5117 Commits

Author SHA1 Message Date
9311c9b4f1 ALSA: snd-usb-caiaq: drop bogus iso packets
Drop inbound packets that are smaller than expected. This has been
observed at the very beginning of the streaming transaction.

And when the hardware is in panic mode (which can only very rarely
happen in case of massive EMI chaos), mute the input channels.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Tested-by: Mark Hills <mark@pogo.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 11:31:08 +01:00
1313e70414 ALSA: snd-usb-caiaq: only warn once on streaming errors
Limit the number of printed warnings to one in case of streaming errors.
printk() happens to be expensive, especially in code called as often as
here.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 11:27:51 +01:00
f1aa298679 Merge branch 'fix/opl3sa2-suspend' into for-linus 2009-03-18 08:04:36 +01:00
a232ee66e0 Merge branch 'fix/hda' into for-linus 2009-03-18 08:04:16 +01:00
6af845e4eb ALSA: Fix vunmap and free order in snd_free_sgbuf_pages()
In snd_free_sgbuf_pags(), vunmap() is called after releasing the SG
pages, and it causes errors on Xen as Xen manages the pages
differently.  Although no significant errors have been reported on
the actual hardware, this order should be fixed other way round,
first vunmap() then free pages.

Cc: Jan Beulich <jbeulich@novell.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 08:04:01 +01:00
82f5d57163 ALSA: mixart, fix lock imbalance
There is an omitted unlock in one snd_mixart_hw_params fail path. Fix it.

Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 08:03:49 +01:00
91054598f7 ALSA: pcm_oss, fix locking typo
s/mutex_lock/mutex_unlock/ on 2 fail paths in snd_pcm_oss_proc_write.
Probably a typo, lock should be unlocked when leaving the function.

Signed-off-by: Jiri Slaby <jirislaby@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 08:03:33 +01:00
36c7b833e5 ALSA: oss-mixer - Fixes recording gain control
At the time of initialization, SNDRV_MIXER_OSS_PRESENT_PVOLUME bit is not
set for MIC (slot 7).
So, the same should not be checked when an application tries to do gain
control for audio recording devices.

Just check slot->present for SNDRV_MIXER_OSS_PRESENT_CVOLUME independently.
Verified with a simple application which opens /dev/dsp for recording and
/dev/mixer for volume control.

Have tested two usb audio mic devices.

Signed-off-by: Viral Mehta <viral.mehta@einfochips.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 07:52:28 +01:00
4a10079345 Merge branch 'fix/hda' into topic/hda 2009-03-18 07:50:56 +01:00
ee5047102c ALSA: snd-hda-intel - add checks for invalid values to *query_supported_pcm()
If ratesp or formatsp values are zero, wrong values are passed to ALSA's
the PCM midlevel code. The bug is showed more later than expected.

Also, clean a bit the code.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 07:50:44 +01:00
c673ba1c23 ALSA: hda - Workaround for buggy DMA position on ATI controllers
The position-buffer on ATI controllers are unreliable as well as
on VIA chips, thus the same workaround for DMA position reading as
VIA is useful for ATI.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 07:46:21 +01:00
09240cf429 ALSA: hda - Fix DMA mask for ATI controllers
ATI controllers (at least some SB0600 models) appear buggy to handle
64bit DMA.  As a workaround, reset GCAP bit0 and let the driver to
use only 32bit DMA on these controllers.

Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-18 07:45:41 +01:00
da88b48b84 Merge branch 'pxa-ssp' into for-2.6.30 2009-03-17 19:07:26 +00:00
323a59613e ALSA: drop outdated and broken sa11xx-uda1341 driver
It depends on L3 support from 2.4 kernel (CONFIG_L3) that never got
merged into mainline. Since there's no way to use it on any of
supported machines (iPaq h3100 or h3600), better drop it for now.
It can be reimplemented later using ASoC infrastructure (there's
already a driver for uda1341 codec in mainline, so only CPU and machine
parts need to be written).

Signed-off-by: Dmitry Artamonow <mad_soft@inbox.ru>
Cc: Russell King <linux@arm.linux.org.uk>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-17 17:58:13 +01:00
dbe36c9dd5 Merge branch 'topic/snd_card_new-err' into topic/drop-l3 2009-03-17 17:57:37 +01:00
d2314e0e27 ASoC: Only deregister AC97 dev if it's name was not "AC97"
The commit 14fa43f53ff3a9c3d8b9662574b7369812a31a97 ("ASoC: Only
register AC97 bus if it's not done already") added a condition for
calling of soc_ac97_dev_register() but not added for calling of
soc_ac97_dev_unregister().  This patch adds same condition for
soc_ac97_dev_unregister().  Without this fix, kernel crashes when
unloading an asoc driver.

Signed-off-by: Atsushi Nemoto <anemo@mba.ocn.ne.jp>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-17 13:59:47 +00:00
37ba1b6283 Merge branch 'fix/opl3sa2-suspend' into topic/isa-misc 2009-03-17 09:28:13 +01:00
dde332b660 ALSA: opl3sa2 - Fix NULL dereference when suspending snd_opl3sa2
Fix the OOPS during a opl3sa2 card suspend
and resume if the driver is loaded but the card
is not found.

Signed-off-by: Krzysztof Helt <krzysztof.h1@wp.pl>
Cc: <stable@kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-17 09:27:47 +01:00
40f49e7ed7 sh: dma: Make G2 DMA configurable.
Follow the PVR2 DMAC change for G2 DMA.

Signed-off-by: Paul Mundt <lethal@linux-sh.org>
2009-03-17 12:47:56 +09:00
60aa49243d Rationalize fasync return values
Most fasync implementations do something like:

     return fasync_helper(...);

But fasync_helper() will return a positive value at times - a feature used
in at least one place.  Thus, a number of other drivers do:

     err = fasync_helper(...);
     if (err < 0)
             return err;
     return 0;

In the interests of consistency and more concise code, it makes sense to
map positive return values onto zero where ->fasync() is called.

Cc: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
2009-03-16 08:34:35 -06:00
db1dd4d376 Use f_lock to protect f_flags
Traditionally, changes to struct file->f_flags have been done under BKL
protection, or with no protection at all.  This patch causes all f_flags
changes after file open/creation time to be done under protection of
f_lock.  This allows the removal of some BKL usage and fixes a number of
longstanding (if microscopic) races.

Reviewed-by: Christoph Hellwig <hch@lst.de>
Cc: Al Viro <viro@ZenIV.linux.org.uk>
Signed-off-by: Jonathan Corbet <corbet@lwn.net>
2009-03-16 08:32:27 -06:00
b9591448e5 ALSA: hda - Fix ALC662 beep again
The previous commit breaks the (digital-) beep on ALC662.
ALC662 has the connection index 0x05 while ALC662 and ALC272 have the
index 0x04 for the beep widget.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-16 15:26:01 +01:00
b8dbed0f09 ALSA: snd-hda-intel: Fix ALC662/ALC663 Beep Amplifier Index
ALC662/663 codecs have Beep Amplifier Index 0x04 not 0x05 in 0x0b NID.
Confirmed by testing on real hardware.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-16 15:23:33 +01:00
852fd9e50f ASoC: Each PXA AC97 DAI needs a separate ops
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-16 14:13:57 +00:00
f2a5d6a2ea ASoC: Fix some missing dai_ops conversions
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-16 14:13:57 +00:00
10d9e3d99e ASoC: twl4030 - Fix build error
CC      sound/soc/codecs/twl4030.o
sound/soc/codecs/twl4030.c:1400: warning: braces around scalar initializer
sound/soc/codecs/twl4030.c:1400: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1401: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1401: warning: initialization from incompatible pointer type
sound/soc/codecs/twl4030.c:1402: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1402: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1402: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1402: warning: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: error: field name not in record or union initializer
sound/soc/codecs/twl4030.c:1403: error: (near initialization for 'twl4030_dai.ops')
sound/soc/codecs/twl4030.c:1403: warning: excess elements in scalar initializer
sound/soc/codecs/twl4030.c:1403: warning: (near initialization for 'twl4030_dai.ops')

Signed-off-by: Joonyoung Shim <jy0922.shim@samsung.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-16 14:13:56 +00:00
4c55bb0149 ALSA: echoaudio: remove line-out volume from vmixer cards
With this patch the drivers do not set the vmixer volume anymore at startup
because it is actually the output volume of the voices and ALSA mandates
that the volume must be 0 by default.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-16 08:38:00 +01:00
9f5d790d1b ALSA: echoaudio: remove line-out volume from vmixer cards
There is a long standing bug in the drivers for cards with a vmixer because
I overlooked a detail in the c++ generic driver by echoaudio. Those cards
do not have a line-out volume control. It is a virtual control provided by
the generic driver. The bug is harmless because the DSP just ignores the
command to change the volume.
*NB:* It breaks alsa-tools/echomixer. A patch for it will follow.

This patch removes the line-out volume control from vmixer-equipped cards.

Signed-off-by: Giuliano Pochini <pochini@shiny.it>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-16 08:37:29 +01:00
26ade896b6 ASoC: Allow choice of ac97 gpio reset line
As the PXA27x series allow 2 gpios to reset the ac97 bus,
allow through platform data configuration the definition of
the correct gpio which will reset the AC97 bus.

This comes from a silicon defect on the PXA27x series, where
the gpio must be manually controlled in warm reset cases.

Signed-off-by: Robert Jarzmik <rjarzmik@free.fr>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-15 20:20:37 +00:00
85fab7802a ASoC: Fix Zylonite for non-networked SSP mode
This also simplifies the code a bit.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-14 11:38:16 +00:00
0ce36c5f7f ASoC: Fix non-networked I2S mode for PXA SSP
Two issues are fixed here:

 - I2S transmits the left frame with the clock low but I don't seem to
   get LRCLK out without SFRMDLY being set so invert SFRMP and set a
   delay.
 - I2S has a clock cycle prior to the first data byte in each channel
   so we need to delay the data by one cycle.

Tested-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-14 11:37:46 +00:00
97fb44eb6b Merge branch 'for-rmk' of git://git.pengutronix.de/git/imx/linux-2.6 into devel
Conflicts:

	arch/arm/mach-at91/gpio.c
2009-03-13 21:44:51 +00:00
58d8395b74 ALSA: hda - Add another HP model with IDT92HD71bx codec
HP laptops require GPIO0 on as EAPD.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-13 17:04:34 +01:00
72d7466468 ASoC: switch PXA SSP driver from network mode to PSP
This switches the pxa ssp port usage from network mode to PSP mode.
Removed some comments and checks for configured TDM channels.
A special case is added to support configuration where BCLK = 64fs. We
need to do some black magic in this case which doesn't look nice but
there is unfortunately no other option than that.

Diagnosed-by: Tim Ruetz <tim@caiaq.de>
Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-13 13:23:34 +00:00
77dd7e17b8 ASoC: Move headset jack registration to device initialization for SDP3430
Move headset jack registration to the codec/machine specific
initialization. Having the jack registration in machine init
causes that the jack device gets initialized but not registered
since the sound card is registered before the jack. Moving jack
registration to device initialization will register the jack
device along with all other devices associated to the card when
the card is registed. As a consequence of jack device registered
properly, the jack is detected as an input device.

Signed-off-by: Misael Lopez Cruz <x0052729@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-13 12:08:53 +00:00
bb6ac72fb1 ALSA: hda - power up before codec initialization
Change the power state of each widget before starting the initialization
work so that all verbs are executed properly.

Also, keep power-up during hwdep reconfiguration.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-13 09:06:31 +01:00
307282c899 ALSA: hda - Add model=vaio for STAC9872
Add the default pin config for model=vaio (in case of broken BIOS).

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-12 18:17:58 +01:00
9421f9543b ALSA: hda - Print multiple out-amp values of pin widgets on Conext codecs
Add a flag to work around the non-standard amp-value handling on
Conexant codecs.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-12 17:06:07 +01:00
3b7523fc82 ALSA: hda - Add comments for the previous fix for conexant codecs
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-12 16:45:01 +01:00
eb5f6d753e ASoC: Replace remaining uses of snd_soc_cnew with snd_soc_add_controls.
The drivers are basically duplicating the same code over and over.
As snd_soc_cnew is going to be made static some time after the next
merge window, we might as well convert them now.

Signed-off-by: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-12 15:43:30 +00:00
6f7cb44ba1 ASoC: Move WM8580 to normal I2C device probe
Refactor the WM8580 device registration to probe via standard I2C device
registration, registering the DAIs once the device has probed via I2C.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-12 15:43:24 +00:00
5d75bc5578 ALSA: hda - fix headphone settings and master volume (Conexant CX20551)
Update the places where the 0x1d widget is used for Conexant 5047, fixing
mismatch introduced after changing the connection.

Signed-off-by: Gregorio Guidi <gregorio.guidi@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2009-03-12 16:41:51 +01:00
65ec1cd1e2 ASoC: Merge dai_ops factor out
Merge Eric Maio's patch to merge snd_soc_dai_ops out of line.  Fixed
merge issues and updated drivers, plus an issue with the ops for the two
s3c2443 AC97 DAIs having been merged.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-11 16:51:31 +00:00
5314adc361 ASoC: Fix formats for s3c24xx-i2s register prints
The register values are all u32 so don't need the long format.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-11 16:28:29 +00:00
02b7cbc399 ASoC: Remove version display from WM8580 driver
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-11 14:40:41 +00:00
aaf1e176fa ASoC: Add initial driver for the WM8400 CODEC
The WM8400 is a highly integrated audio CODEC and power management unit
intended for mobile multimedia application.  This driver supports the
primary audio CODEC features, including:

 - 1W speaker driver
 - Fully differential headphone output
 - Up to 4 differential microphone inputs

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-11 13:49:46 +00:00
5706d50132 ASoC: buildfix for OSK
Buildfix:

  CC      sound/soc/omap/osk5912.o
  sound/soc/omap/osk5912.c: In function 'osk_soc_init':
  sound/soc/omap/osk5912.c:189: error: implicit declaration of function 'clk_get_usecount'
  make[3]: *** [sound/soc/omap/osk5912.o] Error 1

There's no such (standard) clock interface.

Signed-off-by: David Brownell <dbrownell@users.sourceforge.net>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-11 12:49:28 +00:00
cbf1146d5e ASoC: don't touch pxa-ssp registers when stream is running
In pxa_ssp_set_dai_fmt(), check whether there is anything to do at all.
If there would be but the SSP port is in use already, bail out.

Signed-off-by: Daniel Mack <daniel@caiaq.de>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-10 19:44:04 +00:00
090cec81ae ALSA: ASoC: Davinci: Updated sffsdr_hw_params() function to new format
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-10 15:42:48 +00:00
14cbba89ae ALSA: ASoC: Davinci: Replaced DAI format RIGHT_J by DSP_B for SFFSDR
Signed-off-by: Hugo Villeneuve <hugo@hugovil.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
2009-03-10 15:42:48 +00:00