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* ASoC: sh: FSI: Add capture supportKuninori Morimoto2009-10-301-7/+86
| | | | | Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: sh: FSI: Remove DMA supportKuninori Morimoto2009-10-302-122/+20
| | | | | | | | | | SuperH FSI device have the hardware limitation to use DMA. If DMA is used, LCD output will be broken. Maybe there are some solution. But I don't know how to do it now. This patch remove DMA support and use soft transfer. Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Modifying Kconfig/Makefile for AM3517 EVMAnuj Aggarwal2009-10-292-0/+11
| | | | | | | | Modifying the Kconfig and Makefile in sound/soc/omap folder to add support for OMAP3517 / AM3517 EVM in Alsa SoC. Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Adding OMAP3517 / AM3517 EVM support in ASOCAnuj Aggarwal2009-10-291-0/+202
| | | | | | | Adding support for OMAP3517 / AM3517 EVM in Alsa SoC. Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: OMAP3EVM: Use the twl4030_setup_data for headset pop-removalAnuj Aggarwal2009-10-291-0/+7
| | | | | | | | | The pop-removal specific values are configured for TWL4030 codec for OMAP3EVM through this patch. Signed-off-by: Anuj Aggarwal <anuj.aggarwal@ti.com> Acked-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: TWL4030: Add APLL supply for the capture pathPeter Ujfalusi2009-10-291-0/+5
| | | | | | | | Capture path also need the APLL enabled, adding DAPM_SUPPLY for the Virtual ADCs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: TWL4030: Change APLL powering sequencePeter Ujfalusi2009-10-291-2/+24
| | | | | | | | | | | | | | | | | | | It seams that certain part of the twl4030 codec needs the APLL enabled before they are enabled. Paths which has any digital processing needs need the APLL enabled before they can function. For example the vibra output will have some random data after it is enabled and before the APLL also enabled. If only analog components are in use (analog bypass), than it seams, that the APLL does not need to be enabled. This lowers the power consumption with around ~0.005A. Adding DAPM_SUPPLY to the Digital playback route and also to the capture route to enable and disable the APLL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: TWL4030: Vibra motor stop fix when it is driven with audioJari Vanhala2009-10-291-2/+10
| | | | | | | | | | | This patch fixes vibrator playing incoherently, when driven with audio. There is something wrong in switch 3 at H-bridge and VIBRA_SET still affects PWM generator. Slowest value fixes things. Signed-off-by: Jari Vanhala <ext-jari.vanhala@nokia.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: CS4270: export de-emphasis filter as ALSA controlDaniel Mack2009-10-291-0/+1
| | | | | | | | | | | The CS4270 codec features an de-emphasis filter for compensation of audio material filtered by an 50/15 uS algorithm. Not sure whether we should always enable it for 44100Hz sampling frequency, but it should at least be configurable by the user. Signed-off-by: Daniel Mack <daniel@caiaq.de> Acked-by: Timur Tabi <timur@freescale.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Minor SMDK64xx WM8580 cleanupsMark Brown2009-10-291-10/+5
| | | | | | | | Fix up some comments, remove all enable_pin() calls (edge widgets are all enabled by default) and mark the microphone as disabled by default since it requires a resistor fit to connect it. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: TWL4030: Change codec_muted to apll_enabledPeter Ujfalusi2009-10-281-10/+10
| | | | | | | | codec_muted is missleading, change it to apll_enabled, which is what it is doing: enabing and disabling the APLL. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: TWL4030: Remove bypass trackingPeter Ujfalusi2009-10-281-98/+30
| | | | | | | | | | | | | | | | | | Since ASoC core now handling the codec bias differently there is no need to do the tracking of bypass switch states anymore. Handling of the common bit for analog loopbacks is done with DAPM_SUPPLY for the bypass paths. Now this bit is only enabled when there is a complete analog bypass path, compared to the previous implementation, when the global switch was enabled if there were any of the analog bypass switch was on (regardless if there were complete path or not) Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Add regulator support for WM8731Mark Brown2009-10-261-4/+47
| | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: TWL4030: Driver registration via twl4030_codec MFDPeter Ujfalusi2009-10-252-77/+127
| | | | | | | | | Change the way how the twl4030 soc codec driver is loaded/probed. Use the device probing via tlw4030_codec MFD device. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: TWL4030: use the twl4030-codec.h for register descriptionsPeter Ujfalusi2009-10-251-236/+6
| | | | | | | | | Remove the register descriptions from the twl4030.h file and use the linux/mfd/twl4030-codec.h instead, which has the codec related register descriptions also. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: OMAP: Don't try to set unsupported OMAP_DMA_DATA_BURST_16 on OMAP1Janusz Krzysztofik2009-10-221-2/+6
| | | | | | | | | | | | | | | | | | | | | | | After DMA burst mode has been introduced in sound/soc/omap/omap-pcm.c, omap_pcm_prepare() unconditionally calls: omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16); Current implementation of those two functions found in arch/arm/plat-ompa/dma.c doesn't support OMAP_DMA_DATA_BURST_16 on OMAP1 at all, so they both end with BUG() on that machine. That results in ASoC being completely unusable, at least on my OMAP5910 based Amstrad Delta. The patch corrects the problem by not calling those two functions when run on OMAP1 class based machines. Created against linux-2.6.32-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Jarkko Nikula <jhnikula@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: tlv320dac33: typo fix in the headerPeter Ujfalusi2009-10-211-1/+1
| | | | | | | | Fix the definition of DAC33_LTM field, the LTM bits in FIFO_IRQ_MODE_B register are starting at bit 6. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* ASoC: Amstrad Delta minor cleanupsJanusz Krzysztofik2009-10-211-2/+2
| | | | | | | | | Hi Mark, Here is a patch that corrects small omissions I have found in my code. Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-10-191-3/+8
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| * ASoC: Fix possible codec_dai->ops NULL pointer problemsBarry Song2009-10-191-3/+8
| | | | | | | | | | | | | | | | | | | | Some codec DAIs like stac9766, wm9712, wm9713, ad1980 don't register themselves then it loses to the chance to be given a null_dai_ops in snd_soc_register_dai if they have no ops. When functions like soc_pcm_open, soc_pcm_hw_params etc. access the ops field in these DAIs, panic will happen. Signed-off-by: Barry Song <21cnbao@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: Move dereference after NULL testJulia Lawall2009-10-191-1/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If the NULL test on jack is needed, then the derefernce should be after the NULL test. A simplified version of the semantic match that detects this problem is as follows (http://coccinelle.lip6.fr/): // <smpl> @match exists@ expression x, E; identifier fld; @@ * x->fld ... when != \(x = E\|&x\) * x == NULL // </smpl> Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: au1x: psc-ac97: reorganize timeoutsManuel Lauss2009-10-191-13/+25
| | | | | | | | | | | | | | | | Codec read/write functions: wait 21us between the pokings of hardware. Add timeouts to unbounded loops waiting for bits to change. Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: au1x: psc-ac97: verify correct codec register was readManuel Lauss2009-10-191-3/+8
| | | | | | | | | | | | | | Verify that the correct register has been received from the codec. Signed-off-by: Manuel Lauss <manuel.lauss@gmail.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: TWL4030: Only update the needed bits in *set_dai_sysclkPeter Ujfalusi2009-10-191-10/+12
| | | | | | | | | | | | | | | | Do not rewrite the whole register, but only update the needed bits in set_dai_sysclk functions. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-10-151-1/+1
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| * ASoC: Serialize access to dapm_power_widgets()Eero Nurkkala2009-10-131-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Access to damp_power_widgets() is assumed to be single-threaded. Concurrent accesses to dapm_power_widgets() may result in unpredictable behavior. Calls from: close_delayed_work() soc_codec_close() soc_pcm_prepare() soc_suspend() soc_resume_deferred() to snd_soc_dapm_stream_event() do not have the codec->mutex taken to cover the call to dapm_power_widgets(). Thus, take the mutex in these paths also to assure single-threaded use of dapm_power_widgets(). Signed-off-by: Eero Nurkkala <ext-eero.nurkkala@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: Codec driver for Texas Instruments tlv320dac33 codecPeter Ujfalusi2009-10-154-0/+1510
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo audio DAC. TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low power audio playback. The digital interface can use I2S, DSP (A or B), Right and Left justified formats. DAC33 has stereo analog input, which can be bypassed to the analog outputs. Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass' mode (default) and nSample mode (FIFO is in use). a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is working synchronously as a normal codec (it needs constant stream of data on the digital interface). b) The nSample mode implementation uses one interrupt line from DAC33 to the host: Alarm threshold is set to 10ms of audio data (limit by the driver implementation). DAC33 will signal an interrupt, when the FIFO level goes under the Alarm threshold. The host will write to nSample register a value (number of stereo samples), to tell DAC33 how many samples it should read in a burst from the host. When the DAC33 received the number of samples, it disables the clocks on the I2S bus. When the FIFO use again goes under the Alarm threshold, DAC33 signals the host with an interrupt, and the process is repeated. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: finally enable support for eXeda and CM-X300Igor Grinberg2009-10-151-1/+2
| | | | | | | | | | | | | | | | Signed-off-by: Igor Grinberg <grinberg@compulab.co.il> Signed-off-by: Mike Rapoport <mike@compulab.co.il> CC: Mark Brown <broonie@opensource.wolfsonmicro.com> CC: alsa-devel@alsa-project.org Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: Remove snd_soc_suspend_device()Mark Brown2009-10-1517-383/+0
| | | | | | | | | | | | | | | | The PM core will grow pm_link infrastructure in 2.6.33 which can be used to implement the intended functionality of the ASoC-specific device suspend and resume callbacks so drop them. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: S3C: Remove <plat/audio.h>Ben Dooks2009-10-137-7/+1
| | | | | | | | | | | | | | | | | | | | Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h as it provides nothing to the current kernel and is not in any future plans for the system. Signed-off-by: Ben Dooks <ben@simtec.co.uk> Signed-off-by: Simtec Linux Team <linux@simtec.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: TPA6130A2: Make tpa6130a2_power as staticPeter Ujfalusi2009-10-122-2/+1
| | | | | | | | | | | | | | | | The power for the amplifier should be handled internally by the tpa6130a2 driver. Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: Minor fixups to tpa6130a2 driverMark Brown2009-10-091-4/+4
| | | | | | | | | | | | | | | | - Staticise ttpa6130a2_client. - Remove unneeded cast from void. - Use explict NULL rather than 0. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: TPA6130A2 amplifier driverPeter Ujfalusi2009-10-094-0/+531
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Driver for Texas Instruments TPA6130A2 stereo headphone amplifier. The driver provides playback gain control and also pre-defined DAPM_HP widgets and DAPM routings for power management. The DAPM_HP widget names are: "TPA6130A2 Headphone Left" "TPA6130A2 Headphone Right" From soc machine drivers to use with the tpa6130a2 amplifier, the tpa6130a2_add_controls has to be called, which adds the alsa controls and the DAPM routing needed for the tpa6130a2. After that the machine driver can connect the codec's output with 'TPA6130A2 Left' and 'TPA6130A2 Right': {"TPA6130A2 Left", NULL, "CODEC LEFT OUT"}, {"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"}, Internally the left and right channels are powered separately. When none of the channels are needed the amplifier is powered down: hard power: valid GPIO number is passed within platform data soft power: Using the software shutdown of the amplifier Signed-off-by: Peter Ujfalusi <peter.ujfalusi@nokia.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | ASoC: at91sam9g20ek_2mmc board uses same audio connexion as at91sam9g20ekNicolas Ferre2009-10-091-1/+1
| | | | | | | | | | | | | | | | | | The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file to extend the machine ID checking. Signed-off-by: Nicolas Ferre <nicolas.ferre@atmel.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-10-082-10/+2
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| * Merge branch 'upstream/wm8350' into for-2.6.32Mark Brown2009-10-061-2/+2
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| | * ASoC: WM8350 capture PGA mutes are invertedMark Brown2009-10-061-2/+2
| | | | | | | | | | | | | | | Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| * | ASoC: Remove absent SYNC and TDM DAI format options from i.MX SSIMark Brown2009-10-061-8/+0
| | | | | | | | | | | | | | | | | | | | | These should be handled via set_tdm_slot() now and cause build failures as-is. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | Merge branch 'for-2.6.32' into for-2.6.33Mark Brown2009-10-06137-4867/+8136
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| * | Merge branch 'for-2.6.32' of ↵Mark Brown2009-10-062-3/+4
| |\ \ | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into for-2.6.32
| * \ \ Merge branch 'for-linus' of ↵Linus Torvalds2009-10-0323-256/+511
| |\ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'for-linus' of ssh://master.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (21 commits) ALSA: usb - Use strlcat() correctly ALSA: Fix invalid __exit in sound/mips/*.c ALSA: hda - Fix / improve ALC66x parser ALSA: ctxfi: Swapped SURROUND-SIDE mute sound: Make keywest_driver static ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-B1VP ALSA: hda - Fix digita/analog mic auto-switching with IDT codecs ASoC: fix kconfig order of Blackfin drivers ALSA: hda - Added quirk to enable sound on Toshiba NB200 ASoC: Fix dependency of CONFIG_SND_PXA2XX_SOC_IMOTE2 ALSA: Don't assume i2c device probing always succeeds ALSA: intel8x0 - Mute External Amplifier by default for Sony VAIO VGN-T350P ALSA: echoaudio - Re-enable the line-out control for the Mia card ALSA: hda - Resurrect input-source mixer of ALC268 model=acer ALSA: hda - Analog Devices AD1984A add HP Touchsmart model ALSA: hda - Add HP Pavilion dv4t-1300 to MSI whitelist ALSA: hda - CD-audio sound for hda-intel conexant benq laptop ASoC: DaVinci: Correct McASP FIFO initialization ASoC: Davinci: Fix race with cpu_dai->dma_data ASoC: DaVinci: Fix divide by zero error during 1st execution ...
| | * \ \ Merge branch 'fix/hda' into for-linusTakashi Iwai2009-10-035-94/+322
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| | | * | | ALSA: hda - Fix / improve ALC66x parserTakashi Iwai2009-10-021-86/+155
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The auto-parser for ALC662/663/272 codecs doesn't work properly when a speaker is connected to mono NID 0x17, and doesn't handle the dynamic DAC assignment properly. This patch fixes the issues and also improves the assignment of DACs so that HP and speakers can have independent volume controls. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: hda - Fix digita/analog mic auto-switching with IDT codecsTakashi Iwai2009-10-011-6/+14
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When the auto-mic switching between an analog and a digital mic is needed with IDT codecs, the current driver doesn't reset the connection of the digital mux. This patch fixes the behavior by checking both mux connections properly. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: hda - Added quirk to enable sound on Toshiba NB200Manoj Iyer2009-10-011-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Patch was tested on Toshiba NB200 and is found to enable sound. Signed-off-by: Manoj Iyer <manoj.iyer@canonical.com> Cc: stable@kernel.org Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: hda - Resurrect input-source mixer of ALC268 model=acerTakashi Iwai2009-09-301-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the commit fdbc66266c21976027938642f60e0f047149a61a, I mistakenly replaced the capture mixer array for ALC268_ACER to nosrc version although this should be kept to alt_mixer. Now fixed back. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: hda - Analog Devices AD1984A add HP Touchsmart modelMiguel de Barros2009-09-291-0/+139
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Reference: ALSA bug #0004614 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4614 port-A (0x11) - front hp-out port-D (0x12) - rear line out port-E (0x1c) - front mic-in port-F (0x16) - Internal speakers digital-mic (0x17) - Internal mic init verbs, mixers, jack sensing and PCI_QUIRK to support this hardware Signed-off-by: Miguel de Barros <miguel.de.barros@bluewin.ch> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: hda - Add HP Pavilion dv4t-1300 to MSI whitelistDaniel T Chen2009-09-241-0/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | BugLink: http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=547994 Enable MSI by default for this Pavilion model. Signed-off-by: Daniel T Chen <crimsun@ubuntu.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | | * | | ALSA: hda - CD-audio sound for hda-intel conexant benq laptopLukasz Marcinowski2009-09-241-1/+11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | After puting a cd-audio inside my laptop there was no sound out here, so I decided to install alsa-driver with debug level and setup a model=test, it didn't help, but then I look at source code and added this few lines, now cd-audio is working both when playback/recording. [Additional minor fixes of mixer element/item names by tiwai] Signed-off-by: Lukasz Marcinowski <nowymarluk@wp.pl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | | | Merge branch 'fix/asoc' into for-linusTakashi Iwai2009-10-0316-165/+155
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