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* ALSA: intel8x0m: append 'm' to "r_intel8x0"Paul Bolle2011-03-11
| | | | | | | Appending an 'm' will distinguish it from a similar struct in intel8x0.c Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: intel8x0m: add 'm' as "suffix" to static functionsPaul Bolle2011-03-11
| | | | | | | Adding an 'm' will distinguish them from identical names in intel8x0.c. Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: intel8x0m: wait a bit before warm reset checkPaul Bolle2011-03-11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | At every resume a laptop I use prints this message (at KERN_ERR level): ALSA sound/pci/intel8x0m.c:904: AC'97 warm reset still in progress? [0x2] The thing to note here is that 0x2 corresponds to ICH_AC97COLD. Ie, what seems to be happening is that the register involved indicated a warm reset for some time (as the ICH_AC97WARM bit was set) but by the time the warning is printed, and that same register is checked again, that bit is already cleared and only the ICH_AC97COLD bit is still set. It turns out a warm reset needs some time to settle, but it is currently checked right away. The test therefore fails the first time it is done and schedule_timeout_uninterruptible() will be called. Once we return from that jiffies is already (far) past end_time on this laptop, so we exit the loop, print a warning, and exit the function while the warm reset actually succeeded. A way to fix this is to call usleep_range() after writing to the register involved. A handful of tests suggest 500 usecs is a safe value. (This might punish the "finish cold reset" case, but on this laptop such a cold reset apparently never happens, so I can't say for sure.) While we're at it drop the extra single tick from end_time, as it looks rather silly. Signed-off-by: Paul Bolle <pebolle@tiscali.nl> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usbaudio: implement USB autosuspendOliver Neukum2011-03-11
| | | | | | | | | | | | Devices are autosuspended if no pcm nor midi channel is open Mixer devices may be opened. This way they are active when in use to play or record sound, but can be suspended while users have a mixer application running. [Small clean-ups using static inline by tiwai] Signed-off-by: Oliver Neukum <oneukum@suse.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* ALSA: usbaudio: fix suspend/resumeOliver Neukum2011-03-11
| | | | | | | | - ESHUTDOWN must be correctly handled - the optional interrupt endpoint's URB must be stopped and restarted Signed-off-by: Oliver Neukum <oneukum@suse.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* Merge branch 'fix/misc' into topic/miscTakashi Iwai2011-03-11
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| * ALSA: usb-audio: fix oops due to cleanup race when disconnectingTakashi Iwai2011-02-23
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | When a USB audio device is disconnected, snd_usb_audio_disconnect() kills all audio URBs. At the same time, the application, after being notified of the disconnection, might close the device, in which case ALSA calls the .hw_free callback, which should free the URBs too. Commit de1b8b93a0ba "[ALSA] Fix hang-up at disconnection of usb-audio" prevented snd_usb_hw_free() from freeing the URBs to avoid a hang that resulted from this race, but this introduced another race because the URB callbacks could now be executed after snd_usb_hw_free() has returned, and try to access already freed data. Fix the first race by introducing a mutex to serialize the disconnect callback and all PCM callbacks that manage URBs (hw_free and hw_params). Reported-and-tested-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com> Cc: <stable@kernel.org> [CL: also serialize hw_params callback] Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: HDA: Fix mic initialization in VIA auto parserDavid Henningsson2011-02-22
| | | | | | | | | | | | | | | | | | | | This typo caused some microphone inputs not to be correctly initialized on VIA codecs. Reported-By: Mark Goldstein <goldstein.mark@gmail.com> Cc: stable@kernel.org Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: fix one memory leak in sound jackLu Guanqun2011-02-21
| | | | | | | | | | | | Signed-off-by: Lu Guanqun <guanqun.lu@intel.com> Reviewed-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: HDA: Do not announce false surround in Conexant autoDavid Henningsson2011-02-19
| | | | | | | | | | | | | | | | | | | | | | Without this patch, one line-out and one speaker and Conexant's auto parser would announce (non-working) surround capabilities. BugLink: http://bugs.launchpad.net/bugs/721126 Cc: stable@kernel.org Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: HDA: Conexant auto: Handle multiple connections to ADC nodeDavid Henningsson2011-02-19
| | | | | | | | | | | | | | | | | | | | | | | | | | Conexant 20641 has several inputs to its ADC node, with one selector and individual amps for all inputs. This patch adds support in the Conexant auto parser to handle that case. It also means that the pin node's volume is being renamed to "Boost" to avoid name clash with the new volume controls on the ADC node. BugLink: http://bugs.launchpad.net/bugs/719524 Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: HDA: Add position_fix quirk for an Asus deviceDavid Henningsson2011-02-14
| | | | | | | | | | | | | | | | | | | | | | The bug reporter claims that position_fix=1 is needed for his microphone to work. The controller PCI vendor-id is [1002:4383] (rev 40). Reported-by: Kjell L. BugLink: http://bugs.launchpad.net/bugs/718402 Cc: stable@kernel.org Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: caiaq - Fix possible string-buffer overflowTakashi Iwai2011-02-14
| | | | | | | | | | | | | | | | | | Use strlcpy() to assure not to overflow the string array sizes by too long USB device name string. Reported-by: Rafa <rafa@mwrinfosecurity.com> Cc: stable <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: au88x0 - Modify pointer callback to give accurate playback positionRaymond Yau2011-02-14
| | | | | | | | | | Signed-off-by: Raymond Yau <superquad.vortex2@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * Merge branch 'fix/asoc' into for-linusTakashi Iwai2011-02-13
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| | * ASoC: Improve WM8994 digital power sequencingMark Brown2011-02-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | On WM8994 revision D and earlier ensure optimal sequencing with simultaneous usage of AIF1 and AIF2 by tying the signals together so if paths through both are connected the streams are started simultaneously. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
| | * ASoC: Create an AIF1ADCDAT signal widget to match AIF2Mark Brown2011-02-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Due to the different routing for AIF1 and AIF2 we weren't using a single widget to represent the ADCDAT signal. For consistency add one. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
| | * asoc: davinci: da830/omap-l137: correct cpu_dai_nameVaibhav Bedia2011-02-03
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | McASP1 is used on the DA830/OMAP-L137 platform for the codec. This is different from the DA850/OMAP-L138 platform which uses McASP0. This is fixed by adding a new snd_soc_dai_link struct. Signed-off-by: Vaibhav Bedia <vaibhav.bedia@ti.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: fill in snd_soc_pcm_runtime.card before calling snd_soc_dai_link.init()Janusz Krzysztofik2011-02-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The .card member of the snd_soc_pcm_runtime structure pointed to by the snd_soc_dai_link.init() argument used to be initialized before the function being called. This has changed, probably unintentionally, after recent refactorings. Since the function implementations are free to make use of this pointer, move its assignment back before the function is called to avoid NULL pointer dereferences. Created and tested on Amstrad Delta againts linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ALSA: hda - add quirk for Ordissimo EVE using a realtek ALC662Anisse Astier2011-02-11
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This netbook has a only one jack output and an internal mic. By default, mic and jack sense aren't working. Using lenovo-101e parameters makes both work. The device seems based on a Sharetronic Q70, so this should fix audio for this model too. Signed-off-by: Anisse Astier <anisse@astier.eu> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hrtimer: remove superfluous tasklet invocationClemens Ladisch2011-02-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit bb758e9637e5ddc removed snd_hrtimer_callback() from the hardware interrupt handler, thus moving it into a tasklet, but did not tell the ALSA timer framework about this, so the timer handling would now be done in the ALSA timer tasklet scheduled from another tasklet. To fix this, add the flag to tell the ALSA timer framework that the timer handler is already being invoked in a tasklet. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hrtimer: handle delayed timer interruptsClemens Ladisch2011-02-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | If a timer interrupt was delayed too much, hrtimer_forward_now() will forward the timer expiry more than once. When this happens, the additional number of elapsed ALSA timer ticks must be passed to snd_timer_interrupt() to prevent the ALSA timer from falling behind. This mostly fixes MIDI slowdown problems on highly-loaded systems with badly behaved interrupt handlers. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Reported-and-tested-by: Arthur Marsh <arthur.marsh@internode.on.net> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: HDA: Add subwoofer quirk for Acer Aspire 8942GDavid Henningsson2011-02-10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | According to the reporter, node 0x15 needs to be muted for subwoofer to stop sounding. This pin is marked as unused by BIOS, so fix that. BugLink: http://bugs.launchpad.net/bugs/715877 Cc: stable@kernel.org (2.6.37+) Reported-by: Hans Peter Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - Don't handle empty patch filesTakashi Iwai2011-02-10
| | | | | | | | | | | | | | | | | | | | | When an empty string is passed to patch option, the driver should ignore it. Otherwise it gets an error by trying to load it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - Fix missing CA initialization for HDMI/DPTakashi Iwai2011-02-08
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit 53d7d69d8ffdfa60c5b66cc2e9ee0774aaaef5c0 ALSA: hdmi - support infoframe for DisplayPort dropped the initialization of CA field accidentally. This resulted in only two-channel LPCM mode on Nvidia machines. Reference: kernel bug 28592 https://bugzilla.kernel.org/show_bug.cgi?id=28592 Signed-off-by: Takashi Iwai <tiwai@suse.de> Cc: <stable@kernel.org>
| * | ALSA: usbaudio - Enable the E-MU 0204 USBJoseph Teichman2011-02-08
| | | | | | | | | | | | | | | Signed-off-by: Joseph Teichman <josteich@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - switch lfe with side in mixer for 4930gŁukasz Wojniłowicz2011-02-07
| | | | | | | | | | | | | | | | | | | | | | | | Built-in sub-woofer can now be controlled by lfe slider instead of side slider on Acer Aspire 5930g Signed-off-by: Łukasz Wojniłowicz <lukasz.wojnilowicz@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'fix/asoc' into for-linusTakashi Iwai2011-02-04
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| | * ASoC: CX20442: fix NULL pointer dereferenceJanusz Krzysztofik2011-02-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The CX20442 codec driver never provided the snd_soc_codec_driver's .reg_cache_default member. With the latest ASoC framework changes, it seems to be referred unconditionally, resulting in a NULL pointer dereference if missing. Provide it. Created and tested on Amstrad Delta against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Amstrad Delta: fix const related build errorJanusz Krzysztofik2011-02-01
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Amstrad Delta ASoC driver used to override the digital_mute() callback, expected to be not provided by the on-board CX20442 CODEC driver, with its own implementation. While this is still posssible when substituting the whole empty snd_soc_dai_driver.ops member (the CX20442 case), replacing snd_soc_dai_ops.digital_mute only is no longer correct after the snd_soc_dai_driver.ops member has been constified, and results in build error. Drop this actually not used code path in hope the CX20442 driver never provides its own snd_soc_dai_ops structure. Created and tested against linux-2.6.38-rc2 Signed-off-by: Janusz Krzysztofik <jkrzyszt@tis.icnet.pl> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Fix mask/val_mask confusion snd_soc_dapm_put_volsw()Stephen Warren2011-01-28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_soc_dapm_put_volsw() has variables for both the unshifted and shifted mask for updates commit 97404f (ASoC: Do DAPM control updates in the middle of DAPM sequences) got confused between the two of these. Since there's no need to keep a copy of the unshifted mask fix this and simplify the code by using only one mask variable. [Completely rewrote the changelog to describe the issue -- broonie.] Signed-off-by: Stephen Warren <swarren@nvidia.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: DaVinci: fix kernel panic due to uninitialized platform_dataManjunathappa, Prakash2011-01-28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This patch fixes the Kernel panic issue on accessing davinci_vc in cq93vc_probe function. struct davinci_vc is part of platform device's private driver data(codec->dev->p->driver_data) and this is populated by DaVinci Voice Codec MFD driver. Signed-off-by: Manjunathappa, Prakash <prakash.pm@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| | * ASoC: Fix module refcount for auxiliary devicesJarkko Nikula2011-01-26
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Commit f6c2ed5 "ASoC: Fix the device references to codec and platform drivers" moved codec driver refcount increments from soc_bind_dai_link into soc_probe_codec. However, the commit didn't remove try_module_get from soc_probe_aux_dev so the auxiliary device reference counts are incremented twice as the soc_probe_codec is called from soc_probe_aux_dev too. Signed-off-by: Jarkko Nikula <jhnikula@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
| * | ALSA: use linux/io.h to fix compile warningsTakashi Iwai2011-02-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | For helping to reduce Greert's regression list... src/sound/drivers/mtpav.c: error: implicit declaration of function 'inb' src/sound/drivers/mtpav.c: error: implicit declaration of function 'outb' ... Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - Fix memory leaks in conexant jack arraysTakashi Iwai2011-02-02
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The Conexant codec driver adds the jack arrays in init callback which may be called also in each PM resume. This results in the addition of new jack element at each time. The fix is to check whether the requested jack is already present in the array. Reference: Novell bug 668929 https://bugzilla.novell.com/show_bug.cgi?id=668929 Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | Merge branch 'topic/hda' into fix/hdaTakashi Iwai2011-01-31
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| | * | ALSA: HDA: Fix microphone(s) on Lenovo Edge 13David Henningsson2011-01-28
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/708521 This Edge 13 model has an internal mic at 0x1a and should therefore use the asus quirk. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | ALSA: HDA: cxt5066 - Use asus model for Asus U50F, select correct SPDIF outputAndy Robinson2011-01-24
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Changed the Asus A52J quirk to use the asus model instead of the hp_laptop model, which fixes the external mic input. Added an Asus U50F quirk to use the asus model. For the cxt5066 codecs, added checking of the digital output pins to determine which digital output nodes to use instead of always using node 0x21, since some systems have node 0x12 connected to a SPDIF out jack. [A slight modification for better readability by tiwai] Signed-off-by: Andy Robinson <ajr55555@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | ALSA: HDA: Add a new model "asus" for Conexant 5066/205xxDavid Henningsson2011-01-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | BugLink: http://bugs.launchpad.net/bugs/701271 This new model, named "asus", is identical to the "hp_laptop" model, except for the location of the internal mic, which is at pin 0x1a. It is used for Asus K52JU and Lenovo G560. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| | * | ALSA: HDA: Refactor some redundant code for Conexant 5066/205xxDavid Henningsson2011-01-22
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Four very similar procedures - one for each model - now refactored into one. This isn't all duplicated code, but a step in the right direction. Signed-off-by: David Henningsson <david.henningsson@canonical.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: oxygen: fix output routing on Xonar DGClemens Ladisch2011-01-31
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This card uses separate I2S outputs for the front speakers and headphones, and reverses the order of the three speaker outputs. To work around this, add a model-specific callback to adjust the controller's playback routing. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | sound: silent echo'ed messages in MakefileAmerigo Wang2011-01-31
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Silent these echo's, please. Signed-off-by: WANG Cong <amwang@redhat.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | Merge branch 'fix/asoc' into for-linusTakashi Iwai2011-01-28
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| | * | ASoC: correct link specifications for corgi, poodle and spitzDmitry Eremin-Solenikov2011-01-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ASoC DAI link descriptions for Corgi, Poodle and Spitz platforms contained incorrect names for cpu_dai and codec, which effectievly disabled sound on theese platforms. Fix that errors. Signed-off-by: Dmitry Eremin-Solenikov <dbaryshkov@gmail.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| | * | ASoC: Samsung: Fix outdated cpu_dai_name for s3c24xx i2sLars-Peter Clausen2011-01-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | During the multi-component patch the s3c24xx i2s driver was renamed from "s3c24xx-i2s" to "s3c24xx-iis", while the sound board drivers were not updated to reflect this change as well. As a result there is no match between the dai_link and the i2s driver and no sound card is instantiated. This patch fixes the problem by updating the sound board drivers to use "s3c24xx-iis" for the cpu_dai_name. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Jassi Brar <jassi.brar@samsung.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| | * | ASoC: Fix codec device id format used by some dai_linksLars-Peter Clausen2011-01-25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The id part of an I2C device name is created with the "%d-%04x" format string. So for example for an I2C device which is connected to the adapter with the id 0 and has its address set to 0x1a the id part of the devices name would be "0-001a". Currently some sound board drivers have the id part the codec_name field of their dai_link structures set as if it had been created by a "%d-0x%x" format string. For example "0-0x1a" instead of "0-001a". As a result there is no match between the codec device and the dai_link and no sound card is instantiated. This patch fixes it. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| | * | ASoC: Handle low measured DC offsets for wm_hubs devicesMark Brown2011-01-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The DC servo codes are actually signed numbers so need to be treated as such. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Cc: stable@kernel.org
| | * | ASoC: da8xx/omap-l1xx: match codec_name with i2c idsRajashekhara, Sudhakar2011-01-21
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The codec_name entry for da8xx evm in sound/soc/davinci/davinci-evm.c is not matching with the i2c ids in the board file. Without this fix the soundcard does not get detected on da850/omap-l138/am18x evm. Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj@ti.com> Tested-by: Dan Sharon <dansharon@nanometrics.ca> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org (for 2.6.37)
| | * | ASoC: WM8994: fix wrong value in tristate functionQiao Zhou2011-01-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | fix wrong value in wm8994_set_tristate func. when updating reg bits, it should use "value", not "reg". Signed-off-by: Qiao Zhou <zhouqiao@marvell.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
| | * | ASoC: WM8995: Fix incorrect use of snd_soc_update_bits()Dimitris Papastamos2011-01-19
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | In the wm8995_set_tristate() function when updating the register bits use the value and not the register index as the value argument. Signed-off-by: Dimitris Papastamos <dp@opensource.wolfsonmicro.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Cc: stable@kernel.org
generated by cgit v1.2.2 (git 2.25.0) at 2024-12-04 22:07:06 -0500