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2008-12-25Merge branch 'topic/hda' into to-pushTakashi Iwai
2008-12-25Merge branch 'topic/cs5535audio' into to-pushTakashi Iwai
2008-12-25Merge branch 'topic/convert-tasklet' into to-pushTakashi Iwai
2008-12-25Merge branch 'topic/ca0106' into to-pushTakashi Iwai
2008-12-25Merge branch 'topic/audigy-capture-boost' into to-pushTakashi Iwai
2008-12-25Merge branch 'topic/asoc' into to-pushTakashi Iwai
2008-12-25Merge branch 'topic/aoa' into to-pushTakashi Iwai
2008-12-24Merge branch 'fix/hda' into topic/hdaTakashi Iwai
2008-12-24ALSA: hda - Add missing terminators in patch_sigmatel.cHerton Ronaldo Krzesinski
Signed-off-by: Herton Ronaldo Krzesinski <herton@mandriva.com.br> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-22ALSA: ASoC: fix a typo in omp-pcm.cRoel Kluin
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>
2008-12-22ASoC: Fix DSP formats in SSM2602 audio codecJarkko Nikula
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>
2008-12-22ASoC: Fix incorrect DSP format in OMAP McBSP DAI and affected driversJarkko Nikula
- 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>
2008-12-20ALSA: hda: fix incorrect mixer index values for 92hd83xxMatthew Ranostay
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>
2008-12-20ALSA: hda: dinput_mux checkMatthew Ranostay
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>
2008-12-20Merge branch 'topic/ca0106-spdif-stream' into topic/ca0106Takashi Iwai
2008-12-20Merge branch 'topic/ca0106-resume' into topic/ca0106Takashi Iwai
2008-12-20Merge branch 'topic/ca0106-capture-no-44khz' into topic/ca0106Takashi Iwai
2008-12-20Merge branch 'topic/hda-resume-fix' into topic/hdaTakashi Iwai
2008-12-20Merge branch 'topic/pcsp-fix' into topic/miscTakashi Iwai
2008-12-20ALSA: hda - Add quirk for another HP dv7Takashi Iwai
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>
2008-12-20ALSA: ASoC - Add missing __devexit annotation to wm8350.cTakashi Iwai
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>
2008-12-20ALSA: ASoc: DaVinci: davinci-evm use dsp_b modeTroy Kisky
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>
2008-12-20ALSA: ASoC: DaVinci: i2s, evm, pass same value to codec and cpu_daiTroy Kisky
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>
2008-12-20ALSA: ASoC: tlv320aic3x add dsp_aTroy Kisky
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>
2008-12-20ALSA: ASoC: DaVinci: document I2S limitationsTroy Kisky
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>
2008-12-20ALSA: ASoC: DaVinci: davinci-i2s clean upTroy Kisky
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>
2008-12-20ALSA: ASoC: DaVinci: davinci-i2s clean upTroy Kisky
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>
2008-12-20ALSA: ASoC: DaVinci: davinci-i2s add comments to explain polarityTroy Kisky
Document the current polarity choices. Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2008-12-20ALSA: ASoC: DaVinci: davinvi-evm, make requests explicitTroy Kisky
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>
2008-12-20ALSA: ca0106 - disable 44.1kHz captureTakashi Iwai
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>
2008-12-20ALSA: ca0106 - Add missing card->private_data initializationTakashi Iwai
Added the missing card->private_data initialization that caused obvious problems at PM. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-20ALSA: ca0106 - Check ac97 availability at PMTakashi Iwai
Check the availability of ac97 at PM suspend/resume callbacks. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19ALSA: hda - Power up always when no jack detection is availableTakashi Iwai
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>
2008-12-19ALSA: hda - Fix unused variable warnings in patch_sigmatel.cTakashi Iwai
Fixed "unused varible" warnings in patch_sigmatel.c that have been introduced by the last changes. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19Merge branch 'topic/hda-stac-fix' into topic/hdaTakashi Iwai
2008-12-19Merge branch 'fix/asoc' into for-linusTakashi Iwai
2008-12-19Merge branch 'fix/asoc' into topic/asocTakashi Iwai
2008-12-19ALSA: Fix a Oops bug in omap soc driver.Stanley Miao
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>
2008-12-19ALSA: hda - Add probe_only optionTakashi Iwai
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>
2008-12-19ALSA: hda - Use more distinct name for a unique volume in STAC/IDTTakashi Iwai
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>
2008-12-19ALSA: hda - Rework on STAC/IDT auto-configuration codeTakashi Iwai
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>
2008-12-19Merge branch 'fix/hda' into topic/hdaTakashi Iwai
2008-12-19ALSA: hda - Remove non-working headphone control for Dell laptopsTakashi Iwai
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>
2008-12-19ALSA: ca0106 - Fix typo in resume codeTakashi Iwai
The register and channel_id pair were twisted in the pm code... Oh my. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19ALSA: ca0106 - Add IEC958 PCM Stream controlsTakashi Iwai
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>
2008-12-19ALSA: ca0106 - Don't override the values at resumeTakashi Iwai
Don't override some values in ca0106_init_chip() at resume. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19ALSA: hda - Use snd_hda_ctl_add() in patch_sigmatel.cTakashi Iwai
Fixed the call of snd_ctl_add() by replacing with snd_hda_ctl_add() so that this mixer element can be tracked for re-configuration. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19ALSA: hda - Add missing initializations of amp and verb cachesTakashi Iwai
The re-initializations of codec amp and verb caches are missing at reconfig, which may cause Oops occasionally. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2008-12-19Merge branch 'fix/hda' into topic/hdaTakashi Iwai
2008-12-19ALSA: hda - Add no-jd model for IDT 92HD73xxTakashi Iwai
Added the model without the jack-detection for some desktops that have really no jack-detection. The recent driver caused regressions regarding the sound output on such machines. Signed-off-by: Takashi Iwai <tiwai@suse.de>
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