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* Merge branch 'fix/misc' of ↵Linus Torvalds2009-08-253-39/+22
|\ | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'fix/misc' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: sound: pcm_lib: fix unsorted list constraint handling sound: vx222: fix input level control range check ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready()
| * sound: pcm_lib: fix unsorted list constraint handlingClemens Ladisch2009-08-251-31/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | snd_interval_list() expected a sorted list but did not document this, so there are drivers that give it an unsorted list. To fix this, change the algorithm to work with any list. This fixes the "Slave PCM not usable" error with USB devices that have multiple alternate settings with sample rates in decreasing order, such as the Philips Askey VC010 WebCam. http://bugzilla.kernel.org/show_bug.cgi?id=14028 Reported-and-tested-by: Andrzej <adkadk@gmail.com> Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Cc: <stable@kernel.org> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * sound: vx222: fix input level control range checkClemens Ladisch2009-08-241-2/+2
| | | | | | | | | | | | | | | | Fix a logic error in the range check of the input level control that would prevent setting any volume less than the maximum. Signed-off-by: Clemens Ladisch <clemens@ladisch.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready()Bartlomiej Zolnierkiewicz2009-08-231-6/+12
| | | | | | | | | | | | | | | | | | | | | | | | Modify loops in such way that the register value is checked also after the timeout condition, just in case the heavy interrupt load etc. caused the thread to sleep for the time period exceeding the timeout value. While at it remove an extra ALI_STIMER read from snd_ali_stimer_ready(). Reported-by: Jack Byer <ojbyer@usa.net> Signed-off-by: Bartlomiej Zolnierkiewicz <bzolnier@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | Merge branch 'fix/hda' of ↵Linus Torvalds2009-08-203-4/+23
|\ \ | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ALSA: hda - Fix probe of Toshiba laptops with ALC268 codec ALSA: hda: add model for Intel DG45ID/DG45FC boards ALSA: hda: enable speaker output for Compaq 6530s/6531s
| * | ALSA: hda - Fix probe of Toshiba laptops with ALC268 codecTakashi Iwai2009-08-191-2/+13
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are many variants of Toshiba laptops with ALC268 codec, and it seems that a few of them don't work with model=toshiba preset since they have the secondary ALC268 codec just for HDMI output. This is a regression due to the previous clean-up work to merge all Toshiba quirk entries into a single check. This patch adds the identification of such laptops to apply the standard BIOS-probing method. Unfortunately, Toshiba laptops have all the same PCI SSID, so we need to check the codec SSID to identify each device. Tested-by: Alexey Dobriyan <adobriyan@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda: add model for Intel DG45ID/DG45FC boardsWu Fengguang2009-08-191-0/+6
| | | | | | | | | | | | | | | | | | | | | The BIOS pin configs are in fact correct and shall not be overwritten. Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda: enable speaker output for Compaq 6530s/6531sWu Fengguang2009-08-191-2/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | HP Compaq 6530s and 6531s internal speaker is silence or becomes silence within 1 minute after fresh boot. It is found that pin 0x1c must be set to PIN_OUT mode to make the speaker work. This is weird - line-in pin 0x1c and speaker pin 0x16 seem to be unrelated. The codec differences before/after patch are: @@ Node 0x17 [Pin Complex] wcaps 0x40020b: Pin Default 0x41a6e130: [N/A] Mic at Ext Rear Conn = Digital, Color = White DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE - Pin-ctls: 0x24: IN + Pin-ctls: 0x40: OUT @@ Node 0x1c [Pin Complex] wcaps 0x40018d: Pin Default 0x41813021: [N/A] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x2, Sequence = 0x1 - Pin-ctls: 0x24: IN VREF_80 + Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 1 0x24 Tests show that it won't impact (external) Mic recording. Reported-by: "Lin, Ming M" <ming.m.lin@intel.com> Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | Merge branch 'fix/hda' into for-linusTakashi Iwai2009-08-121-6/+14
|\ \ \ | |/ / | | | | | | | | | | | | * fix/hda: ALSA: hda - Don't override ADC definitions for ALC codecs ALSA: hda - Add missing vmaster initialization for ALC269
| * | ALSA: hda - Don't override ADC definitions for ALC codecsTakashi Iwai2009-08-111-6/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | ALC269 and ALC861-VD parsers override the ADC definitions unconditionally without checking the spec definition. This causes the problem when any inconsistent ADC is set up in the device quirk (like ALC272 with digital-mic). This patch avoids the overriding by adding the proper checks. Reference: Novell bnc#529467 https://bugzilla.novell.com/show_bug.cgi?id=529467 Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | ALSA: hda - Add missing vmaster initialization for ALC269Takashi Iwai2009-08-101-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Without the initialization of vmaster NID, the dB information got confused for ALC269 codec. Reference: Novell bnc#527361 https://bugzilla.novell.com/show_bug.cgi?id=527361 Signed-off-by: Takashi Iwai <tiwai@suse.de> Cc: <stable@kernel.org>
* | | Merge branch 'fix/asoc' into for-linusTakashi Iwai2009-08-122-0/+4
|\ \ \ | | | | | | | | | | | | | | | | * fix/asoc: ASoC: Add missing DRV_NAME definitions for fsl/* drivers
| * | | ASoC: Add missing DRV_NAME definitions for fsl/* driversTakashi Iwai2009-08-072-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Module builds are broken due to missing DRV_NAME for efika-audio-fabric and pcm030-audio-fabric. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | Merge branch 'fix/hda' of ↵Linus Torvalds2009-08-045-34/+86
|\ \ \ \ | | |/ / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6 * 'fix/hda' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: ALSA: hda - Read buffer overflow ALSA: hda: Correct EAPD for Dell Inspiron 1525 ALSA: hda: warn on spurious response ALSA: hda: remember last command for each codec ALSA: hda: read CORBWP inside reg_lock ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io ALSA: hda: take cmd_mutex in probe_codec() ALSA: hda: track CIRB/CORB command/response states for each codec ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
| * | | ALSA: hda - Read buffer overflowRoel Kluin2009-08-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | Check whether index is within bounds before testing the element. Signed-off-by: Roel Kluin <roel.kluin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda: Correct EAPD for Dell Inspiron 1525Chengu Wang2009-08-031-1/+8
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit 24918b61b55c21e09a3e07cd82e1b3a8154782dc statically changes the model from dell-bios to dell-3stack to solve the sound decreasing regression (http://lkml.org/lkml/2008/9/12/203), however it leads to another problem that the 2nd headphone jack doesn't work (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3987). So I think the commit 249**2dc is just a workaround. I would like to give a true solution here. The datasheet for STAC9228 says, GPIO2 is the same pin as VOL DOWN, and the EAPD pin is GPIO0. This is why the sound decreases if we set EAPD as GPIO2. This patch changes EAPD to GPIO0 to solve the problem. Signed-off-by: Chengu Wang <wangchengu@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda: warn on spurious responseWu Fengguang2009-08-031-1/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | To help disclose hardware bugs. Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda: remember last command for each codecWu Fengguang2009-08-031-5/+6
| | | | | | | | | | | | | | | | | | | | Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda: read CORBWP inside reg_lockWu Fengguang2009-08-031-1/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | This converts the last CORBWP access outside of reg_lock. Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_ioWu Fengguang2009-08-031-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Just for safety. azx_init_cmd_io() and azx_free_cmd_io() may be called when switching to single command mode. Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda: take cmd_mutex in probe_codec()Wu Fengguang2009-08-031-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Now that each codec will have its own module, it is possible for the user to load one codec while another one is running. So cmd_mutex would be a safe addition to probe_codec(). Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda: track CIRB/CORB command/response states for each codecWu Fengguang2009-08-033-24/+56
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Recently we hit a bug in our dev board, whose HDMI codec#3 may emit redundant/spurious responses, which were then taken as responses to command for another onboard Realtek codec#2, and mess up both codecs. Extend the azx_rb.cmds and azx_rb.res to array and track each codec's commands/responses separately. This helps keep good codec safe from broken ones. Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527Takashi Iwai2009-08-031-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Use model=lenovo instead of model=dallas for Toshiba Satellite A135-S4527 with ALC861-VD codec. Reference: Novell bnc#526325 https://bugzilla.novell.com/show_bug.cgi?id=526325 Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | Merge branch 'fix/oss' into for-linusTakashi Iwai2009-07-312-5/+6
|\ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | * fix/oss: sound: mpu401.c: Buffer overflow sound: aedsp16: Buffer overflow
| * | | | sound: mpu401.c: Buffer overflowRoel Kluin2009-07-291-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | mpu_synth_info[m].name is a char[30], and the minimum length of the data written by sprintf is 31 bytes including terminating null. Signed-off-by: Roel Kluin <roel.kluin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | sound: aedsp16: Buffer overflowRoel Kluin2009-07-291-4/+5
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | DSPVersion is declared as char[3], but the sprintf writes at least 4 bytes including terminating null. Signed-off-by: Roel Kluin <roel.kluin@gmail.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'fix/misc' into for-linusTakashi Iwai2009-07-311-0/+4
|\ \ \ \ \ | | |_|_|/ | |/| | | | | | | | | | | | | * fix/misc: ALSA: sound/aoa: Add kmalloc NULL tests
| * | | | ALSA: sound/aoa: Add kmalloc NULL testsJulia Lawall2009-07-311-0/+4
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Check that the result of kzalloc is not NULL before a dereference. The semantic match that finds this problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @@ expression *x; identifier f; constant char *C; @@ x = \(kmalloc\|kcalloc\|kzalloc\)(...); ... when != x == NULL when != x != NULL when != (x || ...) ( kfree(x) | f(...,C,...,x,...) | *f(...,x,...) | *x->f ) // </smpl> Signed-off-by: Julia Lawall <julia@diku.dk> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'fix/hda' into for-linusTakashi Iwai2009-07-313-4/+6
|\ \ \ \ \ | | |_|/ / | |/| | | | | | | | | | | | | | | | | | | | | | | * fix/hda: ALSA: hda - Increase PCM stream name buf in patch_realtek.c ALSA: hda: fix out-of-bound hdmi_eld.sad[] write ALSA: hda - Add quirk for Dell Studio 1555
| * | | | ALSA: hda - Increase PCM stream name buf in patch_realtek.cTakashi Iwai2009-07-311-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The name buf with size 16 is too short for some codec names, e.g. truncated like "ALC861-VD Analo". Now the size is doubled. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: hda: fix out-of-bound hdmi_eld.sad[] writeRoel Kluin2009-07-291-2/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | e->sad[] is declared with size ELD_MAX_SAD=16, but the guard allows range 0-31. Signed-off-by: Roel Kluin <roel.kluin@gmail.com> Signed-off-by: Wu Fengguang <fengguang.wu@intel.com> Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | ALSA: hda - Add quirk for Dell Studio 1555Takashi Iwai2009-07-281-0/+2
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Added a quirk entry for Dell Studio 1555. Reference: Novell bnc#525244 https://bugzilla.novell.com/show_bug.cgi?id=525244 Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | Merge branch 'fix/usb-audio' into for-linusTakashi Iwai2009-07-261-5/+20
|\ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | * fix/usb-audio: ALSA: usb-audio - Volume control quirk for QuickCam E 3500
| * | | | | ALSA: usb-audio - Volume control quirk for QuickCam E 3500Alexey Fisher2009-07-221-5/+20
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | - E3500 report cval->max more than it actually can handel, so if you set 95% capture level it will be silently muted. - Betwen cval->min and cval-max(real) is 2940 control units, but real are only 7 with cval->res = 384. - Alsa can't handel less than 10 controls, so make it more and set cval->res = 192. Signed-off-by: Alexey Fisher <bug-track@fisher-privat.net> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | Merge branch 'fix/pcm-hwptr' into for-linusTakashi Iwai2009-07-261-1/+35
|\ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * fix/pcm-hwptr: ALSA: pcm - Fix hwptr buffer-size overlap bug ALSA: pcm - Fix warnings in debug loggings ALSA: pcm - Add logging of hwptr updates and interrupt updates ALSA: pcm - Fix regressions with VMware
| * | | | | | ALSA: pcm - Fix hwptr buffer-size overlap bugTakashi Iwai2009-07-231-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The fix 79452f0a28aa5a40522c487b42a5fc423647ad98 introduced another bug due to the missing offset for the overlapped hwptr. When the hwptr goes back to zero, the delta value has to be corrected with the buffer size. Otherwise this causes looping sounds. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | ALSA: pcm - Fix warnings in debug loggingsTakashi Iwai2009-07-231-12/+12
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Add proper cast. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | ALSA: pcm - Add logging of hwptr updates and interrupt updatesTakashi Iwai2009-07-231-0/+25
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Added the logging functionality to xrun_debug to record the hwptr updates via snd_pcm_update_hw_ptr() and snd_pcm_update_hwptr_interrupt(), corresponding to 16 and 8, respectively. For example, # echo 9 > /proc/asound/card0/pcm0p/xrun_debug will record the position and other parameters at each period interrupt together with the normal XRUN debugging. Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | ALSA: pcm - Fix regressions with VMwareTakashi Iwai2009-07-221-1/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | VMware tends to report PCM positions and period updates at utterly wrong timing. This screws up the recent PCM core code that tries to correct the position based on the irq timing. Now, when a backward irq position is detected, skip the update instead of rebasing. (This is almost the old behavior before 2.6.30.) Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | Merge branch 'fix/hda' into for-linusTakashi Iwai2009-07-263-19/+18
|\ \ \ \ \ \ \ | | |_|/ / / / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * fix/hda: ALSA: hda - Fix mute control with some ALC262 models ALSA: hda - Restore GPIO1 properly at resume with AD1984A ALSA: hda - Use snprintf() to be safer
| * | | | | | ALSA: hda - Fix mute control with some ALC262 modelsTakashi Iwai2009-07-241-17/+16
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The master mute switch is wrongly implemented as checking the pointer instead of its value, thus it can be never muted. This patch fixes the issue. Reference: Novell bnc#404873 https://bugzilla.novell.com/show_bug.cgi?id=404873 Signed-off-by: Takashi Iwai <tiwai@suse.de> Cc: <stable@kernel.org>
| * | | | | | ALSA: hda - Restore GPIO1 properly at resume with AD1984ATakashi Iwai2009-07-221-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | The commit 099db17e66294b02814dee01c81d9abbbeece93e introduced a regression at suspend/resume where the GPIO1 bit isn't properly restored, thus the speaker output gets muted initially after resume. The fix is simple, use the cached write for storing GPIO data. Reference: Novell bnc#522764 https://bugzilla.novell.com/show_bug.cgi?id=522764 Signed-off-by: Takashi Iwai <tiwai@suse.de>
| * | | | | | ALSA: hda - Use snprintf() to be saferTakashi Iwai2009-07-221-1/+1
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Use snprint() for creating the jack name string instead of sprintf() in patch_sigmatel.c. Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | Merge branch 'fix/ctxfi' into for-linusTakashi Iwai2009-07-262-12/+9
|\ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * fix/ctxfi: ALSA: ctxfi - Fix uninitialized error checks
| * | | | | | | ALSA: ctxfi - Fix uninitialized error checksTakashi Iwai2009-07-222-12/+9
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Fix a few uninitialized error checks that were introduced recently mistakenlly during the clean-up: sound/pci/ctxfi/ctamixer.c: In function ‘get_amixer_rsc’: sound/pci/ctxfi/ctamixer.c:261: warning: ‘err’ may be used uninitialized in this function sound/pci/ctxfi/ctamixer.c: In function ‘get_sum_rsc’: sound/pci/ctxfi/ctamixer.c:415: warning: ‘err’ may be used uninitialized in this function sound/pci/ctxfi/ctsrc.c: In function ‘get_srcimp_rsc’: sound/pci/ctxfi/ctsrc.c:742: warning: ‘err’ may be used uninitialized in this function Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | Merge branch 'fix/caiaq' into for-linusTakashi Iwai2009-07-264-1/+10
|\ \ \ \ \ \ \ \ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * fix/caiaq: ALSA: snd_usb_caiaq: add support for Audio2DJ
| * | | | | | | | ALSA: snd_usb_caiaq: add support for Audio2DJDaniel Mack2009-07-234-1/+10
| | |_|_|_|_|/ / | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | This adds support for Native Instrument's freshly announced Audio2DJ sound device hardware. Version number bumped to 1.3.19. Signed-off-by: Daniel Mack <daniel@caiaq.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
* | | | | | | | Merge branch 'fix/asoc' into for-linusTakashi Iwai2009-07-261-1/+10
|\ \ \ \ \ \ \ \ | |/ / / / / / / |/| | | | | | / | | |_|_|_|_|/ | |/| | | | | * fix/asoc: ASoC: tlv320aic3x: Enable PLL when not bypassed
| * | | | | | ASoC: tlv320aic3x: Enable PLL when not bypassedChaithrika U S2009-07-231-1/+10
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | PLL was not being enabled when it was not bypassed. This patch enables the PLL when it is used. Additionally, it disables the PLL when it is bypassed. Without this patch, the audio on TI DM646x EVM and DM355 EVM does not work properly. The bit clocks and the frame sync signals from the codec are not correct and hence the playback/record are faster than usual for most sample rates. The reason for this was that the PLL was not enabled when it was not bypassed. Tested on DM6467 EVM, playback tested on DM355 EVM. Signed-off-by: Chaithrika U S <chaithrika@ti.com> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
* | | | | | | Merge branch 'fix/misc' into for-linusTakashi Iwai2009-07-213-9/+6
|\ \ \ \ \ \ \ | | |_|_|_|_|/ | |/| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | * fix/misc: ALSA: ca0106 - Fix the max capture buffer size ALSA: OSS sequencer should be initialized after snd_seq_system_client_init ALSA: sound/isa: convert nested spin_lock_irqsave to spin_lock
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