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2009-06-08ALSA: ctxfi - Fix possible buffer pointer overrunTakashi Iwai
Fix possible buffer pointer overruns. Back to zero when it's equal or over the buffer size. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08ALSA: ctxfi - Remove useless initializations and castTakashi Iwai
Remove useless variable initializations and cast at the beginning of functions. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08ALSA: ctxfi - Fix DMA mask for emu20k2 chipTakashi Iwai
Allow 64bit DMA mask for emu20k2 chip, too. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08ALSA: ctxfi - Make volume controls more intuitiveTakashi Iwai
Change the volume control to dB scale (as the raw data seems so). Also added the TLV dB-scale information. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08ALSA: ca0106 - Fix master volume scaleTakashi Iwai
The master volume dB scale was wrongly defined as 0.50dB setp while it must be 0.25dB step. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-08ALSA: ctxfi - Optimize the native timer handling using wc counterTakashi Iwai
Optimize the timer update routine to look up wall clock once instead of checking the position of each stream at each timer update. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-07ALSA: pcm - Fix update of runtime->hw_ptr_interruptTakashi Iwai
The commit 13f040f9e55d41e92e485389123654971e03b819 made another regression, the missing update of runtime->hw_ptr_interrupt. Since this field is only checked in snd_pcmupdate__hw_ptr_interrupt(), not in snd_pcm_update_hw_ptr(), it must be updated before the hw_ptr change check. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-07ALSA: sgio2audio.c: clean up checkingFigo.zhang
vfree() does it's own 'NULL' check,so no need for check before calling it. Signed-off-by: Figo.zhang <figo1802@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-06ALSA: pcm - Fix a typo in hw_ptr update checkJaroslav Kysela
Fix a typo in the commit 13f040f9e55d41e92e485389123654971e03b819 ALSA: PCM midlevel: Do not update hw_ptr_jiffies when hw_ptr is not changed which causes obvious problems with PA. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-06ASoC: Apostrophe patrolMark Brown
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-06ASoC: codec tlv320aic23 fix bogus divide by 0 messageTroy Kisky
Some code analyzer software mistakenly gives divide by 0 error messages for these lines. This patch will end its confusion. Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-05ALSA: ctxfi - Add missing inclusion of linux/math64.hTakashi Iwai
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05ALSA: Clean up 64bit division functionsTakashi Iwai
Replace the house-made div64_32() with the standard div_u64*() functions. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05ALSA: ctxfi - Set device 0 for mixer control elementsTakashi Iwai
Mixer control elements are usually assigned to device 0. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05ALSA: ctxfi - Clean up / optimizeTakashi Iwai
- Use static tables instead of assigining each funciton pointer - Add __devinit* to appropriate places; pcm, mixer and timer cannot be marked because they are kept in the function table that lives long - Move create_alsa_devs function out of struct ct_atc to mark it __devinit Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05ALSA: ctxfi - Set periods_min to 2Takashi Iwai
Set 2 to minimal periods of playback pcm setups, too. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05ALSA: ctxfi - Use native timer interrupt on emu20k1Takashi Iwai
emu20k1 has a native timer interrupt based on the audio clock, which is more accurate than the system timer (from the synchronization POV). This patch adds the code to handle this with multiple streams. The system timer is still used on emu20k2, and can be used also for emu20k1 easily by changing USE_SYSTEM_TIMER to 1 in cttimer.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05ALSA: ctxfi - Fix previous fix for 64bit DMATakashi Iwai
Remove unneeded substitution to 32bit int to make it really working. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05ALSA: support Sony Vaio TTGuido Günther
with BIOS probing only we offer a non functional headphone swith and volume slider. Signed-off-by: Guido Günther <agx@sigxcpu.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05ALSA: ctxfi - Fix endian-dependent codesTakashi Iwai
The UAA-mode check in hwct20k1.c is implemented with the endian-dependent codes. Fix to be more portable (and readable). Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-05ALSA: ctxfi - Allow 64bit DMATakashi Iwai
emu20kx chips support 64bit address PTE. Allow the DMA bit mask to accept 64bit address, too. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04ASoC: fix NULL pointer dereference in soc_suspend()Daniel Mack
In case the initalization of an soc_device failed, there is no codec associated with it. soc_suspend() will still dereference the pointer and cause an Ooops when entering the sleep mode. This happens on our board with a multi-target kernel image when booted on a machine without audio circuits. This patch makes the code bail out very early in this special case. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-04ALSA: hda_intel: fix build error when !PMAlexander Beregalov
Fix this build error when CONFIG_PM is not set: ound/pci/hda/hda_intel.c: In function 'azx_bus_reset': sound/pci/hda/hda_intel.c:1270: error: implicit declaration of function 'snd_pcm_suspend_all' sound/pci/hda/hda_intel.c:1271: error: implicit declaration of function 'snd_hda_suspend' sound/pci/hda/hda_intel.c:1272: error: implicit declaration of function 'snd_hda_resume' Signed-off-by: Alexander Beregalov <a.beregalov@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04ALSA: Add missing __devexit_p() markersJean Delvare
3 ISA sound drivers lack their __devexit_p() markers, which would cause build failures when the kernel is built without hotplug support. Signed-off-by: Jean Delvare <khali@linux-fr.org> Cc: Kyle McMartin <kyle@mcmartin.ca> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04ASoC: Fix build error in twl4030.cTakashi Iwai
Fix the (likely cut-n-paste) error by commit 16a30fbb0d3aa4ee829a2dd3d0e314e2b5ae96a9, which causes the error below: sound/soc/codecs/twl4030.c: In function 'twl4030_read_reg_cache': sound/soc/codecs/twl4030.c:152: error: 'cache' undeclared (first use in this function) Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04ALSA: Core - clean up snd_card_set_id* calls and remove possible id collisionJaroslav Kysela
Move locking outside snd_card_set_id_internal() function and rename it to snd_card_set_id_no_lock() for better function description. User defined id is just copied to card structure at allocation time. The real unique id procedure is called in snd_card_register() to ensure real atomicity. Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04ALSA: hda - More Aspire 8930G fixesHector Martin
Enable all three capture channels, including the missing nid 7 which is the only one capable of capturing DMIC input Enable Headphone amp for the HP jack. This causes a volume boost for headphones, but does not cause any noticeable effect for light loads like other amps, so there is no need to make it configurable. Add Input Mix capture mux setting to capture the output of the playback input mux (that is, what goes out the speakers except for PCM) Hack another coef register because the stereo DMIC for some reason produces a nonstandard sum/difference signal. I found a bit to make it just use the sum signal for both channels, which makes it behave like a standard mono microphone. The stereo is useless anyway (they're 1cm apart). Tested working: Three capture channels, mic in, line in, DMIC. Tested not working: CD. Not sure why, might be unconnected in the actual hardware or a CD drive issue. Also looked at SPDIF. It appears to work (emitter lights up inside the HP out jack) but I lack a proper miniTOSLINK cable to test it. Signed-off-by: Hector Martin <hector@marcansoft.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-04ALSA: burgundy: timeout message is off by one.Roel Kluin
Timeout message is off by one. Signed-off-by: Roel Kluin <roel.kluin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-03ALSA: Fix double locking of card list in snd_card_register()Mark Brown
The introduction of snd_card_set_id() added a lock on the card list to the old choose_default_id() function when using it to implement the new API call. This lock is needed to allow us to walk the list and check to see if our new name is a duplicate. Unfortunately this causes a lockup when called from snd_card_register() (in cases where no ID is supplied for the card) since the card list is already locked there. Fix this fairly hideously by factoring out the implementation and using a flag to indicate if the lock should be held. A better fix would probably be to refactor snd_card_register() to move the _set_id() outside the locking region but I can't immediately see anything I can convince myself is safe. Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-03ASoC: SSM2602: assign last substream to the master when shutting downCliff Cai
Fixes crash when shutting down. Signed-off-by: Cliff Cai <cliff.cai@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-03ASoC: Blackfin: document how anomaly 05000250 is handledSonic Zhang
Signed-off-by: Sonic Zhang <sonic.zhang@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-03ASoC: Blackfin: set the transfer size according the ac97_frame sizeCliff Cai
Signed-off-by: Cliff Cai <cliff.cai@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Signed-off-by: Bryan Wu <cooloney@kernel.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-03ASoC: SSM2602: remove unsupported sample ratesCliff Cai
Signed-off-by: Cliff Cai <cliff.cai@analog.com> Signed-off-by: Mike Frysinger <vapier@gentoo.org> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-06-03ALSA: powermac - Replace the rest of __init*Takashi Iwai
All __initdata should be __devinitdata as platform device is hotpluggable. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-03ALSA: sound/ppc: update annotations of serveral functionsStephen Rothwell
[I am not sure if this is the correct approach as I don't know if any of this actual hardware or drivers are really hot pluggable.] Gets rid of these build warnings: WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x5c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_new() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_new(). If .snd_pmac_new is only used by .snd_pmac_probe then annotate .snd_pmac_new with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x10c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_burgundy_init() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_burgundy_init(). If .snd_pmac_burgundy_init is only used by .snd_pmac_probe then annotate .snd_pmac_burgundy_init with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x164): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_daca_init() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_daca_init(). If .snd_pmac_daca_init is only used by .snd_pmac_probe then annotate .snd_pmac_daca_init with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1dc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_init() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_tumbler_init(). If .snd_pmac_tumbler_init is only used by .snd_pmac_probe then annotate .snd_pmac_tumbler_init with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x1ec): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_tumbler_post_init() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_tumbler_post_init(). If .snd_pmac_tumbler_post_init is only used by .snd_pmac_probe then annotate .snd_pmac_tumbler_post_init with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x28c): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_awacs_init() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_awacs_init(). If .snd_pmac_awacs_init is only used by .snd_pmac_probe then annotate .snd_pmac_awacs_init with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2bc): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_pcm_new() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_pcm_new(). If .snd_pmac_pcm_new is only used by .snd_pmac_probe then annotate .snd_pmac_pcm_new with a matching annotation. WARNING: sound/ppc/snd-powermac.o(.devinit.text+0x2f8): Section mismatch in reference from the function .snd_pmac_probe() to the function .init.text:.snd_pmac_attach_beep() The function __devinit .snd_pmac_probe() references a function __init .snd_pmac_attach_beep(). If .snd_pmac_attach_beep is only used by .snd_pmac_probe then annotate .snd_pmac_attach_beep with a matching annotation. Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-03ALSA: usb-audio - errata corrige for quirkAndrea Borgia
Cut'n'paste mistake, whose likely result was nothing at all. Correct version is "USB_DEVICE", not "USB_DEVICE_VENDOR_SPEC". Signed-off-by: Andrea Borgia <andrea@borgia.bo.it> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02ALSA: bt87x - Add a quirk entry for Askey Computer Corp. MagicTView'99Takashi Iwai
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02ALSA: ca0106 - Add missing card->mixername field setupTakashi Iwai
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02Merge branch 'topic/ctxfi-fix' into topic/ctxfiTakashi Iwai
2009-06-02ALSA: ctxfi - Support SG-buffersTakashi Iwai
Use SG-buffers instead of contiguous pages. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02ALSA: ctxfi - Remove PAGE_SIZE limitationTakashi Iwai
Remove the limitation of PAGE_SIZE to be 4k by defining the own page size and macros for 4k. 8kb page size could be natively supported, but it's disabled right now for simplicity. Also, clean up using upper_32_bits() macro. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02ALSA: ctxfi - Fix supported PCM formatsTakashi Iwai
The device seems supporting only U8, S16, S24_3LE, S32. Other linear formats result in bad outputs. Also, added the support for 32bit float format, which wasn't listed in the original code. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02ALSA: ctxfi - Fix PCM device namingTakashi Iwai
PCM names for surround streams should be also fixed as well as the mixer element names. Also, a bit clean up for PCM name setup. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02ALSA: ctxfi - Fix surround mixer namesTakashi Iwai
We usually pick up "Surround" mixer for the rear output, and "Side" for the extra surround. Fix the channel mapping to follow it. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02ALSA: ALSA: ctxfi - Release PCM resources at each prepare callTakashi Iwai
The prepare callback can be called multiple times, thus it needs to release and acquire the resource again by itself at the second or later call. Simply add pcm_release_resources() at the beginning of each prepare callback in ctatc.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02ALSA: Remove invalid GENERIC_MIX PCM sublassTakashi Iwai
SNDRV_PCM_SUBCLASS_GENERIC_MIX is mostly for h/w multi-stream playback devices, but ca0106 and emu10k1x don't support it (unlike emu10k1). We shouldn't set that flag to avoid confusion. Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02ALSA: snd_usb_caiaq: bump version numberDaniel Mack
Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02ALSA: snd_usb_caiaq: give better shortnameDaniel Mack
If not passed as module option, provide an own card ID with the newly introduced snd_set_card_id() call. This will prevent ALSA from calling choose_default_name() which only takes the last part of a name containing whitespaces. This for example caused 'Audio 4 DJ' to be shortened to 'DJ', which was not very descriptive. The implementation now takes the short name and removes all whitespaces from it which is much nicer. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-06-02Merge branch 'topic/core-id-check' into topic/caiaqTakashi Iwai
2009-06-02ALSA: Core - add snd_card_set_id() functionJaroslav Kysela
Introduce snd_card_set_id() function to allow lowlevel drivers to set default identification name for card slot. The function checks also for identification name collisions and tries to create unique name. Also, the snd_card_create() function is simplified, because this new function is used. As bonus, proper name collision checks are evaluated at the card create time. Signed-off-by: Jaroslav Kysela <perex@perex.cz> Signed-off-by: Takashi Iwai <tiwai@suse.de>
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