Commit Graph

4236 Commits

Author SHA1 Message Date
Takashi Iwai
9649745c86 Merge branch 'topic/snd-hrtimer' into to-push 2008-12-25 11:40:32 +01:00
Takashi Iwai
a9c3c7e04b Merge branch 'topic/pcxhr-update' into to-push 2008-12-25 11:40:31 +01:00
Takashi Iwai
cc4910850f Merge branch 'topic/oxygen' into to-push 2008-12-25 11:40:30 +01:00
Takashi Iwai
a802269781 Merge branch 'topic/jack-mechanical' into to-push 2008-12-25 11:40:29 +01:00
Takashi Iwai
a65056205c Merge branch 'topic/hda' into to-push 2008-12-25 11:40:28 +01:00
Takashi Iwai
313769d9ed Merge branch 'topic/cs5535audio' into to-push 2008-12-25 11:40:28 +01:00
Takashi Iwai
8afabfa74b Merge branch 'topic/convert-tasklet' into to-push 2008-12-25 11:40:27 +01:00
Takashi Iwai
86b3aa390b Merge branch 'topic/ca0106' into to-push 2008-12-25 11:40:26 +01:00
Takashi Iwai
e4456e7161 Merge branch 'topic/audigy-capture-boost' into to-push 2008-12-25 11:40:26 +01:00
Takashi Iwai
5c8261e44e Merge branch 'topic/asoc' into to-push 2008-12-25 11:40:25 +01:00
Takashi Iwai
facef8685b Merge branch 'topic/aoa' into to-push 2008-12-25 11:40:24 +01:00
Takashi Iwai
7645c4bfbb Merge branch 'fix/hda' into topic/hda 2008-12-24 11:04:08 +01:00
Herton Ronaldo Krzesinski
574f3c4f5c ALSA: hda - Add missing terminators in patch_sigmatel.c
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-24 11:03:56 +01:00
Roel Kluin
472346da9c ALSA: ASoC: fix a typo in omp-pcm.c
Fix a typo (& and &&)

Signed-off-by: Roel Kluin <roel.kluin@gmail.com>
Signed-off-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-22 18:33:26 +01:00
Jarkko Nikula
c691348587 ASoC: Fix DSP formats in SSM2602 audio codec
Thanks to Troy Kisky <troy.kisky@boundarydevices.com> for noticing.

- DSP_A format has 1-bit data delay which corresponds to SSM6202 submode 2
- DSP_B has 0-bit data delay which corresponds to submode 1
- Currently driver sets them opposite so swap the submode setting

Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-22 16:23:22 +00:00
Jarkko Nikula
bd25867a6c ASoC: Fix incorrect DSP format in OMAP McBSP DAI and affected drivers
- OMAP McBSP DAI driver claims to support DSP_A format which has 1-bit data
  delay but configures link for 0-bit data delay which is in fact DSP_B
- Fix this by changing format from DSP_A to DSP_B
- Fix also TLV320AIC23 codec and OSK5912 machine drivers since the same
  error is populated also there

Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-22 16:15:20 +00:00
Matthew Ranostay
74b7ff48a9 ALSA: hda: fix incorrect mixer index values for 92hd83xx
Fixed incorrect mixer index values for 92hd83xx codec's audio
input mixer.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-20 23:49:45 +01:00
Matthew Ranostay
f8ccbf65af ALSA: hda: dinput_mux check
Add check to determine if dinput_mux is set by any of patch_stac*() functions,
otherwise a invalid pointer my be referenced causing gibberish to mixer values.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-20 23:48:44 +01:00
Takashi Iwai
ebef7cfc81 Merge branch 'topic/ca0106-spdif-stream' into topic/ca0106 2008-12-20 23:43:06 +01:00
Takashi Iwai
6bcdbd55b4 Merge branch 'topic/ca0106-resume' into topic/ca0106 2008-12-20 23:43:00 +01:00
Takashi Iwai
6a8436419d Merge branch 'topic/ca0106-capture-no-44khz' into topic/ca0106 2008-12-20 23:42:55 +01:00
Takashi Iwai
8326e32c1e Merge branch 'topic/hda-resume-fix' into topic/hda 2008-12-20 23:41:18 +01:00
Takashi Iwai
55fa518867 Merge branch 'topic/pcsp-fix' into topic/misc 2008-12-20 23:39:47 +01:00
Takashi Iwai
69dfaefee4 ALSA: hda - Add quirk for another HP dv7
Added the model=hp-m4 quirk for another HP dv7 (103c:30fc) with IDT
92HD71b* codec.

Reference: Novell bnc#461108
	https://bugzilla.novell.com/show_bug.cgi?id=461108

Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-20 16:57:50 +01:00
Takashi Iwai
a31501d104 ALSA: ASoC - Add missing __devexit annotation to wm8350.c
Added the missing __devexit annotation to wm8350_codec_remove():
  sound/soc/codecs/wm8350.c:1546: warning: 'wm8350_codec_remove' defined but not used

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-20 16:50:53 +01:00
Troy Kisky
d6f833965e ALSA: ASoc: DaVinci: davinci-evm use dsp_b mode
Sense DaVinci does not support true I2S mode and
we don't have to use the hack, use dsp_b mode instead

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-20 13:05:39 +00:00
Troy Kisky
9e031624d5 ALSA: ASoC: DaVinci: i2s, evm, pass same value to codec and cpu_dai
Fix the meaning of SND_SOC_DAIFMT_NB_NF to match that
used in the codec.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-20 13:05:39 +00:00
Troy Kisky
a24f4f6826 ALSA: ASoC: tlv320aic3x add dsp_a
Add SND_SOC_DAIFMT_DSP_A mode option.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-20 13:05:38 +00:00
Troy Kisky
07d8d9dca4 ALSA: ASoC: DaVinci: document I2S limitations
DaVinci does not support true I2S or right justified
mode so not all I2S codecs will work with it when the codec is
master. Document this limitation.

Add dsp_a, dsp_b mode options

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-20 13:05:38 +00:00
Troy Kisky
69ab820c86 ALSA: ASoC: DaVinci: davinci-i2s clean up
Minor, just move a block of code to make next patch clearer.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-20 13:05:38 +00:00
Troy Kisky
21903c1c9e ALSA: ASoC: DaVinci: davinci-i2s clean up
Just at little cleanup of davinci_i2s_set_dai_fmt

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-20 13:05:38 +00:00
Troy Kisky
664b4af859 ALSA: ASoC: DaVinci: davinci-i2s add comments to explain polarity
Document the current polarity choices.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-20 13:05:38 +00:00
Troy Kisky
1152a1959f ALSA: ASoC: DaVinci: davinvi-evm, make requests explicit
Add constants with a value of 0 to show more explicitly
what is being requested.

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-20 13:05:37 +00:00
Takashi Iwai
ff75427a7f ALSA: ca0106 - disable 44.1kHz capture
The capture with 44.1kHz on ca0106 seems to cause loud noises on
later playbacks, which doesn't support 44.1kHz.  A simple fix is to
disable 44.1kHz, as the "default" PCM with dsnoop is anyway only with
48kHz.

Reference: Novell bnc#447624
	https://bugzilla.novell.com/show_bug.cgi?id=447624

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-20 11:20:55 +01:00
Takashi Iwai
72077aa336 ALSA: ca0106 - Add missing card->private_data initialization
Added the missing card->private_data initialization that caused obvious
problems at PM.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-20 11:12:51 +01:00
Takashi Iwai
50232d62ca ALSA: ca0106 - Check ac97 availability at PM
Check the availability of ac97 at PM suspend/resume callbacks.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-20 09:42:09 +01:00
Takashi Iwai
eb63212868 ALSA: hda - Power up always when no jack detection is available
When no jack detection is available, the pins should be always
turned on since it can't be turned on/off dynamically via unsol
events.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19 16:41:06 +01:00
Takashi Iwai
9158923228 ALSA: hda - Fix unused variable warnings in patch_sigmatel.c
Fixed "unused varible" warnings in patch_sigmatel.c that have been
introduced by the last changes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19 15:59:40 +01:00
Takashi Iwai
6030634ac3 Merge branch 'topic/hda-stac-fix' into topic/hda 2008-12-19 15:43:24 +01:00
Takashi Iwai
5bd9c69649 Merge branch 'fix/asoc' into for-linus 2008-12-19 15:37:12 +01:00
Takashi Iwai
70043058a6 Merge branch 'fix/asoc' into topic/asoc 2008-12-19 15:36:58 +01:00
Stanley Miao
19b3f31609 ALSA: Fix a Oops bug in omap soc driver.
There will be a Oops or frequent underrun messages when playing music with
omap soc driver, this is because a data region is incorretly sized, other data
region will be overwriten when writing to this data region.

Signed-off-by: Stanley Miao <stanley.miao@windriver.com>
Acked-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19 15:36:35 +01:00
Takashi Iwai
d4d9cd0338 ALSA: hda - Add probe_only option
Added probe_only module option to hd-audio driver.
This option specifies whether the driver creates and initializes the
codec-parser after probing.  When this option is set, the driver skips
the codec parsing and initialization but gives you proc and other
accesses.  It's useful to see the initial codec state for debugging.

The default of this value is off, so the default behavior is as same
as before.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19 15:19:11 +01:00
Takashi Iwai
766245348d ALSA: hda - Use more distinct name for a unique volume in STAC/IDT
When the line_out has only one DAC and it's unique (i.e. not shared
by other outputs), assign a more reasonable and distinct mixer name
such as "Headphone" or "Speaker".

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19 15:09:38 +01:00
Takashi Iwai
c21ca4a872 ALSA: hda - Rework on STAC/IDT auto-configuration code
The current auto-configuration code has several problems especially
for the new IDT codecs, e.g. wrong assignment of pins and DACs or
coupled volume for speaker and headphone.

This patch is a fairly large rewrite of the auto-configuration code.
Some remaks

- mic_switch and line_switch contain NIDs instead of bool
- dac_list isn't fixed for IDT 92HD* codecs now, they are all probed
- extra HP and speakers are stored in extra_dacs[].

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19 15:09:36 +01:00
Takashi Iwai
03c6901ea2 Merge branch 'fix/hda' into topic/hda 2008-12-19 14:24:13 +01:00
Takashi Iwai
8f55c1e51f ALSA: hda - Remove non-working headphone control for Dell laptops
The previous commit re-enabled hp_nid setup for IDT92HD73*, but
it's unneeded indeed for Dell laptops that have multiple headphones.
Setting the extra hp_nid results in a non-working "Headpohne" mixer
control.  Thus hp_nid should be 0 for these dell models.

Also, the automatic addition of hp_nid should check whether it's
a dual-HP model or not.  For dual-HPs, the pins are already checked
by the early workaround.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19 14:23:08 +01:00
Takashi Iwai
8df0f70751 ALSA: ca0106 - Fix typo in resume code
The register and channel_id pair were twisted in the pm code...
Oh my.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19 13:55:17 +01:00
Takashi Iwai
3d4758299f ALSA: ca0106 - Add IEC958 PCM Stream controls
Added "IEC958 PCM Stream" controls for the per-stream IEC958 status
bits.  Using this instead of "IEC958 Default" is safer since the status
bits will be recovered to the default states after closing the PCM
stream.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19 12:13:18 +01:00
Takashi Iwai
86effd7e12 ALSA: ca0106 - Don't override the values at resume
Don't override some values in ca0106_init_chip() at resume.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19 12:04:06 +01:00