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2010-01-26ASoC: DaVinci: Fix stream restart errorChaithrika U S
Sometimes after a suspend-resume cycle, the ALSA application restarts the stream when resume fails and McASP fails to work as the clock is not enabled. This patch corrects this bug. Testes on TI DA850/OMAP-L138 EVM. Signed-off-by: Chaithrika U S <chaithrika@ti.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-26ALSA: hda - Add support for more the 8 streamsWei Ni
In azx_stream_start() and azx_stream_stop(), it use azx_readb/azx_writeb to read/write SIE, it just enable/disable 8 streams. But according to the HDA spec, it support 30 streams, and the new HDA controller will support more then 8 streams. So we should use azx_readl/azx_writel to read/write SIE. Signed-off-by: Wei Ni <wni@nvidia.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-26ALSA: cs46xx: Fix cpu idling with resumeFlorian Zumbiehl
Make sure that capture DMA doesn't stay enabled after system resume as that potentially prevents the processor from entering deep sleep states. Signed-off-by: Florian Zumbiehl <florz@florz.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-25Merge branch 'fix/hda' into for-linusTakashi Iwai
2010-01-25Merge branch 'for-2.6.33' into for-2.6.34Mark Brown
2010-01-25ASoC: ad1836: reset and restore clock control mode in suspend/resume entryBarry Song
tests show frequent suspend/resume(frequent poweroff/on ad1836 internal components) maybe make ad1836 clock mode wrong sometimes after wakeup. This patch reset/restore ad1836 clock mode while executing PM, then ad1836 can always resume to right clock status. Signed-off-by: Barry Song <21cnbao@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-25ASoC: add DAI and platform / DMA drivers for SH SIUGuennadi Liakhovetski
Several SuperH platforms, including sh7722, sh7343, sh7354, sh7367 include a Sound Interface Unit (SIU). This patch adds DAI and platform / DMA drivers for this interface. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-25ALSA: hda - Remove the COEF setup for ALC267/ALC268Takashi Iwai
The COEF setup for model=auto seems problematic on some laptops, resulting in the silent speaker output. Better to disable it for now. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-25ALSA: hda - Remove coef output in Realtek proc filesTakashi Iwai
The output of COEF index/value in the proc file for Realtek codecs is rather useless since the value varies together with the index. Let's get rid of it again. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-25ASoC: fix a memory-leak in wm8903Guennadi Liakhovetski
Remember to free the temporary register-cache. Signed-off-by: Guennadi Liakhovetski <g.liakhovetski@gmx.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
2010-01-25ALSA: hda - add possibility to choose speakers configuration for 4930gŁukasz Wojniłowicz
Now one can choose speaker configuration in e.g. PulseAudio mixer Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-24ALSA: hda - Change headphone pin control with master volume on cx5051Takashi Iwai
The HP pin (0x16) control has to be changed dynamically depending on the master volume switch as well as the speaker pin (0x1a). Otherwise the headphone still sounds with master off. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-24ALSA: hda - Fix SPDIF output widget for Cxt5051 codecTakashi Iwai
Fixed the wrongly set up for SPDIF output on Conexant 5051 codec. It must point to the audio out widget instead of a pin. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-24ALSA: hda - initialize mic port on cxt5051 codec dynamicallyTakashi Iwai
Initialize the mic ports B & C on Conexant 5051 codec dynamically according to the mic jack detection, instead of static init arrays. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-24ALSA: hda - Merge playback controls for Cx5051 codec modelsTakashi Iwai
All cx5051 codec models have the same Master playback mixer definitions. Merge them together. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-24ALSA: hda - Add support for Toshiba Satellite M300Takashi Iwai
Added the support for Toshiba Satellite M300 with Conexant 5051 codec. Since the laptop has no port C connection and the pin reports always the jack sense true, we need to ignore port-C unsol event. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-23ALSA: hda - Fix HP dv6736 capture mixer nameTakashi Iwai
Use the standard "Capture" mixer name for HP dv6736 with Cxt5051 codec. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-23ALSA: hda - Minor fixes for Compaq Presario F700 quirkTakashi Iwai
Minor fixes for HP Compaq Presario F700 quirks with Cxt5051 codec: - changed the capture mixer elements to the standard name. - fixed the quirk name string without a space - sorted the quirk list - updated the documentation Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-21Merge branch 'topic/noncached-mmap' into topic/miscTakashi Iwai
2010-01-21ALSA: hda - AD1988 codec - fix SPDIF-input mixer initialization (unmute)Jaroslav Kysela
The SPDIF-input pin 0x1c is muted by default in hardware. Unmute appropriate pin to get captured samples instead zeros. Tested on Lenovo Thinkstation. Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-01-21Merge remote branch 'alsa/devel' into topic/miscTakashi Iwai
2010-01-21ASoC: Use BIAS_OFF when idle for wm_hubs devicesMark Brown
This provides a small power saving when audio is inactive. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-01-21ASoC: Support turning off bias when the CODEC is idleMark Brown
Currently ASoC always maintains the bias of the CODEC while the system is active. With older mobile CODECs this is required since the outputs are referenced to a non-zero voltage and enabling or disabling this voltage without audible pops or clicks in the output takes too long to do when starting or stopping audio. As a result of features such as ground referenced outputs and class D speaker drivers current generation devices are able to power on and off much more quickly without these system level issues so provide a new flag idle_bias_off in snd_soc_codec which will cause the core to turn off the CODEC bias. The distinction between STANDBY and OFF is still maintained. This is partly for consistency but also allows for potential future extensions such as per-machine overrides or deferring the bias removal. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-01-21ASoC: Remove console DAPM debug codeMark Brown
The same information is now visible via debugfs and with large modern devices dumping everything to the console can be very resource intensive, causing more harm than good. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
2010-01-21ALSA: pcm_core: Fix wake_up() optimizationJaroslav Kysela
This change fixes the "ALSA: pcm_lib - optimize wake_up() calls for PCM I/O" commit. New sleeping queue is introduced to separate user space and kernel space wake_ups. runtime->nowake is renamed to twake (transfer wake). Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-01-20ASoC: tlv320dac33: Burst mode BCLK divider configurationPeter Ujfalusi
Add possibility to configure the burst mode BCLK divider through platform data structure. The BCLK divider changes the actual speed of the serial bus in burst mode, which is faster than the sampling frequency of the running stream. In this way platforms can experiment with the optimal burst speed without the need to modify the codec driver itself. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-20ASoC: tlv320dac33: BCLK divider fixPeter Ujfalusi
The BCLK divider was not configured in case of mode7. This leads to unpredictable behavior when switching between FIFO modes. Configure the BCLK divider depending on the fifo_mode (FIFO is in use, or FIFO bypass). Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-20ALSA: hda - Fix HP T5735 automuteTakashi Iwai
This patch fixes the aut-mute setup on HP T5735 with ALC262 codec. Instead of wrong amp, use pin control toggling for muting the speaker now. Tested-by: Lee Trager <lee.trager@hp.com> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-19Merge branch 'fix/hda' into topic/hdaTakashi Iwai
Conflicts: sound/pci/hda/patch_realtek.c
2010-01-19ALSA: hda - Turn on EAPD only if available for Realtek codecsTakashi Iwai
Some codecs disable widgets used for output pins and reserve as vendor- spec widgets. Thus we need to check the widget type and pin cap before actually sending SET_EAPD verbs in the auto-configuration mode. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-19ALSA: hda - Fix parsing pin node 0x21 on ALC259Takashi Iwai
ALC259 has a widget NID 0x21 for the output pin, but it wasn't handled properly in alc268_new_analog_output(). Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-19ASoC: tlv320dac33: Correct the prefill number of samplesPeter Ujfalusi
Set the prefill number of samples as the same as the lower threshold in mode7. In this way the codec will read the same amount of data on startup and during the running playback. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-18Merge remote branch 'alsa/devel' into topic/miscTakashi Iwai
2010-01-18sound: virtuoso: add Xonar DS supportClemens Ladisch
Add experimental support for the Asus Xonar DS. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-01-18sound: seq_timer: simplify snd_seq_timer_set_tick_resolution() parametersClemens Ladisch
As snd_seq_timer_set_tick_resolution() is always called with the same three fields of struct snd_seq_timer, it suffices to give that as the only parameter. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Jaroslav Kysela <perex@perex.cz>
2010-01-18ALSA: pcm - Call pgprot_noncached() for vmalloc'ed buffersTakashi Iwai
pgprot_noncached() can be set for vmalloc'ed buffers safely, and we'd need non-cached behavior more or less, even for the intermediate ring- buffers. Now snd_pcm_lib_mmap_vmalloc() is added as the common PCM mmap callback that is coupled with snd_pcm_lib_alloc_vmalloc_buffer() & co. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-18ALSA: pcm - Remove unneeded ifdef pgprot_noncachedTakashi Iwai
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-18Merge branch 'fix/hda' into for-linusTakashi Iwai
2010-01-18ALSA: Remove warning message for invalid OSS minor rangesTakashi Iwai
When a card instance with a higher card number is registered, warning messages are spewed eventually with stack traces due to the invalid minor number for OSS device registration. For example, thinkpad-acpi registers the card number 29 as default, and you'll see always these messages. This is rather confusing (and worries users), thus better to return simply the error code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-17Merge branch 'mxc-audio' into for-2.6.34Mark Brown
Conflicts: arch/arm/plat-mxc/Makefile (dual add) sound/soc/imx/mx27vis_wm8974.c (API updates & removal)
2010-01-17ASoC: Mark new i.MX drivers as BROKEN until arch/arm mergedMark Brown
Currently they don't build due to cross tree dependencies, they will be reenabled once the arch/arm side has merged. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-17ALSA: hda - Fix capture on Sony VAIO with single inputTakashi Iwai
Sony VAIO VGN-P11G with ALC262 codec has only one input pin, and the recording doesn't work with model=auto because ALC262 parser sets the wrong cap NIDs to choose the route and the default route for the sole input pin wasn't initialized properly. This patch solves these issues. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2010-01-17ASoC: Remove old i.MX driver codeMark Brown
This has been superceeded by Sascha's new driver but was not removed in the patch series due to cutdowns for review. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-17ASoC: i.MX SSI driver does not yet support master modeMark Brown
The clocks for the SSI block need handling before this can work. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-17ASoC: Convert new i.MX SSI driver to use static DAI arrayMark Brown
While dynamically allocated DAIs are the way forward the core doesn't yet support anything except matching with a pointer to the actual DAI so convert to doing that so that machine drivers don't have to jump through hoops to register themselves. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Javier Martin <javier.martin@vista-silicon.com>
2010-01-17ASoC: Fix i.MX audio build for i.MX3xMark Brown
Don't unconditionally include the i.MX2x DMA driver, the arch/arm functions it uses aren't available for i.MX3x. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Javier Martin <javier.martin@vista-silicon.com>
2010-01-17ASoC: Add a new imx-ssi sound driverSascha Hauer
The old driver has the number of SSI units in the system hardcoded, does not make use of the device model and works only on i.MX21/27. This driver replaces it. It works in DMA mode on i.MX21/27 and using an FIQ handler on other systems. It also supports AC97 mode of the SSI units. Signed-off-by: Sascha Hauer <s.hauer@pengutronix.de> Acked-by: Javier Martin <javier.martin@vista-silicon.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-15ASoC: support more sample rates on raumfeld devicesDaniel Mack
Add support for sample rates other than 44100Khz on raumfeld audio devices. At startup time, call snd_soc_dai_set_sysclk() with 0 as 'freq' argument so it offers all the sample rates. Later, the function is called again to give proper constraints. Use the external audio clock generator to provide double data rate clocks as the PXA's internal baud generator does anything but what's described in the datasheets. Signed-off-by: Daniel Mack <daniel@caiaq.de> Cc: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-15ASoC: cs4270: allow passing freq=0 in set_dai_sysclk()Daniel Mack
For setups with variable MCLKs, the current logic of limiting the available sampling rates at startup time is not sufficient. We need to be able to change the setting at a later point, and so the codec must offer all possible rates until the hw_params are given. This patches allows that by passing 0 as 'freq' argument to cs4270_set_dai_sysclk(). Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2010-01-15ALSA: hda - Fix mute led GPIO on HP dv-series notebooksKunal Gangakhedkar
On my laptop (HP dv6-1110ax), there are no OEM strings in SMBIOS of type "HP_Mute_LED*". Hence, the GPIO for the mute button LED doesn't get set properly. I didn't find the strings in my cousin's laptop (HP dv9500t CTO) either. As per the documentation of find_mute_led_gpio(), these strings occur in HP B-series systems - so, before scanning the SMBIOS strings, we need to check if we're dealing with a B-series system. Need to get confirmation from HP if this logic takes care of all the systems. I'm trying to poke a friend there. Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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