- 09 Feb, 2007 40 commits
-
-
Tobin Davis authored
This patch adds support for the Toshiba M105-S3041 laptop (ALC861). Signed-off-by:
Tobin Davis <tdavis@dsl-only.net> Signed-off-by:
Takashi Iwai <tiwai@suse.de> Signed-off-by:
Jaroslav Kysela <perex@suse.cz>
-
Raúl Sánchez Siles authored
soc-dapm ·Removed list_for_each since the loop is list_for_each_entry() and not list_for_each(). Thanks to Liam Girdwood and Seth Forshee. at91-i2s ·Fixed typo in dai modes definition. ·Fixed struct member name in at91_ssc_info->ssc_state. ·Fixed compilation problem, ssc_state is bundled in at91_ssc_info. Signed-off-by:
Raúl Sánchez Siles <rss@barracuda.es> Signed-off-by:
Seth Forshee <seth.forshee@gmail.com> Signed-off-by:
-
Tobin Davis authored
This patch changes the default configuration for the Asus P5GD1 motherboard from 5stack to asus, as reported by stelek on linuxquestions.org http://www.linuxquestions.org/questions/showthread.php?p=2556497#post2556497Signed-off-by:
-
Clement Guedez authored
This patch adds the support of the ESI Waveterminal 192M soundcard to the ice1724 familly ALSA driver. It's a semi-professionnal soundcard for home studio : many I/O and a quality of sound is good, better than consumer cards, but less musical than professional cards. It use a Via Envy24ht chipset as ice1724 soundcard, Sigmatel stac9640 ADC/DAC for the analog I/O as Prodigy192, and Atmel ak4114 for S/PDIF as ESI Julia. Is working : the 8 analog outputs, the analog inputs 1&2, the mic input 1, the coaxial & optical digital outputs. Signed-off-by:
Clement Guedez <klem.dev@gmail.com> Signed-off-by:
-
Tobin Davis authored
This patch adds the Asus P5W DH to the ALC882 config table as a 6stack-dig system. Signed-off-by:
-
Tobin Davis authored
This patch adds the macpro and fixes a typo in the ALC882 section of ALSA-Configuration.txt. Signed-off-by:
-
John Daiker authored
Went rummaging through usbaudio.c and found some castings that aren't needed as far as I can see. Part of the KernelJanitors TODO list. Signed-off-by:
John Daiker <daikerjohn@gmail.com> Signed-off-by:
-
Randy Cushman authored
Previously, ac97_codec.c was coded to support AD1986 and AD1986A CODECs using code written for the AD1985 CODEC. This allowed the LINE_OUT and HEADPHONE jacks to function properly, however register differences between the CODECs prevented line and microphone inputs from functioning. Specifically, this patch fixes issues with the following mixer controls: 'V_REFOUT', 'Spread Front to Surround and Center/LFE', 'Exchange Front/Surround', 'Surround Jack Mode', and 'Channel Mode'. This patch removes the undocumented AD1888 control 'High Pass Filter Enable' and adds the new control 'Exchange Mic/Line In'. Signed-off-by:
Randy Cushman <rcushman_linux@earthlink.net> Signed-off-by:
-
Randy Cushman authored
This patch replaces the 'V_REFOUT Enable' mixer switch control with a listbox control for the AD1985 CODEC. Previous patch 'AD1888 mixer controls for DC mode' added controls that were propogated to multiple codecs. For the AD1985 codec, the bits VREFH and VREFD function differently, preventing the 'V_REFOUT Enable' control from setting V_REFOUT to Hi-Z. This patch also corrects an issue in which register bits relating to mixer controls 'Surround Jack Mode' and 'Channel Mode'. The register bits controlled by these controls were being set at boot time to states inconsistent with the stored values of these controls. Signed-off-by:
-
Takashi Iwai authored
Fix the changes realted to delayed_work in soc/codecs/wm8750.c. Signed-off-by:
-
Randy Cushman authored
This patch fixes the Microphone and LINE_IN select logic for Analog Devices surround codecs with shared jacks. The existing code can never utilize the shared jacks for Microphone and LINE_IN due to the reversed jack selection logic. The patched code correctly selects the shared jack for input if the 'Channel Mode' selector does not specify that the jack is to be used for output. Specifically, in '2ch' mode the Center/LFE jack is used for microphone input and the Surround jack is used for LINE_IN, in '4ch' mode the Center/LFE jack is used for microphone input and the Surround jack is used for output, and in '6ch' mode both jacks are used for output. Signed-off-by:
-
Takashi Iwai authored
Use global workqueue for simplicity instead of own workqueue in SoC core and wm8750 codes. Signed-off-by:
-
Takashi Iwai authored
Use global workqueue for simplicity instead of own workqueue. Signed-off-by:
-
Takashi Iwai authored
Use global workqueue for simplicity. The unsolicited event frequency isn't so high to have own queue. Signed-off-by:
-
Robert P. J. Day authored
Change the two remaining instances in the tree of kcalloc(1,...) to the corresponding kzalloc() call. Signed-off-by:
Robert P. J. Day <rpjday@mindspring.com> Signed-off-by:
-
James C Georgas authored
The STAC9708/11 AC97 codecs implement the PCM Out Path & Mute bit in the General Purpose register (0x20:F), even though they don't implement the actual function in the mixer. Since the alsa tests for the function by toggling the bit and reading it back to see if it changed, it mistakenly creates a useless control. This patch explicitly removes the control when the codec is an STAC9708/11. I put the check in patch_sigmatel_stac9708_specific(), because I have an SBLive with this chip on it. I don't know if the STAC9758 or other codecs also behave this way. If they do, then this check could maybe go in patch_sigmatel_stac97xx_specific(), or some other more general function. Signed-off-by:
James C Georgas <jgeorgas@rogers.com> Signed-off-by:
-
Krzysztof Helt authored
This is a comment fix to avoid misleading about locking in the dbri_cmdsend. Signed-off-by:
Krzysztof Helt <krzysztof.h1@wp.pl> Signed-off-by:
-
Liam Girdwood authored
This patch adds copyright and credit for my good friend Richard Purdie from OpenedHand for his help and code contribution throughout the development of the core code. Many thanks Richard (I guess we overlooked this in trying to get everything working well). It also adds some extra comments wrt to DAI clock matching. Signed-off-by:
Liam Girdwood <lg@opensource.wolfsonmicro.com> Signed-off-by:
-
Liam Girdwood authored
This patch removes some trailing white space from the WM9712 ASoC codec driver. Signed-off-by:
-
Tobin Davis authored
This patch adds the Turbo-X Coeus G610P to the alc880 config table, based on user provided information. Signed-off-by:
-
Johannes Berg authored
When the machine resumes the onyx codec might be in a weird state. Hence, simply fully reset it once (and keep the code to take it out of suspend in case the suspend of the codec chip survives a reset). Signed-off-by:
Johannes Berg <johannes@sipsolutions.net> Signed-off-by:
-
Matt Porter authored
Adds support for handling EAPD on 9205 codecs Signed-off-by:
Matt Porter <mporter@embeddedalley.com> Signed-off-by:
-
Toshimune Konno authored
This patch supports Audiotrack 7.1 XT. 7.1XT is almost same hardware as 7.1LT. so using 7.1 LT's code. Signed-off-by:
Toshimune Konno <heitouk@nifty.com> Signed-off-by:
-
Takashi Iwai authored
Fixed the renamed AC97_BUS in soc/pxa/Kconfig file. Signed-off-by:
-
Tobin Davis authored
This patch adds limited support for Intel-based MacPro workstations. Currently, the front headphone jack is not functioning, but line out and line in are working. S/PDIF not tested. Signed-off-by:
-
Andrew Morton authored
From: Andrew Morton <akpm@osdl.org> I converted the workqueues to per-device while I was there. It seems strange to create a new kernel thread (on each CPU!) and to then only have a single global work to ever be queued upon it. Plus without this, I'd have to use the _NAR stuff, gawd help me. Does that workqueue really need to be per-cpu? Does that workqueue really need to exist? Why not use keventd? Cc: Takashi Iwai <tiwai@suse.de> Cc: David Howells <dhowells@redhat.com> Signed-off-by:
Andrew Morton <akpm@osdl.org> Signed-off-by:
-
Andrew Morton authored
From: Andrew Morton <akpm@osdl.org> Cc: Takashi Iwai <tiwai@suse.de> Signed-off-by:
-
James Courtier-Dutton authored
Signed-off-by:
James Courtier-Dutton <James@superbug.co.uk> Signed-off-by:
-
James Courtier-Dutton authored
Fixes ALSA bug#2286 Signed-off-by:
-
Andrew Morton authored
Cc: Takashi Iwai <tiwai@suse.de> Signed-off-by:
-
Olaf Hering authored
create sysfs driver symlink for snd-aoa in /sys/bus/aoa-soundbus/devices/*/ Acked-by:
Olaf Hering <olaf@aepfle.de> Signed-off-by:
-
Olaf Hering authored
create sysfs device symlinks for snd-aoa in /sys/class/sound/controlC0 This allows hald to recognize the device as sound device. Furthermore it allows the desktop user to actually access the sound device nodes. hald and related packages will modify the acl attributes. Fixes https://bugzilla.novell.com/show_bug.cgi?id=106294Acked-by:
-
Randy Dunlap authored
fixes: include/sound/pcm.h:62: warning: 'struct snd_pcm_substream' declared inside parameter list Signed-off-by:
Randy Dunlap <randy.dunlap@oracle.com> Signed-off-by:
-
James Courtier-Dutton authored
Notebook. Digital playback and capture now works, but it is not bit accurate because it passes through a resampler. Bit accurate playback and capture will be implemented later via the p17v. Signed-off-by:
-
James Courtier-Dutton authored
Implement functionallity in order to fixe ALSA bug#2058. Signed-off-by:
-
Teru KAMOGASHIRA authored
Following patch will make the driver to use the 44.1kHz SRC automatically if the pcm source is 44.1kHz signed 16bit stereo. The SRC is available in YMF754 only. Signed-off-by:
Teru KAMOGASHIRA <teru@sodan.ecc.u-tokyo.ac.jp> Signed-off-by:
Clemens Ladisch <clemens@ladisch.de> Signed-off-by:
-
Joseph Chan authored
This patch is VIA first release for HD audio codec, VT1708(A) and it provides geneneral HD audio driver features. Signed-off-by:
Joseph Chan <josephchan@via.com.tw> Signed-off-by:
-
Takashi Iwai authored
Added a description about spdif_aclink option for snd-intel8x0 driver in ALSA-Configuration.txt. Signed-off-by:
-
Adrian Bunk authored
Make the following needlessly global functions static: - dapm_power_widgets() - dapm_mux_update_power() - dapm_mixer_update_power() - dapm_free_widgets() Signed-off-by:
Adrian Bunk <bunk@stusta.de> Signed-off-by:
-
Jonathan Woithe authored
The following patch creates a new 'Mono speaker' control in alsamixer when the Realtek 'acer' model is used with hda_intel. This is needed so the internal mono speaker (when present) can be controlled. This new control won't do anything in Acer laptops which are not fitted with a mono speaker. Acer models which are known to have a mono speaker are the C20x tablet series but there may be others. I guess we could define a new model specifically for Acers with mono speakers but this seems a bit silly given that such a model will be identical to the normal 'acer' model except for this added control. This patch also adds the C20x tablets to the list of PCI ids associated with the 'acer' model. This means that owners of C20x machines will no longer have to supply 'model=acer' when loading hda_intel. Signed-off-by:
Jonathan Woithe <jwoithe@physics.adelaide.edu.au> Signed-off-by:
-
