSA: HDA: Remove redundant LPIB quirks for ATI chipset
Now that we have changed the position_fix default for ATI and AMD
to be LPIB (see commit 50e3bbf989), we can remove the quirks that
were added for ATI chipsets.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-17 18:48:23 +02:00
Grant Likely f8db4cc4f2 Merge branch 'spi/merge' into spi/next 2011-06-17 08:32:26 -06:00
Takashi Iwai 3409fcd1f7 Merge branch 'fix/hda' into topic/misc 2011-06-17 14:54:47 +02:00
Takashi Iwai ad2409413d ALSA: hda - Fix no NID error with VIA codecs
The via driver spews warnigs like
	hda-codec: no NID for mapping control Independent HP:0:0
with some codecs because snd_hda_add_nid() is called with nid=0.
This patch fixes it by skipping the call when no corresponding widget
is found.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-17 14:23:46 +02:00
Mike Frysinger 4d1e46b7ef ASoC: Blackfin: allow SPI for SSM2602 parts
This board has hardware switches for selecting SPI or I2C, so don't
require I2C for this driver.

Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-17 10:40:16 +01:00
Clemens Ladisch cf6f1ff17f ALSA: isight: adjust for new queueing API
Since commit 13882a82ee (optimize iso queueing by setting
wake only after the last packet), drivers are required to call
fw_iso_context_queue_flush() after queueing a batch of packets.
The missing call would have an effect only if the controller
queue underruns, but then the DMA would stop completely.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-17 11:37:29 +02:00
Clemens Ladisch f4b1e98aa9 ALSA: firewire-speakers, oxygen, ua101: allow > 10 s periods
Since commit f2b3614cef (Don't check DMA time-out too shortly),
drivers need no longer restrict their PCM period length to be shorter
than 10 seconds.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-17 11:36:30 +02:00
Torsten Schenk 0ec5258d68 ALSA: 6fire - Fix signedness bug
Fixed remaining issues of the signedness bug discovered by Dan Carpenter.
A check was remaining that tests if unsigned rt->rate is >= 0.
Changed that so that rt->rate now consistently uses ARRAY_SIZE(rates)
as invalid rate value and not -1.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-16 21:31:33 +02:00
Barry Song 5d0e7f6170 ASoC: AD1836: rename suspend/resume funcs
Use less specific names for suspend/resume to match the probe/remove funcs
where these are now used.

Signed-off-by: Barry Song <barry.song@analog.com>
Signed-off-by: Scott Jiang <scott.jiang@analog.com>
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:42:10 +01:00
Mike Frysinger 0679059a41 ASoC: AD1836: fix codec name
The codec name should not have a "-codec" suffix since this is not part of
a MFD.  This was incorrectly changed during the multi-component updated.

Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:42:01 +01:00
Mike Frysinger d4d80f5e46 ASoC: AD1836: fix intermixed tab/space indentation
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:41:53 +01:00
Mike Frysinger 15e8705129 ASoC: AD1836: drop unnecessary spi register check
The only thing the init func does is register a spi driver, so if that
fails, we return the value back up to the caller who will display an
error message for us.  So drop the redundant checking/message.

Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:41:46 +01:00
Mike Frysinger 42f32c5591 ASoC: AD1836: clean up comment headers
Signed-off-by: Mike Frysinger <vapier@gentoo.org>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:41:38 +01:00
Lars-Peter Clausen 53a93d58fb ASoC: Blackfin: Add bf5xx-adau1701 machine driver
Add a machine driver to support the ADAU1701 SigmaDSP processors on
Analog Devices BF5XX evaluation boards.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:24:25 +01:00
Lars-Peter Clausen 631ed8a213 ASoC: Add ADAU1701 codec driver
This patch adds support for the Analog Devices ADAU1701 SigmaDSP.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-16 11:24:17 +01:00
Mark Brown b83e60c000 ASoC: Clean up some coding style nits in the bf5xx-i2s-pcm driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-16 11:24:16 +01:00
Harry Butterworth 030aba53ea ALSA: ctxfi: Change PLL initialization code
This is a reworked patch from Creative to change the PLL code to address
unreliable 44100Hz initialization.

Signed-off-by: Harry Butterworth <heb1001@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-16 07:28:57 +02:00
Takashi Iwai e72888e91c ALSA: lola - Fix section mismatch
Add missing __devinit.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-15 15:14:49 +02:00
Liam Girdwood 0445bdf4ae ASoC: dapm - Refactor widget IO functions in preparation for platform widgets.
This time with soc_widget_update_bits reflecting recent soc_update_bits changes.

Currently widget IO is tightly coupled to the CODEC drivers. Future platform DSP
devices have mixer components that can alter power usage and hence require full
DAPM support.

This provides a generic widget IO operation wrapper in preparation for
future patches that implement platform driver DAPM.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-14 14:25:32 +01:00
Takashi Iwai b3c705aa9e ALSA: rawmidi - Use workq for event handling
Kill tasklet usage in rawmidi core code.  Use workq for the event callback
instead of tasklet (which is used only in core/seq/seq_midi.c).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-14 14:37:06 +02:00
Takashi Iwai 30bdee0259 ALSA: es1968,maestro3 - Use work for hw-volume control
Instead of tasklet, use workq for handling the hw-volume control.
This reduces lots of spinlocks.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-14 13:59:51 +02:00
Takashi Iwai 85e4d95da0 Merge branch 'test/pci-rename' into topic/misc 2011-06-14 08:56:42 +02:00
Takashi Iwai ca2585afa0 ALSA: hda - Fix missing static inline to beep dummy function
The commit 2308f4add3 missed static inline
thus it resulted in multiple-definitions error at linking.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-14 08:14:32 +02:00
Harry Butterworth b028b81835 ALSA: ctxfi: Implement a combined capabilities query method to replace multiple have_x query methods.
Signed-off-by: Harry Butterworth <heb1001@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-14 07:34:03 +02:00
Harry Butterworth 55309216ba ALSA: ctxfi: Add support for Creative Titanium HD
Initialise model-specific DAC and ADC parts.
Add controls for output and mic source selection.
Rename some mixer controls according to ControlNames.txt.
Remove Playback switches for Line-in and IEC958-in - these
were controlling the input mute/unmute which affected
capture too.  Use the capture switches to control the
input mute/unmute instead - it's less confusing.
Initialise the WM8775 to invert the left-right clock
to swap the left and right channels of the mic and aux
input.

Signed-off-by: Harry Butterworth <heb1001@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-14 07:32:29 +02:00
Jesper Juhl 37f7ec38ea ALSA: 6fire: Fix double-free bug in usb6fire_fw_ezusb_upload()
We have a double-free bug in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload().
We already call release_firmware(fw) on line 258, so when we then do it
again after usb6fire_fw_ezusb_write() returns <0, we have a double-free.
Easily fixed by just removing the last call to release_firmware().

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-14 07:27:26 +02:00
Mark Brown 65fdd5c05a Merge branch 'for-3.0' into for-3.1
Trival fixup for move of I/O code into separate file.

Conflicts:
	sound/soc/soc-cache.c
2011-06-13 19:21:09 +01:00
Mark Brown e9c039052b ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O bus
This will be removed in -next so let's drop it from mainline as soon as
we can in order to minimise surprises.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-13 19:17:55 +01:00
Mark Brown 2231571214 ASoC: Don't use codec->control_data in bulk write
In order to facilitate merging with the register map I/O replace the use
of control_data for the bulk writes with direct lookup of the client data
from the device.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-13 19:17:33 +01:00
Mark Brown bf3a9e137c ASoC: Add weak routes for sidetone style paths
Normally DAPM will power up any connected audio path. This is not ideal
for sidetone paths as with sidetone paths the audio path is not wanted in
itself, it is only desired if the two paths it provides a sidetone between
are both active. If the sidetone path causes a power up then it can be
hard to minimise pops as we first power up either the sidetone or the main
output path and then power the other, with the second power up potentially
introducing a DC offset.

Address this by introducing the concept of a weak path. If a path is marked
as weak then DAPM will ignore that path when walking the graph, though all
the relevant controls are still available to the application layer to allow
these paths to be configured.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-13 18:59:33 +01:00
Mark Brown 5bef44f9b4 ASoC: Move register I/O code into a separate file
For clarity and to help ongoing refactoring in this area create a new file
to contain the physical I/O functions, separating them out from the cache
operations.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-13 18:56:10 +01:00
Greg Thelen 9857edfd4d ALSA: hda: check make_exec_verb() return value
If given a -1 cmd parameter then make_exec_verb() returns -1 without
setting the res output value.

Prior to this change snd_hda_codec_read() assumed that make_exec_verb()
unconditionally set res regardless of the cmd value.

This change explicitly checks the make_exec_verb() return value before
consuming the potentially unset res value.

Signed-off-by: Greg Thelen <gthelen@google.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-13 17:52:06 +02:00
Mark Brown f0c4205b54 ASoC: Factor out redundant read() functions
We've got a whole bunch of functions which just call straight through to
do_hw_read(). Simplify this situation by removing them and using hw_read()
directly.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-13 14:22:57 +01:00
Takashi Iwai 5ab775c707 Merge branch 'fix/hda' into topic/hda 2011-06-13 08:37:53 +02:00
Joe Perches 2308f4add3 ALSA: hda - Fix beep_device compilation warnings
Using static inline functions can reduce compilation messages
and macro misuse.

 sound/pci/hda/patch_conexant.c: In function ‘patch_cxt5045’:
 sound/pci/hda/patch_conexant.c:1232:3: warning: statement with no effect

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-13 08:36:26 +02:00
Takashi Iwai 54463a66b9 ALSA: hda - Fix wrong auto-mute type for Acer Aspire-one
The auto-mute setup for Acer Aspire-one with ALC268 was set wrongly
during the clean-up of auto-mute function.  Fixed now.

Tested-by: Borislav Petkov <bp@alien8.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-13 08:32:06 +02:00
Florian Zeitz ac5d4b404e ALSA: emu10k1: Add details for E-mu 0404 PCIe version
This patch adds the necessary details to support the PCIe version of
E-MU's 0404 card.
From comparing the PCBs it seems the PCIe version just added a PCIe
chipset and left all other components pretty much in place.
For anyone intrigued to take a look at the PCB there are pictures I took
at <http://babelmonkeys.de/~florob/E-MU%200404/>.

Signed-off-by: Florian Zeitz <florob@babelmonkeys.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-12 21:24:17 +02:00
Adrian Knoth efef054e8c ALSA: hdspm - Add firmware revision ID for RME MADI PCI version
The PCI version of the RME HDSP MADI card uses 0xcf as revision ID. Just
add this to the list of supported cards.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-12 18:50:08 +02:00
Adrian Knoth fedf1535ab ALSA: hdspm - Fix jumping external wordclock frequency in AutoSync mode
When using Word Clock on RME MADI cards, AutoSync mode was alternating
betweeen MADI and WC due to a typo: AutoSync is indicated in the second
status register (status2), not the first one (status).

While the proc output was always correct, the reported WC frequency to
ALSA was unstable as mentioned in

http://mailman.alsa-project.org/pipermail/alsa-devel/2008-March/006723.html

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-12 18:49:56 +02:00
Adrian Knoth c0da00145f ALSA: hdspm - Fix locking in snd_hdspm_midi_input_read
For the MIDI part, we need to acquire (and release) the hmidi->lock,
access to the global hdspm structure is serialized through
hmidi->hdspm->lock instead.

Signed-off-by: Adrian Knoth <adi@drcomp.erfurt.thur.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-12 18:49:42 +02:00
Takashi Iwai 05e205429d Merge branch 'fix/asoc' into for-linus 2011-06-10 17:49:34 +02:00
Takashi Iwai 934c2b6d0c ALSA: use KBUILD_MODNAME for request_irq argument in sound/pci/*
The name argument of request_irq() appears in /proc/interrupts, and
it's quite ugly when the name entry contains a space or special letters.
In general, it's simpler and more readable when the module name appears
there, so let's replace all entries with KBUILD_MODNAME.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-10 16:36:37 +02:00
Daniel T Chen 7ab1fc0af3 ALSA: hda: Fix inaudible internal speakers on CyberpowerPC Gamer Xplorer N57001 laptop
BugLink: https://launchpad.net/bugs/761171

The original reporter needs the model=auto quirk for his internal
speakers to be audible in the latest daily snapshot, so add an entry in
the quirk table for his PCI SSID.

A trivially different version of this patch using the model=asus quirk
should be applied to the 2.6.38 and 2.6.39 stable kernels. We don't use
the asus quirk in 3.0-rc2, because 3.0-rc2's autoparser is much
improved.

Reported-and-tested-by: tomdeering7
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-10 16:31:24 +02:00
Takashi Iwai 3733e424c4 ALSA: Use KBUILD_MODNAME for pci_driver.name entries
The convention for pci_driver.name entry in kernel drivers seem to be
the module name or equivalent ones.  But, so far, almost all PCI sound
drivers use more verbose name like "ABC Xyz (12)", and these are fairly
confusing when appearing as a file name.

This patch converts the all pci_driver.name entries in sound/pci/* to
use KBUILD_MODNAME for more unified appearance.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-10 16:20:20 +02:00
Takashi Iwai 890ee02ac1 ALSA: Use %pV for snd_printk()
Clean up snd_printk() helper using the %pV prefix for recursive printks.
This also automagically fixes an Oops with RO/NX-enabled modules.

Tested-by: Maarten Lankhorst <m.b.lankhorst@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-10 15:35:20 +02:00
Takashi Iwai c0a20263db ALSA: hda - Fix initialization of hp pins with master_mute in Realtek
Some Reatlek model quirks use master_mute bool switch for controlling
the master-mute of outputs.  For these cases, the initialization of HP
pins/amps were forgotten during the transition to the common automute
helper function in 3.0 development time, and resulted in the muted HP
output as default.

This patch fixes the issue by adjusting the HP output explicitly with
master_mute switch.

Tested-by: Michal Hocko <mhocko@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-10 15:28:15 +02:00
Takashi Iwai 20f5e0b36d ALSA: hda - Fix invalid unsol tag for some alc262 model quirks
The tag number was forgotten to be fixed after cleaning up the model
quirks for ALC262 fujitsu and lenovo-3000 models.

Tested-by: Michal Hocko <mhocko@suse.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-10 15:27:32 +02:00
Takashi Iwai 8b0bd2266f ALSA: hda - Fix SSYNC register value for non-Intel controllers
SSYNC register was once defined as 0x34-37 in the old Intel datasheet,
but corrected later to 0x38-3b.  For fixing the register usage, a new
bit-flag is introduced for indicating the old ICH SSYNC register, and
ICH* PCI entries are added explicitly to enable this quirk.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-10 15:00:07 +02:00
Takashi Iwai 695cd4a34e ALSA: hda - Disable SPDIF only when no pin config set for HP with AD1981
Some HP laptops with AD1981 have SPDIF connections, but currently the
driver disables it statically.  Better to check the pin default config
to judge whether to enable or disable the SPDIF.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-10 14:37:04 +02:00
Naveen Krishna Chatradhi f192c0ab24 ASoC: SMDKV310: Enable SPDIF device
Signed-off-by: Naveen Krishna Chatradhi <ch.naveen@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-10 11:05:09 +01:00
Mark Brown 0f28f8e567 Merge branch 'for-3.0' into for-3.1 2011-06-10 11:03:54 +01:00
Uwe Kleine-König 2f2b3cf1dd sound/atmel_ssc_dai: add a missing space to an error message
Signed-off-by: Uwe Kleine-König <u.kleine-koenig@pengutronix.de>
Acked-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-10 10:55:20 +01:00
Sangbeom Kim 33195500ed ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register
If DMA active status should be checked, I2SCON register should be referenced.
In this patch, Fix the incorrect referencing of I2SCON register.

Reported-by : Lakkyung Jung <lakkyung.jung@samsung.com>
Signed-off-by: Sangbeom Kim <sbkim73@samsung.com>
Acked-by: Jassi Brar <jassisinghbrar@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-06-10 10:54:46 +01:00
Liam Girdwood 91d5e6b4f5 ASoC: pcm - rename snd_codec_close() to snd_pcm_close().
Make sure we follow naming convention for all PCM ops.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-09 19:29:35 +01:00
Liam Girdwood b8c0dab9bf ASoC: core - PCM mutex per rtd
In preparation for the new ASoC Dynamic PCM support (AKA DSP support).

The new ASoC Dynamic PCM core allows DAIs to be dynamically re-routed
at runtime between the PCM device end (or Frontend - FE) and the physical DAI
(Backend - BE) using regular kcontrols (just like a hardware CODEC routes
audio in the analog domain). The Dynamic PCM core therefore must be
able to call PCM operations for both the Frontend and Backend(s) DAIs at
the same time.

Currently we have a global pcm_mutex that is used to serialise
the ASoC PCM operations. This patch removes the global mutex
and adds a mutex per RTD allowing the PCM operations to be reentrant and
allow control of more than one DAI at at time. e.g. a frontend PCM hw_params()
could configure multiple backend DAI hw_params() with similar or different
hw parameters at the same time.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-09 19:29:29 +01:00
Mark Brown 2c36c2ce00 Merge branch 'for-3.0' into for-3.1 2011-06-09 15:07:42 +01:00
Liam Girdwood ddee627cf6 ASoC: core - Separate out PCM operations into new file.
In preparation for Dynamic PCM support (AKA DSP support).

There will be future patches that add support to allow PCMs to be dynamically
routed to multiple DAIs at startup and also during stream runtime. This patch
moves the ASoC core PCM operaitions into a new file called soc-pcm.c.  This will
in simplify the ASoC core features into distinct files.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-09 15:07:27 +01:00
Lars-Peter Clausen 4b80b8c2ee ASoC: snd_soc_new_{mixer,mux,pga} make sure to use right DAPM context
Currently it is possible that snd_soc_new_{mixer,mux,pga} is called with a
DAPM context not matching the widgets context. This can lead to a wrong
prefix_len calculation, which will result in undefined behaviour. To avoid
this always use the DAPM context from the widget itself.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-06-09 15:06:54 +01:00
Ralf Baechle 8e1b5adfbe i8253: Make pcsp sound driver use the shared i8253_lock
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Acked-by: Takashi Iwai <tiwai@suse.de>
Cc: alsa-devel@alsa-project.org
Link: http://lkml.kernel.org/r/20110601180610.532642190@duck.linux-mips.net
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>
2011-06-09 15:01:39 +02:00
Ralf Baechle 334955ef96 i8253: Create linux/i8253.h and use it in all 8253 related files
Signed-off-by: Ralf Baechle <ralf@linux-mips.org>
Cc: linux-mips@linux-mips.org
Link: http://lkml.kernel.org/r/20110601180610.054254048@duck.linux-mips.net
Signed-off-by: Thomas Gleixner <tglx@linutronix.de>

 arch/arm/mach-footbridge/isa-timer.c |    2 +-
 arch/mips/cobalt/time.c              |    2 +-
 arch/mips/jazz/irq.c                 |    2 +-
 arch/mips/kernel/i8253.c             |    2 +-
 arch/mips/mti-malta/malta-time.c     |    2 +-
 arch/mips/sgi-ip22/ip22-time.c       |    2 +-
 arch/mips/sni/time.c                 |    2 +-
 arch/x86/kernel/apic/apic.c          |    2 +-
 arch/x86/kernel/apm_32.c             |    2 +-
 arch/x86/kernel/hpet.c               |    2 +-
 arch/x86/kernel/i8253.c              |    2 +-
 arch/x86/kernel/time.c               |    2 +-
 drivers/block/hd.c                   |    2 +-
 drivers/clocksource/i8253.c          |    2 +-
 drivers/input/gameport/gameport.c    |    2 +-
 drivers/input/joystick/analog.c      |    2 +-
 drivers/input/misc/pcspkr.c          |    2 +-
 include/linux/i8253.h                |   11 +++++++++++
 sound/drivers/pcsp/pcsp.h            |    2 +-
 19 files changed, 29 insertions(+), 18 deletions(-)
2011-06-09 15:01:37 +02:00
Mark Brown bf564ea997 Merge branch 'for-3.0' into for-3.1 2011-06-09 12:02:26 +01:00
Timur Tabi 147dfe90f7 ASoC: p1022ds: fix incorrect referencing of device tree properties
Device tree integer properties are encoded in big-endian format, but some of
the Freescale ASoC drivers were assuming that the host is in big-endian format
as well.  Although this is true, it's better to use endian-safe accessors.

Also add a check for a failed ioremap() call in the SSI driver.

Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-09 12:02:08 +01:00
Timur Tabi 0cd114fff9 ASoC: fsl: fix initialization of DMA buffers
The DMA (PCM) driver used by some Freescale PowerPC supports separate DAIs
for playback and capture, so DMA buffers should be allocated only for the
initialized streams.  Instead of checking for the number of active channels,
which apparently is not reliable, check to see if the actual stream object
exists.

Also provide a better name for the DMA interrupt.

Signed-off-by: Timur Tabi <timur@freescale.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-09 11:53:11 +01:00
Mark Brown 3115ae1746 ASoC: WM8804 does not support sample rates below 32kHz
Reported-by: Kieran O'Leary <Kieran.O'Leary@wolfsonmicro.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
2011-06-09 11:51:07 +01:00
Mika Westerberg 51e2cc0c51 ASoC: ep93xx: convert to use the DMA engine API
Now that we have the EP93xx DMA engine driver in place, we convert the ASoC
drivers (I2S, AC97 and PCM) to take advantage of this new API. There are no
functional changes.

Signed-off-by: Mika Westerberg <mika.westerberg@iki.fi>
Acked-by: H Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Grant Likely <grant.likely@secretlab.ca>
2011-06-08 15:45:59 -06:00
Ricardo Neri 2763f45d40 ASoC: twl6040 - According to TWL6040 specification, gain start at 6dB and not -6dB.
Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
2011-06-08 19:52:17 +01:00
Mark Brown 995e54f5fe ASoC: Fix mismerge of Speyside set_bias_level_post()
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-08 17:51:38 +01:00
Mark Brown 47cb55a99b Merge branch 'for-3.0' into for-3.1 2011-06-08 15:25:07 +01:00
Mark Brown 22cb839bc8 ASoC: Support Speyside build variants with WM8962 fitted
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-08 15:24:49 +01:00
Mark Brown 417ceff939 ASoC: Defer all WM8962 clocking configuration until power up
Don't require an audio rate SYSCLK in hw_params() in order to better
support microphone detection use cases.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-08 15:24:48 +01:00
Mark Brown 8f63aaa887 ASoC: Implement base 5 band EQ control for WM8962
ReTune Mobile modes are not currently supported.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-08 15:24:48 +01:00
Mark Brown 649a1a0ef2 ASoC: Report errors when we have a WM8962 IRQ and don't get FLL lock
We really should be getting the interrupt - if we don't get one it's very
likely that the configuration is incorrect and audio will fail. Also
increase the timeout substantially in this case for safety.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-08 15:24:47 +01:00
Mark Brown c7356da9e2 ASoC: Factor out I2C usage in WM8962 driver
The chip can actually support SPI so we shouldn't assume we've got an I2C
device even though that's the most common configuration.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-08 15:24:46 +01:00
Mark Brown ffdaa48aed ASoC: Suppress restore of default register values for rbtree cache sync
Currently the rbtree code will write out the entire register map when
doing a cache sync which is wasteful and will slow things down. Check
to see if the value we're about to write is the default and don't bother
restoring it if it is, either the value will have been retained or the
device will have been reset and holds the value already.

We should really store the defaults in the nodes but this resolves the
immediate issue.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-08 15:24:36 +01:00
Mark Brown 0f82bdf572 ASoC: Fix WM8962 headphone volume update for use of advanced caches
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Cc: stable@kernel.org
2011-06-08 15:18:17 +01:00
Liam Girdwood 0168bf0d13 ASoC: core - Allow components to probe/remove in sequence.
Some ASoC components depend on other ASoC components to provide clocks and
power resources in order to probe() and vice versa for remove().

Allow components to be ordered so that components can be probed() and removed()
in sequences that conform to their dependencies.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-07 18:38:27 +01:00
Liam Girdwood 552d1ef6b5 ASoC: core - Optimise and refactor pcm_new() to pass only rtd
Currently pcm_new() passes in 3 arguments :- card, pcm and DAI.

Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains
card, pcm and DAI along with other members too that are useful too.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-07 18:38:27 +01:00
Takashi Iwai b4a655e81d ALSA: hda - Judge playback stream from stream id in azx_via_get_position()
Instead of checking the azx_dev index with a fixed number (4), check
the stream direction of the assigned substream.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-07 12:26:56 +02:00
Takashi Iwai a810364a04 ALSA: hda - Handle -1 as invalid position, too
When reading from the position-buffer results in -1, handle as it's
invalid and falls back to LPIB mode as well as 0.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-07 12:23:23 +02:00
Mark Brown cf3383fbb0 Merge branch 'for-3.0' into for-3.1 2011-06-07 09:49:47 +01:00
Lars-Peter Clausen 064d58ee3a ASoC: Blackfin: bf5xx-ad1836: Fix codec device name
Fix the codec_name field of the dai_link to match the actual device name
of the codec. Otherwise the card won't be instantiated.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-06-07 09:49:28 +01:00
Lars-Peter Clausen 0c8e2917f2 ASoC: AD1836: Fix build error
Commit f97d0c6d5f ("ASoC: AD1836: Add input gain control for ADC2") contained
a typo in the register name, causing a build error. This patch fixes it.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-07 09:48:21 +01:00
Greg Dietsche bca6b39979 ASoC: wm8940: remove unnecessary if statements
removing unnecessary if(ret) checks

This updated patch corrects a minor spelling problem in the commit message
and resolves two other (similar) issues found in wm8940.c by Jonathan Cameron.

Signed-off-by: Greg Dietsche <Gregory.Dietsche@cuw.edu>
Acked-by: Jonathan Cameron <jic23@cam.ac.uk>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-07 09:47:40 +01:00
Daniel T Chen 0a1896b27b ALSA: hda: Fix quirk for Dell Inspiron 910
BugLink: https://launchpad.net/bugs/792712

The original reporter states that sound from the internal speakers is
inaudible until using the model=auto quirk. This symptom is due to an
existing quirk mask for 0x102802b* that uses the model=dell quirk. To
limit the possible regressions, leave the existing quirk mask but add
a higher priority specific mask for the reporter's PCI SSID.

Reported-and-tested-by: rodni hipp
Cc: <stable@kernel.org> [2.6.38+]
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-07 07:26:01 +02:00
Mark Brown 46758dee72 Merge branch 'for-3.0' into for-3.1 2011-06-06 21:57:54 +01:00
Lars-Peter Clausen 8ca695f273 ASoC: AD1836: Fix setting the PCM format
Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-06-06 21:55:10 +01:00
Lars-Peter Clausen f97d0c6d5f ASoC: AD1836: Add input gain control for ADC2
The AD1836 has a PGA for its second ADC. This patch adds a control for
adjusting the the gain of the PGA.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-06 21:54:43 +01:00
Lars-Peter Clausen 583eadab21 ASoC: AD1836: Remove unused fields from private struct
The control_type field is never used, so it can be removed.  The
control_data field is used to initialize the codec's control_data field,
but since this is also done by the snd-soc-cache core, the redundant
assignment can be removed and the field can be dropped.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-06 21:54:43 +01:00
Lars-Peter Clausen 874ce77bc3 ASoC: AD1836: Add AD1835/AD1837/AD1838/AD1839 support
The AD183X codec devices are mostly register compatible and can easily be
supported by the same driver.  The main difference between those devices
is the number of DACs and ADCs.

This patch adjusts the driver to allocate the controls, DAPM widgets and
routes for the DACs and ADCs dynamically based on the chip type.

The AD1836 is a bit special in that it supports different modes for its second
ADC, so it needs some special handling. Right now the driver hardcodes the mode
to the differential PGA mode.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-06 21:54:42 +01:00
Lars-Peter Clausen 2cf0342822 ASoC: AD1836: Use snd_soc_update_bits for read-modify-write
Use snd_soc_update_bits for read-modify-write register access instead of
open-coding it using snd_soc_read and snd_soc_write.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-06 21:54:41 +01:00
Lars-Peter Clausen 90bc11d1d0 ASoC: AD1836: Add ADC/DAC controls helper macros
The different ADC and DAC controls follow the same scheme, so add some helper
macros for declaring them.
This should make the code a bit more readable and also decreases the code size
a bit.

Signed-off-by: Lars-Peter Clausen <lars@metafoo.de>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-06 21:54:41 +01:00
Mark Brown 85e9e76638 ASoC: Manage Speyside system clocking only in bias management
Now that the CODEC driver supports it defer configuration of the system
clock until bias management which is a much more idiomatic place to do
system power control and makes things a lot more happy when we're using
both interfaces.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 21:47:28 +01:00
Mark Brown cc4c670a41 ASoC: Only provide a default bias level update for CODEC contexts
This allows the card driver to use the bias level variable more easily in
multi component systems.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 21:47:05 +01:00
Mark Brown d4c6005f8e ASoC: Add context parameter to card DAPM callbacks
The card callback will get called for each DAPM context in the card so it
can be useful for it to know which device is currently undergoing a
transition.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 21:46:45 +01:00
Mark Brown 171ec6b089 ASoC: Simplify logic in snd_soc_dapm_set_bias_level()
No functional changes but much less indentation.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 21:46:19 +01:00
Mark Brown 4113e44316 ASoC: Remove trace for DAPM bias level logging
It's redundant now thanks to the use of the generic trace infrastructure.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 21:46:00 +01:00
Mark Brown 88d960864e ASoC: Indentation fix for null loop operation
More with the legibility.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 21:45:44 +01:00
Mark Brown dfcc9047c9 ASoC: Don't bring the CODEC up to full power for supplies and biases
If the only widgets active within a CODEC are supplies and micbiases we
are not passing audio, we are probably just doing microphone detection.
This will not generally require either fully accurate reference voltages
or much power so

If this turns out to be unsuitable for some systems we can provide a
facility to override this decision.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 21:45:44 +01:00
Mark Brown 56fba41f8f ASoC: Specify target bias state directly as a bias state
Rather than a simple flag to say if we want the DAPM context to be at full
power specify the target bias state. This should have no current effect
but is a bit more direct and so makes it easier to change our decisions
about the which bias state to go into in future.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 21:45:44 +01:00
Mark Brown 6dffdea700 ASoC: Allow WM8915 BCLK calculation outside hw_params()
Allow more dynamic management of the device clocking by allowing BCLK to
be calculated when we set SYSCLK. This means that if the system is idle
when hw_params() runs then we don't try to use the SYSCLK used in that case
to set up the BCLK dividers, we can instead wait until a later point such
as bias level configuration. This makes it easier to manage low power modes.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 21:45:44 +01:00
Mark Brown bd4f2acb8d Merge branch 'for-3.0' into for-3.1 2011-06-06 19:34:58 +01:00
Mark Brown fd137e2bba ASoC: Check for NULL register bank in snd_soc_get_cache_val()
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 19:34:11 +01:00
Mark Brown 78bf3c9ab6 ASoC: Enforce the mask in snd_soc_update_bits()
Avoids issues if someone does a read followed by restore and doesn't mask
out only the bits being updated.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 12:48:42 +01:00
Mark Brown 51b3b5cabb ASoC: Error out when FLL lock interrupt is not delivered on WM8915
When the FLL locks on the WM8915 an interrupt is generated.  For safety
error out if we don't get that interrupt when the IRQ output of the
WM8915 is hooked up.  Since we *really* expect an interrupt but the
threaded IRQ handler may take a bit longer than expected to get
scheduled also dramatically increase the delay in this case.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 12:47:57 +01:00
Mark Brown ea7b437836 ASoC: Suppress noop SYSCLK updates in WM8915
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 12:47:17 +01:00
Mark Brown 84abd1b395 Merge branch 'for-3.0' into for-3.1 2011-06-06 12:47:06 +01:00
Mark Brown 6ac340623c ASoC: Add missing break in WM8915 FLL source selection
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 12:46:52 +01:00
Mark Brown 1622ee1822 ASoC: Only update SYSCLK_ENA when pausing WM8915 SYSCLK
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-06 12:46:33 +01:00
Mark Brown aa72f6899b Merge branch 'for-3.0' into for-3.1 2011-06-06 12:26:02 +01:00
Stephen Warren 384a48d715 ALSA: hda: HDMI: Support codecs with fewer cvts than pins
The general concept of this change is to create a PCM device for each
pin widget instead of each converter widget. Whenever a PCM is opened,
a converter is dynamically selected to drive that pin based on those
available for muxing into the pin.

The one thing this model doesn't support is a single PCM/converter
sending audio to multiple pin widgets at once.

Note that this means that a struct hda_pcm_stream's nid variable is
set to 0 except between a stream's open and cleanup calls. The dynamic
de-assignment of converters to PCMs occurs within cleanup, not close,
in order for it to co-incide with when controller stream IDs are
cleaned up from converters.

While the PCM for a pin is not open, the pin is disabled (its widget
control's PIN_OUT bit is cleared) so that if the currently routed
converter is used to drive a different PCM/pin, that audio does not
leak out over a disabled pin.

We use the recently added SPDIF virtualization feature in order to
create SPDIF controls for each pin widget instead of each converter
widget, so that state is specific to a PCM.

In order to support this, a number of more mechanical changes are made:

* s/nid/pin_nid/ or s/nid/cvt_nid/ in many places in order to make it
  clear exactly what the code is dealing with.

* We now have per_pin and per_cvt arrays in hdmi_spec to store relevant
  data. In particular, we store a converter's capabilities in the per_cvt
  entry, rather than relying on a combination of codec_pcm_pars and
  the struct hda_pcm_stream.

* ELD-related workarounds were removed from hdmi_channel_allocation
  into hdmi_instrinsic in order to simplifiy infoframe calculations and
  remove HW dependencies.

* Various functions only apply to a single pin, since there is now
  only 1 pin per PCM. For example, hdmi_setup_infoframe,
  hdmi_setup_stream.

* hdmi_add_pin and hdmi_add_cvt are more oriented at pure codec parsing
  and data retrieval, rather than determining which pins/converters
  are to be used for creating PCMs.

This is quite a large change; it may be appropriate to simply read the
result of the patch rather than the diffs. Some small parts of the change
might be separable into different patches, but I think the bulk of the
change will probably always be one large patch. Hopefully the change
isn't too opaque!

This has been tested on:

* NVIDIA GeForce 400 series discrete graphics card. This model has the
  classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM
  audio to a PC monitor that supports audio.

* NVIDIA GeForce 520 discrete graphics card. This model is the new
  1 codec n converters m pins m>n model. Tested stereo PCM audio to a
  PC monitor that supports audio.

* NVIDIA GeForce 400 series laptop graphics chip. This model has the
  classical 1:1:1 codec:converter:pcm widget model. Tested stereo PCM,
  multi-channel PCM, and AC3 pass-through to an AV receiver.

* Intel Ibex Peak laptop. This model is the new 1 codec n converters m
  pins m>n model. Tested stereo PCM, multi-channel PCM, and AC3 pass-
  through to an AV receiver.

Note that I'm not familiar at all with AC3 pass-through. Hence, I may
not have covered all possible mechanisms that are applicable here. I do
know that my receiver definitely received AC3, not decoded PCM. I tested
with mplayer's "-afm hwac3" and/or "-af lavcac3enc" options, and alsa a
WAV file that I believe has AC3 content rather than PCM.

I also tested:
* Play a stream
* Mute while playing
* Stop stream
* Play some other streams to re-assign the converter to a different
  pin, PCM, set of SPDIF controls, ... hence hopefully triggering
  cleanup for the original PCM.
* Unmute original stream while not playing
* Play a stream on the original pin/PCM.

This was to test SPDIF control virtualization.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-06 12:58:14 +02:00
Stephen Warren 2def8172c6 ALSA: hda: hdmi_eld_update_pcm_info: update a stream in place
A future change won't store an entire hda_pcm_stream just to represent
the capabilities of a codec; a custom data-structure will be used. To
ease that transition, modify hdmi_eld_update_pcm_info to expect the
hda_pcm_stream to be pre-initialized with the codec's capabilities, and
to update those capabilities in-place based on the ELD.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-06 12:58:09 +02:00
Stephen Warren 3aaf898025 ALSA: hda: Separate generic and non-generic implementations
A future change will significantly rework the generic implementation
in order to support codecs with a different number of pins and
converters. Isolate the more custom codec variants from this change by
duplicating the small portions of generic code they share. This
simplifies the later rework of that previously shared code, since we
don't have to consider the more custom codecs, and also prevents
support for those codecs from regressing.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-06 12:58:05 +02:00
Stephen Warren 74b654c957 ALSA: hda: Virtualize SPDIF out controls
The SPDIF output controls apply to converter widgets. A future change
will create a PCM device per pin widget, and hence a set of SPDIF output
controls per pin widget, for certain HDMI codecs. To support this, we
need the ability to virtualize the SPDIF output controls. Specifically:

* Controls can be "unassigned" from real hardware when a converter is
  not used for the PCM the control was created for.
* Control puts only write to hardware when they are assigned.
* Controls can be "assigned" to real hardware when a converter is picked
  to support output for a particular PCM.
* When a converter is assigned, the hardware is updated to the cached
  configuration.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-06 12:51:59 +02:00
Stephen Warren 7c93597627 ALSA: hda: Allow multple SPDIF controls per codec
Currently, the data that backs the kcontrols created by
snd_hda_create_spdif_out_ctls is stored directly in struct hda_codec. When
multiple sets of these controls are stored, they will all manipulate the
same data, causing confusion. Instead, store an array of this data, one
copy per converter, to isolate the controls.

This patch would cause a behavioural change in the case where
snd_hda_create_spdif_out_ctls was called multiple times for a single codec.
As best I can tell, this is never the case for any codec.

This will be relevant at least for some HDMI audio codecs, such as the
NVIDIA GeForce 520 and Intel Ibex Peak. A future change will modify the
driver's handling of those codecs to create multiple PCMs per codec. Note
that this issue isn't affected by whether one creates a PCM-per-converter
or PCM-per-pin; there are multiple of both within a single codec in both
of those codecs.

Note that those codecs don't currently create multiple PCMs for the codec
due to the default HW mux state of all pins being to point at the same
converter, hence there is only a single converter routed to any pin, and
hence only a single PCM.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-06 12:48:59 +02:00
Stephen Warren c3d5210575 ALSA: hda: Gate ELD usage only by whether ELD is valid
It's perfectly valid for an ELD to contain no SADs. This simply means that
only basic audio is supoprted.

In this case, we still want to limit a PCM's capabilities based on the ELD.

History:

* Originally, ELD application was limited solely by sad_count>0, which
  was used to check that an ELD had been read.
* Later, eld_valid was added to the conditions to satisfy.

This change removes the original sad_count>0 check, which when squashed
with the above two changes ends up replacing if (sad_count) with
if (eld_valid).

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-06 12:48:45 +02:00
Mark Brown 05d3962cc9 Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into for-3.0 2011-06-06 10:38:23 +01:00
Linus Torvalds 0d6925d43b 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: usb - turn off de-emphasis in s/pdif for cm6206
  ALSA: asihpi: Use angle brackets for system includes
  ALSA: fm801: add error handling if auto-detect fails
  ALSA: hda - Check pin support EAPD in ad198x_power_eapd_write
  ALSA: hda - Fix HP and Front pins of ad1988/ad1989 in ad198x_power_eapd()
  ALSA: 6fire: Don't leak firmware in error path
  ASoC: Fix wm_hubs input PGA ZC bits
  ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared
2011-06-06 17:51:28 +09:00
Takashi Iwai 3190dad97b Merge branch 'fix/asoc' into for-linus 2011-06-06 09:28:49 +02:00
Linus Torvalds bb3d6bf191 Revert "ASoC: Update cx20442 for TTY API change"
This reverts commit ed0bd2333c.

Since we reverted the TTY API change, we should revert the ASoC update
to it too.

Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Greg Kroah-Hartman <gregkh@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2011-06-04 07:00:50 +09:00
Eric Lammerts 157186bc18 ALSA: usb - turn off de-emphasis in s/pdif for cm6206
CM6206: Turn off de-emphasis channel status bit in S/PDIF output.

Signed-off-by: Eric Lammerts <eric@lammerts.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-03 18:22:56 +02:00
Ricardo Neri 68d1c4a73c ASoC: OMAP: Update Makefile and Kconfig for HDMI audio
Update Makefile and Kconfig to build HDMI audio support for
OMAP4 SDP and Panda boards.

Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
2011-06-03 13:43:46 +01:00
Ricardo Neri 55b95e0e60 ASoC: OMAP4: Add HDMI Audio machine driver for OMAP4 boards
Add machine driver for HDMI audio on OMAP4 boards. This driver is
in charge of putting together the HDMI audio codec and the CPU DAI
and register the HDMI sound card with ALSA.

Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
2011-06-03 13:43:45 +01:00
Ricardo Neri bca2e41d31 ASoC: OMAP: Add CPU DAI driver for HDMI
Addition of the HDMI CPU DAI driver for OMAP4. This driver is in
charge of configuring DMA settings for HDMI. Also, it finds
the HDMI video device and determines if audio playback can proceed.

Signed-off-by: Ricardo Neri <ricardo.neri@ti.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Liam Girdwood <lrg@ti.com>
2011-06-03 13:43:45 +01:00
Joe Perches d50a2fb636 ALSA: asihpi: Use angle brackets for system includes
Use the normal include style.

Signed-off-by: Joe Perches <joe@perches.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-03 11:46:37 +02:00
Joachim Eastwood 840d8e5e96 ASoC: atmel_ssc: Don't try to free ssc if request failed
We should only call ssc_free() when ssc_request() succeeds or bad
things will happen.

Signed-off-by: Joachim Eastwood <joachim.eastwood@jotron.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-03 10:04:17 +01:00
Dan Carpenter 9676001559 ALSA: fm801: add error handling if auto-detect fails
In the original code if auto detect failed and tea575x_tuner == 4
then we copy bogus information to chip->tea.card.  I've changed the
autodetect code to cleanup and return -ENODEV on error instead.

Signed-off-by: Dan Carpenter <error27@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-03 10:11:17 +02:00
Raymond Yau a01ef051d5 ALSA: hda - Check pin support EAPD in ad198x_power_eapd_write
Check whether the pin supports EAPD in ad198x_power_eapd_write.

Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-03 10:08:46 +02:00
Takashi Iwai 4dffbe03d1 ALSA: hda - Fix HP and Front pins of ad1988/ad1989 in ad198x_power_eapd()
In ad198x_power_eapd(), wrong pin NIDs are used for controlling EAPD for
HP and Front outputs of AD1988/AD1989.  These are actually same with the
ones for AD1984 & co, port-A is 0x11 and port-D 0x12.

Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-03 10:05:02 +02:00
Mark Brown e6a9be0bb0 ASoC: Use a lower detection rate when monitoring headphones on WM8915
We only need to increase the detection rate to maximum if we're monitoring
for button presses as the response times needed for user interaction there
are much lower.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-02 18:57:08 +01:00
Jesper Juhl bf0be0e951 ALSA: 6fire: Don't leak firmware in error path
One of the error paths in
sound/usb/6fire/firmware.c::usb6fire_fw_ezusb_upload() neglects to free
the memory allocated for the firmware before returning, thus leaking the
memory.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-06-02 19:56:31 +02:00
Mark Brown 1e025a3692 ASoC: Update speyside audio driver for hardware revision 2
Revision 2 of the Speyside platform supplies a 32kHz clock on MCLK2 rather
than MCLK1.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-01 20:20:59 +01:00
Mark Brown cf4a39105a ASoC: Remove internally generated WM8915 supplies
DCVDD and MICVDD are intended to be (and almost always are) generated by
on-board LDOs which are transparently controlled by the driver so we
shouldn't really be requesting them from the regulator API. If the driver
is updated to support external supply of these then we will need to change
the way we handle this.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-01 19:43:34 +01:00
Julia Lawall a2dc56c8a0 ASoC: add missing clk_put to nuc900-ac97
This goto is after the call to clk_get, so it should go to the label that
includes a call to clk_put.

A simplified version of the semantic match that finds this problem is as
follows: (http://coccinelle.lip6.fr/)

// <smpl>
@r exists@
expression e1,e2;
statement S;
@@

e1 = clk_get@p1(...);
... when != e1 = e2
    when != clk_put(e1)
    when any
if (...) { ... when != clk_put(e1)
               when != if (...) { ... clk_put(e1) ... }
* return@p3 ...;
 } else S
// </smpl>

Signed-off-by: Julia Lawall <julia@diku.dk>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-06-01 19:20:22 +01:00
Mark Brown a1e9adc00e ASoC: Support edge triggered IRQs for WM8915
Really this should be something the IRQ core can cope with for us but since
it doesn't currently do so (at least for threaded interrupts like this) do
so in the driver. This allows us to run with interrupt controllers that
only support edge triggered interrupts.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-01 19:19:19 +01:00
Mark Brown 37aa716a57 ASoC: Staticize ak4641_dai
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-06-01 19:18:59 +01:00
Mark Brown d21685ec25 Merge branch 'for-2.6.40' into for-2.6.41 2011-05-30 10:54:18 +08:00
Axel Lin 74ab24af4f ASoC: Remove redundant freq assignment for max98095->sysclk/max98088->sysclk
Current implementation set max98095->sysclk/max98088->sysclk to freq twice.
Set it once is enough, this patch removes the first assignment in case
we may set invalid clock frequency to max98095->sysclk/max98088->sysclk.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Acked-by: Peter Hsiang <peter.hsiang@maxim-ic.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-29 01:57:21 +08:00
Linus Torvalds 2a56d22202 Merge branch 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm
* 'for-linus' of master.kernel.org:/home/rmk/linux-2.6-arm: (45 commits)
  ARM: 6945/1: Add unwinding support for division functions
  ARM: kill pmd_off()
  ARM: 6944/1: mm: allow ASID 0 to be allocated to tasks
  ARM: 6943/1: mm: use TTBR1 instead of reserved context ID
  ARM: 6942/1: mm: make TTBR1 always point to swapper_pg_dir on ARMv6/7
  ARM: 6941/1: cache: ensure MVA is cacheline aligned in flush_kern_dcache_area
  ARM: add sendmmsg syscall
  ARM: 6863/1: allow hotplug on msm
  ARM: 6832/1: mmci: support for ST-Ericsson db8500v2
  ARM: 6830/1: mach-ux500: force PrimeCell revisions
  ARM: 6829/1: amba: make hardcoded periphid override hardware
  ARM: 6828/1: mach-ux500: delete SSP PrimeCell ID
  ARM: 6827/1: mach-netx: delete hardcoded periphid
  ARM: 6940/1: fiq: Briefly document driver responsibilities for suspend/resume
  ARM: 6938/1: fiq: Refactor {get,set}_fiq_regs() for Thumb-2
  ARM: 6914/1: sparsemem: fix highmem detection when using SPARSEMEM
  ARM: 6913/1: sparsemem: allow pfn_valid to be overridden when using SPARSEMEM
  at91: drop at572d940hf support
  at91rm9200: introduce at91rm9200_set_type to specficy cpu package
  at91: drop boot_params and PLAT_PHYS_OFFSET
  ...
2011-05-27 19:51:32 -07:00
Linus Torvalds 46f2cc8051 ALSA: fix hda AZX_DCAPS_NO_TCSEL quirk check in driver_caps
Commit 9477c58e33 ("ALSA: hda - Reorganize controller quriks with bit
flags") changed the driver type compares into various quirk bits.
However, the check for AZX_DCAPS_NO_TCSEL got reverted: instead of
clearing TCSEL for chipsets that have that standard capability, it
cleared then when the NO_TCSEL bit was set.

This can lead to noise and repeated sounds - a weird "echo" behavior.
As the comment just above says: "Ensuring these bits are 0 clears
playback static on some HD Audio codecs".  Which is definitely true at
least on my Core i5 Westmere system.

Cc: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>
2011-05-27 19:45:28 -07:00
Linus Torvalds 09cefbb605 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: (26 commits)
  ASoC: Fix power down for widgetless per-card DAPM context case
  ASoC: wm1250-ev1: Define "WM1250 Output" with SND_SOC_DAPM_OUTPUT
  ASoC: Remove duplicate linux/delay.h inclusion.
  ASoC: sam9g20_wm8731: use the proper SYSCKL value
  ASoC: wm8731: fix wm8731_check_osc() connected condition
  ALSA: hda - Reorganize controller quriks with bit flags
  ALSA: hda - Use snd_printd() in snd_hda_parse_pin_def_config()
  ALSA: core: remove unused variables.
  ALSA: HDA: Increase MAX_HDMI_PINS
  ALSA: PCM - Don't check DMA time-out too shortly
  MAINTAINERS: add FireWire audio maintainer
  ALSA: usb-audio: more control quirks for M-Audio FastTrack devices
  ALSA: usb-audio: add new quirk type QUIRK_AUDIO_STANDARD_MIXER
  ALSA: usb-audio: export snd_usb_feature_unit_ctl
  ALSA: usb-audio: rework add_control_to_empty()
  ALSA: usb-audio: move assignment of chip->ctrl_intf
  ALSA: hda - Use model=auto for Lenovo G555
  ALSA: HDA: Unify HDMI hotplug handling.
  ALSA: hda - Force AD1988_6STACK_DIG for Asus M3N-HT Deluxe
  ASoC: core - remove superfluous new line.
  ...
2011-05-27 10:10:51 -07:00
Mark Brown ea02c63d57 ASoC: Fix wm_hubs input PGA ZC bits
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-05-27 22:17:09 +08:00
Stephen Warren a5fe6be42e ASoC: Tegra: Enable Kaen HP_MUTE at boot
We want the default state of the HP_MUTE signal to be asserted, so that
the headphones are muted before the first audio playback. Without this,
the headphones are left unmuted until shortly after the first audio
playback completes.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-27 22:13:54 +08:00
Mark Brown 2ac8b6f41a ASoC: Use explicit endianness conversion in snd_soc_16_8_write()
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-05-27 22:01:40 +08:00
Mark Brown 94228bcf8c ASoC: Use cpu_to_be16() in 8x16 write
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-05-27 22:00:58 +08:00
Mark Brown f06f136fe0 ASoC: Convert 7x9 write to use cpu_to_be16()
Run the data through cpu_to_be16() so it's at least clear what we're up to.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-05-27 22:00:38 +08:00
Stephen Warren 1007da0604 ASoC: Fix dapm_is_shared_kcontrol so everything isn't shared
Commit af46800 ("ASoC: Implement mux control sharing") introduced
function dapm_is_shared_kcontrol.

When this function returns true, the naming of DAPM controls is derived
from the kcontrol_new. Otherwise, the name comes from the widget (and
possibly a widget's naming prefix).

A bug in the implementation of dapm_is_shared_kcontrol made it return 1
in all cases. Hence, that commit caused a change in control naming for
all controls instead of just shared controls.

Specifically, a control is always considered shared because it is always
compared against itself. Solve this by never comparing against the widget
containing the control being created.

Equally, controls should never be shared between DAPM contexts; when the
same codec is instantiated multiple times, the same kcontrol_new will be
used. However, the control should no be shared between the multiple
instances.

I tested that with the Tegra WM8903 driver:
* Shared is now mostly 0 as expected, and sometimes 1.
* The expected controls are still generated after this change.

However, I don't have any systems that have a widget/control naming
prefix, so I can't test that aspect.

Thanks for Jarkko Nikula for pointing out how to fix this.

Reported-by: Liam Girdwood <lrg@ti.com>
Tested-by: Jarkko Nikula <jhnikula@gmail.com>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-27 21:49:36 +08:00
Takashi Iwai cf73df1e29 Merge branch 'fix/asoc' into for-linus 2011-05-27 08:03:03 +02:00
Takashi Iwai d1227e3fe0 Merge branch 'fix/misc' into for-linus 2011-05-27 08:02:59 +02:00
Linus Torvalds 9f1912c48c Merge branch 'for-next' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6
* 'for-next' of git://git.kernel.org/pub/scm/linux/kernel/git/sameo/mfd-2.6: (57 commits)
  regulator: Fix 88pm8607.c printk format warning
  input: Add support for Qualcomm PMIC8XXX power key
  input: Add Qualcomm pm8xxx keypad controller driver
  mfd: Add omap-usbhs runtime PM support
  mfd: Fix ASIC3 SD Host Controller Configuration size
  mfd: Fix omap_usbhs_alloc_children error handling
  mfd: Fix omap usbhs crash when rmmoding ehci or ohci
  mfd: Add ASIC3 LED support
  leds: Add ASIC3 LED support
  mfd: Update twl4030-code maintainer e-mail address
  mfd: Correct the name and bitmask for ab8500-gpadc BTempPullUp
  mfd: Add manual ab8500-gpadc batt temp activation for AB8500 3.0
  mfd: Provide ab8500-core enumerators for chip cuts
  mfd: Check twl4030-power remove script error condition after i2cwrite
  mfd: Fix twl6030 irq definitions
  mfd: Add phoenix lite (twl6025) support to twl6030
  mfd: Avoid to use constraint name in 88pm860x regulator driver
  mfd: Remove checking on max8925 regulator[0]
  mfd: Remove unused parameter from 88pm860x API
  mfd: Avoid to allocate 88pm860x static platform data
  ...
2011-05-26 12:14:20 -07:00
Linus Torvalds 829ae27329 Merge branch 'omap-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tmlind/linux-omap-2.6
* 'omap-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tmlind/linux-omap-2.6: (33 commits)
  OMAP3: PM: Boot message is not an error, and not helpful, remove it
  OMAP3: cpuidle: change the power domains modes determination logic
  OMAP3: cpuidle: code rework for improved readability
  OMAP3: cpuidle: re-organize the C-states data
  OMAP3: clean-up mach specific cpuidle data structures
  OMAP3 cpuidle: remove useless SDP specific timings
  usb: otg: OMAP4430: Powerdown the internal PHY when USB is disabled
  usb: otg: OMAP4430: Fixing the omap4430_phy_init function
  usb: musb: am35x: fix compile error when building am35x
  usb: musb: OMAP4430: Power down the PHY during board init
  omap: drop board-igep0030.c
  omap: igep0020: add support for IGEP3
  omap: igep0020: minor refactoring
  omap: igep0020: name refactoring for future merge with IGEP3
  omap: Remove support for omap2evm
  arm: omap2plus: GPIO cleanup
  omap: musb: introduce default board config
  omap: move detection of NAND CS to common-board-devices
  omap: use common initialization for PMIC i2c bus
  omap: consolidate touch screen initialization among different boards
  ...
2011-05-26 12:11:54 -07:00
Samuel Ortiz e45be4b5fc mfd: Use mfd cell platform_data for wm8400 cells platform bits
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.

Reviewed-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
2011-05-26 19:45:14 +02:00
Samuel Ortiz cb5811cf32 mfd: Use mfd cell platform_data for davinci cells platform bits
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.

Cc: Miguel Aguilar <miguel.aguilar@ridgerun.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
2011-05-26 19:45:13 +02:00
Samuel Ortiz a4579ad2bb mfd: Use mfd cell platform_data for twl4030 codec cells platform bits
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.

Cc: Peter Ujfalusi <peter.ujfalusi@nokia.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Liam Girdwood <lrg@slimlogic.co.uk>
Cc: Dmitry Torokhov <dtor@mail.ru>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
2011-05-26 19:45:09 +02:00
Samuel Ortiz 9e554696c0 mfd: Use mfd cell platform_data for wl1273 cells platform bits
With the addition of a platform device mfd_cell pointer, MFD drivers
can go back to passing platform data back to their sub drivers.
This allows for an mfd_cell->mfd_data removal and thus keep the
sub drivers MFD agnostic. This is mostly needed for non MFD aware
sub drivers.

Cc: Matti Aaltonen <matti.j.aaltonen@nokia.com>
Signed-off-by: Samuel Ortiz <sameo@linux.intel.com>
2011-05-26 19:45:02 +02:00
Jarkko Nikula ea77b94774 ASoC: Fix power down for widgetless per-card DAPM context case
Commit 52ba67b ("ASoC: Force all DAPM contexts into the same bias state")
powers up all the DAPM contexts in a card if any DAPM context becomes
active. Unfortunately power down newer happens if per-card DAPM context
doesn't have any widgets.

Reason for this is that power state of per-card DAPM context without
widgets is never cleared and thus all the DAPM contexts remain permanently
active. Test for widgetless calling DAPM context in dapm_power_widgets()
doesn't work for per-card DAPM context since power change is never
originating from widgetless per-card DAPM context.

Fix this by pre-clearing power state flag of non-codec DAPM context at the
beginning of power sequence.

Signed-off-by: Jarkko Nikula <jhnikula@gmail.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-26 22:58:14 +08:00
Axel Lin 979f486944 ASoC: wm1250-ev1: Define "WM1250 Output" with SND_SOC_DAPM_OUTPUT
Codec output pin should be defined with SND_SOC_DAPM_OUTPUT.

Signed-off-by: Axel Lin <axel.lin@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-26 22:11:24 +08:00
Jesper Juhl 65afc4118d ASoC: Remove duplicate linux/delay.h inclusion.
It's enough to include linux/delay.h just once in
sound/soc/codecs/wm8915.c, so remove the duplicate.

Signed-off-by: Jesper Juhl <jj@chaosbits.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-26 22:11:23 +08:00
Nicolas Ferre 6bb74a7293 ASoC: sam9g20_wm8731: use the proper SYSCKL value
at91sam9g20 is providing master clock to wm8731: not using a crystal but an
external MCLK. We can avoid conflict and save power using WM8731_SYSCLK_MCLK as
we do not need oscillator to be powered.

Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-26 22:11:19 +08:00
Nicolas Ferre 5a195b4450 ASoC: wm8731: fix wm8731_check_osc() connected condition
The crystal oscillator is only enabled if the WM8731_SYSCLK_XTAL master clock
is specified. Fix the connected() struct snd_soc_dapm_route function to take
this into account. Oscillator is not enabled on machine that need it otherwise.

Machine drivers have to make sure that they use the proper SYSCLK value.

Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-26 22:11:16 +08:00
Nicolas Ferre 2cdcd951c4 ASoC: atmel_ssc_dai: fix ssc error path
We do not have to free a resource that is not allocated yet.

Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-26 22:10:11 +08:00
Nicolas Ferre 97b4fc3c44 ASoC: trivial: typo in atmel_pcm_dma_params strucutre comment
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-26 22:10:07 +08:00
Nicolas Ferre 7309d2e28d ASoC: trivial: typo in debug comment
Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-26 22:09:28 +08:00
Stephen Warren 82e14e8bdd ASoC: core: Don't schedule deferred_resume_work twice
For cards that have two or more DAIs, snd_soc_resume's loop over all
DAIs ends up calling schedule_work(deferred_resume_work) once per DAI.
Since this is the same work item each time, the 2nd and subsequent
calls return 0 (work item already queued), and trigger the dev_err
message below stating that a work item may have been lost.

Solve this by adjusting the loop to simply calculate whether to run the
resume work immediately or defer it, and then call schedule work (or not)
one time based on that.

Note: This has not been tested in mainline, but only in chromeos-2.6.38;
mainline doesn't support suspend/resume on Tegra, nor does the mainline
Tegra ASoC driver contain multiple DAIs. It has been compile-checked in
mainline.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-26 22:06:08 +08:00
Takashi Iwai 9477c58e33 ALSA: hda - Reorganize controller quriks with bit flags
Introduce bit-flags indicating the necessary controller quirks, and
set them in pci driver_data field.  This simplifies the checks in the
driver code and avoids the pci-id lookup in different places.

Also, this patch adds the PCI ID entry for AMD Hudson.  AMD Hudson
requires a similar workaround like ATI SB while other generic ATI and
AMD controllers don't need but some ATI-HDMI quirks.  So, we need a
different entry for Hudson.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-26 14:43:07 +02:00
Takashi Iwai 0b6267376d ALSA: hda - Use snd_printd() in snd_hda_parse_pin_def_config()
Fixed the wrong usage of snd_printdd() for debug prints of input
entries.  It should be snd_printd() like others.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-26 14:10:44 +02:00
Luca Tettamanti 78fa2c4d24 ALSA: core: remove unused variables.
Drop a few variables that are never read.

Signed-off-by: Luca Tettamanti <kronos.it@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-26 08:19:04 +02:00
Stephen Warren 739266566a ALSA: HDA: Increase MAX_HDMI_PINS
The recently introduced NVIDIA GeForce GT 520 has 4 pins within a single
codec. Bump MAX_HDMI_PINS to accomodate this. Also bump MAX_HDMI_CVTS
to match it; this might be needed later too.

Signed-off-by: Stephen Warren <swarren@nvidia.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-26 08:17:59 +02:00
Takashi Iwai f2b3614cef ALSA: PCM - Don't check DMA time-out too shortly
When the PCM period size is set larger than 10 seconds, currently the
PCM core may abort the operation with DMA-error due to the fixed timeout
for 10 seconds.  A similar problem is seen in the drain operation that
has a fixed timeout of 10 seconds, too.

This patch fixes the timeout length depending on the period size and
rate, also including the consideration of no_period_wakeup flag.

Reported-by: Raymond Yau <superquad.vortex2@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-26 08:09:38 +02:00
Daniel Mack b6f7d7c8bf ASoC: Fix comment in cs4270 codec driver
The comment does not reflect reality anymore since the multi-component
monster patch landed. Things are matched by names now, and not by
exporting and referencing a struct. Fix it to avoid confusion.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Acked-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-26 09:26:22 +08:00
Russell King ae1d3b974e Merge branch 'for-rmk' of git://github.com/at91linux/linux-2.6-at91 into devel-stable 2011-05-26 00:41:21 +01:00
Russell King 586893ebc4 Merge branch 'for-rmk' of git://git.kernel.org/pub/scm/linux/kernel/git/kgene/linux-samsung into devel-stable
Conflicts:
	arch/arm/Kconfig
	arch/arm/mach-exynos4/mach-nuri.c
2011-05-25 21:47:48 +01:00
Ben Gardiner bb5b5fd4d4 ASoC: davinci-pcm: comments for the conversion to BATCH mode
In the previous commit 'ASoC: davinci-pcm: convert to BATCH mode', the phase
offset of 2 was mentioned in the commit message but not well commented in the
source.

Add descriptive comments of the phase offset with and without ping-pong
buffers enabled.

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-25 22:59:07 +08:00
Ben Gardiner 52e2c5d38e ASoC: davinci-pcm: convert to BATCH mode
The davinci-pcm driver's snd_pcm_ops pointer function currently calls into
the edma controller driver to read the current positions of the edma channels
to determine pos to return to the ALSA framework. In particular,
davinci_pcm_pointer() calls edma_get_position() and the latter has a comment
indicating that "Its channel should not be active when this is called" whereas
the channel is surely active when snd_pcm_ops.pointer is called.

The operation of davinci-pcm in capture and playback appears to follow close
the other pcm drivers who export SNDRV_PCM_INFO_BATCH except that davinci-pcm
does not report it's positions from pointer() using the last transferred
chunk. Instead it peeks directly into the edma controller to determine the
current position as discussed above.

Convert the davinci-pcm driver to BATCH mode: count the periods elapsed in the
prtd->period member and use its value to report the 'pos' to the alsa
framework in the davinci_pcm_pointer function.

There is a phase offset of 2 periods between the position used by dma setup
and the position reported in the pointer function. Either +2 in the dma
setup or -2 in the pointer function (with wrapping, both) accounts for this
offset -- I opted for the latter since it makes the first-time setup clearer.

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-25 19:14:31 +08:00
Ben Gardiner 10ab3bfda4 ASoC: davinci-pcm: extract period elapsed functions
Extract functions that modify the prtd->period member in preparation for
conversion to BATCH mode playback.

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-25 19:14:31 +08:00
Ben Gardiner ef39eb6f21 ASoC: davinci-pcm: fix audible glitch on 2nd ping-pong playback
The release of the dma channels was being performed in prepare and there was a
edma_resume call for the asp-channel only being executed on START, RESUME and
PAUSE_RELEASE.

The mcasp on da850evm with ping-pong buffers enabled was exhibiting an audible
glitch on every playback after the first. It was determined through trial and
error that the following two changes fix this problem:

1) Move the edma_start calls from prepare to trigger and 2) reverse the order
of starting the asp and ram channels.

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-25 19:14:30 +08:00
Ben Gardiner acb8e2666e ASoC: davinci-pcm: increase the maximum channels
Based on the registration of davinci-mcasp.1 in the davinci-evm platform
setup for da830 and dm6467, davinci-pcm can handle more than the currently
reported maximum channels of 2.

Increase the maximum channels to 384 to match the maximum reported by
davinci-mcasp.1.

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-25 19:14:29 +08:00
Ben Gardiner 8e56d5b834 ASoC: davinci-pcm: expand the .formats
Based on the data_type test in ping_pong_dma_setup, davinci-pcm is capable of
handling data of width up to and including 32bits.

"
	if ((data_type == 0) || (data_type > 4)) {
		printk(KERN_ERR "%s: data_type=%i\n", __func__, data_type);
		return -EINVAL;
	}
"

Update the .format member of the snd_pcm_hardware instances it registers to
reflect this capability.

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-25 19:14:29 +08:00
Ben Gardiner fb1e9703af ASoC: davinci-pcm: trivial: make ping-pong params setup symmetrical
The setup of the pong channel uses EDMA_CHAN_SLOT instead of & 0x3f as the
setup of the ping channel does.

Make the setup of ping and pong symmetric. There is no functional change
introduced by this patch.

Signed-off-by: Ben Gardiner <bengardiner@nanometrics.ca>
Reviewed-by: Steven Faludi <stevenfaludi@nanometrics.ca>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-25 19:14:28 +08:00
Daniel Mack d5a0bf6cc5 ALSA: usb-audio: more control quirks for M-Audio FastTrack devices
Make use of the freshly introduced methods to re-use standard mixer
handling and add some controls that are hidden but implemented in a
standard conform way on M-Audio's FastTrack devices.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Original-code-by: Felix Homann <linuxaudio@showlabor.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 09:37:08 +02:00
Daniel Mack 014950b013 ALSA: usb-audio: add new quirk type QUIRK_AUDIO_STANDARD_MIXER
This quirk type will let the driver assume that there is a standard
mixer on a given interface, or that a specific mixer quirks will handle
the device.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 09:36:59 +02:00
Daniel Mack 9e38658f70 ALSA: usb-audio: export snd_usb_feature_unit_ctl
In order to allow quirks functions to hook up to the standard feature
unit op tables, this patch exports a pointer to the struct that is used
internally.

That way, all the code handling the control can be kept private, and
external code can reference the symbol to re-use it.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 09:36:20 +02:00
Daniel Mack ef9d597089 ALSA: usb-audio: rework add_control_to_empty()
This patch renames add_control_to_empty() to snd_usb_mixer_add_control()
and exports it, so the quirks functions can make use of it.

Also, as "struct mixer_build" is private to mixer.c, rewrite the
function to take an argument of type "struct usb_mixer_interface"
instead.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 09:34:34 +02:00
Daniel Mack 5875c2cb76 ALSA: usb-audio: move assignment of chip->ctrl_intf
This is needed for upcoming changes to the quirks mechanism.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 09:34:19 +02:00
Takashi Iwai af4ccf4f86 ALSA: hda - Use model=auto for Lenovo G555
The new auto-parser fixes problems on Lenovo G555.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 07:33:20 +02:00
Stephen Warren 5d44f927a5 ALSA: HDA: Unify HDMI hotplug handling.
This change unifies the initial handling of a pin's state with the code to
update a pin's state after a hotplug (unsolicited response) event. The
initial probing, and all updates, are now routed through hdmi_present_sense.

The stored PD and ELDV status is now always derived from GetPinSense verb
execution, and not from the data in the unsolicited response. This means:

a) The WAR for NVIDIA codec's UR.PD values ("old_pin_detect") can be
   removed, since this only affected the no-longer-used unsolicited
   response payload.

b) In turn, this means that most NVIDIA codecs can simply use
   patch_generic_hdmi instead of having a custom variant just to set
   old_pin_detect.

c) When PD && ELDV becomes true, no extra verbs are executed, because the
   GetPinSense that was previously executed by snd_hdmi_get_eld (really,
   hdmi_eld_valid) has simply moved into hdmi_present_sense.

d) When PD && ELDV becomes false, there is a single extra GetPinSense verb
   executed for codecs where old_pin_detect wasn't set, i.e. some NVIDIA,
   and all ATI/AMD and Intel codecs. I doubt this will be a performance
   issue.

The new unified code in hdmi_present_sense also ensures that eld->eld_valid
is not set unless eld->monitor_present is also set. This protects against
potential invalid combinations of PD and ELDV received from HW, and
transitively from a graphics driver.

Also, print the derived PD/ELDV bits from hdmi_present_sense so the kernel
log always displays the actual state stored, which will differ from the
values in the unsolicited response for NVIDIA HW where old_pin_detect was
previously set.

Finally, a couple of small tweaks originally by Takashi:

* Clear the ELD content to zero before reading it, so that if it's not
  read (i.e. when !(PD && ELDV)) it's in a known state.

* Don't show ELD fields in /proc ELD files when the ELD isn't valid.

The only possibility I can see for regression here is a codec where the
GetPinSense verb returns incorrect data. However, we're already exposed
to that, since that data is used (a) from hdmi_add_pin to set up the
initial pin state, and (b) within snd_hda_input_jack_report to query
a pin's presence value. As such, I don't believe any HW has bugs here.

Includes-changes-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Stephen Warren <swarren@nvidia.com>
Acked-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 07:31:32 +02:00
Tony Vroon 4e60b4f830 ALSA: hda - Force AD1988_6STACK_DIG for Asus M3N-HT Deluxe
The microphone input on the back panel (pink connector)
stopped operating correctly after an upgrade from
2.6.35 to 2.6.38; the actual problem manifests itself
as a lack of microphone bias voltage (VREF_HIZ) on
node 0x17.
With AD1988_6STACK_DIG the maximum bias voltage (VREF_80)
is applied and the headset operates correctly.

Signed-off-by: Tony Vroon <tony@linx.net>
Tested-by: Doug Redlich <pbrigade@nxltech.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-25 07:30:39 +02:00
Liam Girdwood 92505299a1 ASoC: core - remove superfluous new line.
Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-25 04:45:47 +08:00
Linus Torvalds f50d1d9e8d Merge git://git.kernel.org/pub/scm/linux/kernel/git/brodo/pcmcia-2.6
* git://git.kernel.org/pub/scm/linux/kernel/git/brodo/pcmcia-2.6:
  pcmcia: Make struct pcmcia_device_id const, sound drivers edition
  staging: pcmcia: Convert pcmcia_device_id declarations to const
  pcmcia: Convert pcmcia_device_id declarations to const
  pcmcia: Make declaration and uses of struct pcmcia_device_id const
  pcmcia/sa1100: put sa11x0_pcmcia_hw_init[] to .devinit.data
2011-05-24 13:28:35 -07:00
Liam Girdwood 61b61e3c5c ASoC: core - fix module reference counting for CPU DAIs
Currently CODEC and platform drivers have their module reference count
incremented soc_probe_dai_link() whilst CPU DAI drivers have their reference
count incremented in soc_bind_dai_link().

CPU DAIs should have their reference count incremented in soc_probe_dai_link()
just like the CODEC and platform drivers.

Signed-off-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-24 23:25:34 +08:00
Daniel Mack 477a66948e ASoC: fix raumfeld platform
Commit f0fba2ad (ASoC: multi-component - ASoC Multi-Component Support)
broke support for Raumfeld platforms as it didn't take into account the
different hardware features on individual devices.

In particular, Raumfeld speakers have no S/PDIF output, so the members
of the snd_soc_card struct must be set dynamically.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org
2011-05-24 23:25:00 +08:00
Kuninori Morimoto 23ca853392 ASoC: sh: fsi: add fsi_hw_startup/shutdown
This patch is preparation of cleanup suspend/resume patch.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-24 18:42:07 +08:00
Kuninori Morimoto cda828cafe ASoC: sh: fsi: cleanup suspend/resume
Current FSI driver was using saved_xxx variable for suspend/resume.
OTOH, the start and stop of power/clock are controlled by
fsi_hw_startup/fsi_hw_shutdown in current FSI driver.
The all necessary registers value are set by fsi_hw_startup.

So, if fsi_hw_shutdown is called when "suspend" is generated,
and fsi_hw_startup is called at "resume",
the saved_xxx are not needed.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-24 18:42:02 +08:00
Kuninori Morimoto 4c48125331 ASoC: sh: fsi: remove fsi_module_init/kill
FSIA/B ports is enabled by default when power-on,
and current FSI is supporting RuntimePM.
In addition, current fsi_module_init/kill doesn't care
simultaneous playback/recorde.
This mean FSI port control is not needed.
This patch remove fsi_module_init/kill

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-24 18:41:57 +08:00
Kuninori Morimoto 2da658927c ASoC: sh: fsi: make sure fsi_stream_push/pop access by spin lock
fsi_stream_push/pop might be called in same time.
This patch protect it.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-24 18:41:52 +08:00
Kuninori Morimoto 9478e0b60f ASoC: sh: fsi: remove pm_runtime from fsi_dai_set_fmt.
pm_runtime_get/put_sync were used to access FSI register in fsi_dai_set_fmt
which is called when ALSA probe.
But this register value will disappear after pm_runtime_put_sync
if platform is supporting RuntimePM.
To solve this issue, this patch adds new variable for format,
and remove pm_runtime_get/put_sync from fsi_dai_set_fmt.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-24 18:41:48 +08:00
Kuninori Morimoto 2e651bafa9 ASoC: sh: fsi: tidyup unclear variable naming
Some variables on this driver were a unclear naming,
and were different unit (byte, frame, sample).
And some functions had wrong name
(ex. it returned "sample width" but name was "fsi_get_frame_width").
This patch tidy-up this issue, and the minimum unit become "sample".
Special thanks to Takashi YOSHII.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-24 18:41:44 +08:00
Kuninori Morimoto 1ddddd3635 ASoC: sh: fsi: irq control moves to fsi_port_start/stop
Using fsi_irq_enable/disable in fsi_port_start/stop is very natural.
This patch is preparation of cleanup suspend/resume patch.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-24 18:41:40 +08:00
Kuninori Morimoto 4f56cde17e ASoC: sh: fsi: add fsi_set_master_clk
Current FSI driver is using set_rate call back function which is for
master mode.
By this patch, it is used from fsi_set_master_clk.
This patch is preparation of cleanup suspend/resume patch.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-24 18:41:36 +08:00
Kuninori Morimoto 0ffe296add ASoC: sh: fsi: tidyup parameter of fsi_stream_push
It is possible to create buff_len and period_len
from substream->runtime.
This patch is preparation of tidyup unclear variable naming patch.

Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2011-05-24 18:41:32 +08:00
Mark Brown 60c655e62f ASoC: Convert 16x16 write to use cpu_to_be16()
Make it clear what we're doing.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: Liam Girdwood <lrg@ti.com>
2011-05-24 18:41:09 +08:00
Takashi Iwai e2df82ffb8 ALSA: hda - Fix speaker auto-mute in Cxt auto-parser
Fix some logic failures in auto-mute handling in Conexant auto-parser.
Also, modify codes to be a bit more understandable.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-24 12:15:53 +02:00
Tony Lindgren 9b28b11e2a Merge branch 'for_2.6.40/pm-cleanup' of ssh://master.kernel.org/pub/scm/linux/kernel/git/khilman/linux-omap-pm into omap-for-linus 2011-05-24 00:45:06 -07:00
Linus Torvalds 99dff58562 Merge branch 'tty-next' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/tty-2.6
* 'tty-next' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/tty-2.6: (48 commits)
  serial: 8250_pci: add support for Cronyx Omega PCI multiserial board.
  tty/serial: Fix break handling for PORT_TEGRA
  tty/serial: Add explicit PORT_TEGRA type
  n_tracerouter and n_tracesink ldisc additions.
  Intel PTI implementaiton of MIPI 1149.7.
  Kernel documentation for the PTI feature.
  export kernel call get_task_comm().
  tty: Remove to support serial for S5P6442
  pch_phub: Support new device ML7223
  8250_pci: Add support for the Digi/IBM PCIe 2-port Adapter
  ASoC: Update cx20442 for TTY API change
  pch_uart: Support new device ML7223 IOH
  parport: Use request_muxed_region for IT87 probe and lock
  tty/serial: add support for Xilinx PS UART
  n_gsm: Use print_hex_dump_bytes
  drivers/tty/moxa.c: Put correct tty value
  TTY: tty_io, annotate locking functions
  TTY: serial_core, remove superfluous set_task_state
  TTY: serial_core, remove invalid test
  Char: moxa, fix locking in moxa_write
  ...

Fix up trivial conflicts in drivers/bluetooth/hci_ldisc.c and
drivers/tty/serial/Makefile.

I did the hci_ldisc thing as an evil merge, cleaning things up.
2011-05-23 12:23:20 -07:00
Takashi Iwai 313d2c0652 ALSA: hda - Fix initial capture-source with auto-mic for Cxt auto-parser
Fix the initialization of capture-source route when auto-mic is enabled
for Conexant auto-parser.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-23 20:27:02 +02:00
Takashi Iwai 506a4196d4 ALSA: hda - Fix auto-mic detection in Conexant codec-parser
Fix the auto-mic detection for Cxt auto-parser due to off-by-one
missing initialization.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2011-05-23 20:07:15 +02:00
Linus Torvalds 710421cc7d 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: (346 commits)
  ASoC: core: Don't set "(null)" as a driver name
  ALSA: hda - Use LPIB for ATI/AMD chipsets as default
  Revert "ALSA: hda - Use position_fix=3 as default for AMD chipsets"
  ASoC: Tegra: Fix compile when debugfs not enabled
  ASoC: spdif-dit: Add missing MODULE_*
  SOUND: OSS: Remove Au1550 driver.
  ALSA: hda - add Intel Panther Point HDMI codec id
  ALSA: emu10k1 - Add dB range to Bass and Treble for SB Live!
  ALSA: hda - Remove PCM mixer elements from Virtual Master of realtek
  ALSA: hda - Fix input-src parse in patch_analog.c
  ASoC: davinci-mcasp: enable ping-pong SRAM buffers
  ASoC: add iPAQ hx4700 machine driver
  ASoC: Asahi Kasei AK4641 codec driver
  ALSA: hda - Enable Realtek ALC269 codec input layer beep
  ALSA: intel8x0m: enable AMD8111 modem
  ALSA: HDA: Add jack detection for HDMI
  ALSA: sound, core, pcm_lib: fix xrun_log
  ASoC: Max98095: Move existing NULL check before pointer dereference.
  ALSA: sound, core, pcm_lib: xrun_log: log also in_interrupt
  ALSA: usb-audio - Add support for USB X-Fi S51 Pro
  ...
2011-05-23 08:52:38 -07:00