Commit Graph

5654 Commits

Author SHA1 Message Date
Takashi Iwai
a68c4d1133 ALSA: dummy - Fake buffer allocations
Instead of allocating the real buffers, use a fake buffer and ignore
read/write in the dummy driver so that we can save the resources.
For mmap, a single page (unique to the direction, though) is reused
to all buffers.

When the app requires to read/write the real buffers, pass fake_buffer=0
module option at loading time.  This will get back to the old behavior.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-07 09:01:10 +02:00
Takashi Iwai
b5d1078173 ALSA: dummy - Fix the timer calculation in systimer mode
Fix the expire-time calculation in the systimer mode when the buffer
size isn't aligned to the period size.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-04 08:45:11 +02:00
Takashi Iwai
b142037b4c ALSA: dummy - Better jiffies handling
In the system-timer mode, snd-dummy driver issues each tick to update
the position.  This is highly inefficient and even inaccurate if the
timer can't be triggered at each tick.

Now rewritten to wake up only at the period boundary.  The position
is calculated from the current jiffies.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-03 16:01:06 +02:00
Takashi Iwai
c631d03c68 ALSA: dummy - Support high-res timer mode
Allow snd-dummy driver to use high-res timer as its timing source
instead of the system timer.  The new module option "hrtimer" is added
to turn on/off the high-res timer support.  It can be switched even
dynamically via sysfs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-09-03 15:59:26 +02:00
Linus Torvalds
a206e9417f Merge branch 'fix/misc' of 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()
2009-08-25 09:47:06 -07:00
Clemens Ladisch
b1ddaf681e sound: pcm_lib: fix unsorted list constraint handling
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>
2009-08-25 08:52:34 +02:00
Clemens Ladisch
edd1365e90 sound: vx222: fix input level control range check
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>
2009-08-24 09:46:08 +02:00
Bartlomiej Zolnierkiewicz
70bdbd3d1a ALSA: ali5451: fix timeout handling in snd_ali_{codecs,timer}_ready()
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>
2009-08-23 18:58:07 +02:00
Linus Torvalds
a1d1251115 Merge branch 'fix/hda' of 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
2009-08-20 10:19:39 -07:00
Takashi Iwai
3abf2f3639 ALSA: hda - Fix probe of Toshiba laptops with ALC268 codec
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>
2009-08-19 20:05:02 +02:00
Wu Fengguang
ae709440ed ALSA: hda: add model for Intel DG45ID/DG45FC boards
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>
2009-08-19 12:10:25 +02:00
Wu Fengguang
150fe14c1a ALSA: hda: enable speaker output for Compaq 6530s/6531s
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>
2009-08-19 12:07:27 +02:00
Takashi Iwai
8884be98bc Merge branch 'fix/hda' into for-linus
* fix/hda:
  ALSA: hda - Don't override ADC definitions for ALC codecs
  ALSA: hda - Add missing vmaster initialization for ALC269
2009-08-12 08:05:20 +02:00
Takashi Iwai
909a2607a5 Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: Add missing DRV_NAME definitions for fsl/* drivers
2009-08-12 08:05:19 +02:00
Takashi Iwai
dd704698f5 ALSA: hda - Don't override ADC definitions for ALC codecs
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>
2009-08-11 08:45:11 +02:00
Takashi Iwai
100d5eb36b ALSA: hda - Add missing vmaster initialization for ALC269
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>
2009-08-10 11:57:05 +02:00
Takashi Iwai
afc5e65245 ASoC: Add missing DRV_NAME definitions for fsl/* drivers
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>
2009-08-07 16:33:53 +02:00
Linus Torvalds
6ce90c430b Merge branch 'fix/hda' of 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
2009-08-04 15:39:55 -07:00
Roel Kluin
4b35d2ca23 ALSA: hda - Read buffer overflow
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>
2009-08-03 08:34:06 +02:00
Chengu Wang
84d3dc200f ALSA: hda: Correct EAPD for Dell Inspiron 1525
The commit 24918b61b5 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>
2009-08-03 08:30:56 +02:00
Wu Fengguang
e310bb0646 ALSA: hda: warn on spurious response
To help disclose hardware bugs.

Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:27:53 +02:00
Wu Fengguang
feb273404f ALSA: hda: remember last command for each codec
Signed-off-by: Wu Fengguang <fengguang.wu@intel.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-08-03 08:27:39 +02:00
Wu Fengguang
c32649feb4 ALSA: hda: read CORBWP inside reg_lock
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>
2009-08-03 08:26:55 +02:00
Wu Fengguang
cdb1fbf231 ALSA: hda: take reg_lock in azx_init_cmd_io/azx_free_cmd_io
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>
2009-08-03 08:26:42 +02:00
Wu Fengguang
a678cdee25 ALSA: hda: take cmd_mutex in probe_codec()
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>
2009-08-03 08:26:23 +02:00
Wu Fengguang
deadff1665 ALSA: hda: track CIRB/CORB command/response states for each codec
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>
2009-08-03 08:26:13 +02:00
Takashi Iwai
ce577e8cf5 ALSA: hda - Fix quirk for Toshiba Satellite A135-S4527
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>
2009-08-03 08:23:52 +02:00
Takashi Iwai
ec86fe5209 Merge branch 'fix/oss' into for-linus
* fix/oss:
  sound: mpu401.c: Buffer overflow
  sound: aedsp16: Buffer overflow
2009-07-31 10:17:45 +02:00
Takashi Iwai
d62e345f14 Merge branch 'fix/misc' into for-linus
* fix/misc:
  ALSA: sound/aoa: Add kmalloc NULL tests
2009-07-31 10:17:44 +02:00
Takashi Iwai
6280b61af5 Merge branch 'fix/hda' into for-linus
* 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
2009-07-31 10:17:42 +02:00
Julia Lawall
f065fabc86 ALSA: sound/aoa: Add kmalloc NULL tests
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>
2009-07-31 10:14:58 +02:00
Takashi Iwai
aa563af763 ALSA: hda - Increase PCM stream name buf in patch_realtek.c
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>
2009-07-31 10:06:34 +02:00
Roel Kluin
a987004fbc sound: mpu401.c: Buffer overflow
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>
2009-07-29 14:41:24 +02:00
Roel Kluin
c45ec06c74 sound: aedsp16: Buffer overflow
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>
2009-07-29 14:37:12 +02:00
Roel Kluin
78735cffc2 ALSA: hda: fix out-of-bound hdmi_eld.sad[] write
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>
2009-07-29 14:35:20 +02:00
Takashi Iwai
626f5cefc6 ALSA: hda - Add quirk for Dell Studio 1555
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>
2009-07-28 00:54:39 +02:00
Takashi Iwai
57e4a5c4f8 Merge branch 'fix/usb-audio' into for-linus
* fix/usb-audio:
  ALSA: usb-audio - Volume control quirk for QuickCam E 3500
2009-07-26 11:07:08 +02:00
Takashi Iwai
b88158846f Merge branch 'fix/pcm-hwptr' into for-linus
* 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
2009-07-26 11:07:07 +02:00
Takashi Iwai
de5d674c02 Merge branch 'fix/hda' into for-linus
* 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
2009-07-26 11:07:06 +02:00
Takashi Iwai
f35e2965b2 Merge branch 'fix/ctxfi' into for-linus
* fix/ctxfi:
  ALSA: ctxfi - Fix uninitialized error checks
2009-07-26 11:07:05 +02:00
Takashi Iwai
29769d533b Merge branch 'fix/caiaq' into for-linus
* fix/caiaq:
  ALSA: snd_usb_caiaq: add support for Audio2DJ
2009-07-26 11:07:04 +02:00
Takashi Iwai
7679d5c65b Merge branch 'fix/asoc' into for-linus
* fix/asoc:
  ASoC: tlv320aic3x: Enable PLL when not bypassed
2009-07-26 11:07:03 +02:00
Takashi Iwai
8de56b7deb ALSA: hda - Fix mute control with some ALC262 models
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>
2009-07-24 16:54:31 +02:00
Daniel Mack
b30c494773 ALSA: snd_usb_caiaq: add support for Audio2DJ
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>
2009-07-23 16:31:58 +02:00
Takashi Iwai
947ca210f1 ALSA: pcm - Fix hwptr buffer-size overlap bug
The fix 79452f0a28 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>
2009-07-23 16:21:08 +02:00
Takashi Iwai
8935064043 ALSA: pcm - Fix warnings in debug loggings
Add proper cast.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-07-23 14:28:37 +02:00
Takashi Iwai
cedb8118e8 ALSA: pcm - Add logging of hwptr updates and interrupt updates
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>
2009-07-23 11:09:03 +02:00
Chaithrika U S
06c71282a9 ASoC: tlv320aic3x: Enable PLL when not bypassed
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>
2009-07-23 08:14:29 +01:00
Takashi Iwai
4012ade933 ALSA: hda - Restore GPIO1 properly at resume with AD1984A
The commit 099db17e66 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>
2009-07-22 18:15:10 +02:00
Takashi Iwai
68110661e8 ALSA: ctxfi - Fix uninitialized error checks
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>
2009-07-22 17:05:15 +02:00