This reverts commit 8dc840f8.  Christian
Pellegrin <chripell@gmail.com> reported that on some systems the patch
caused DMA to fail which is much more serious than the original skipped
audio issue.  Further investigation by Dave shows that the behaviour
depends on the clock speed of the SoC - a better fix is neeeded.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Originally it was put too tight limits to support only 44.1 kHz and 48 kHz
sample rates in McBSP DAI driver. Extend it now to 8 kHz - 96 kHz. With
96 kHz and 2*16 bits, bit clock is 3.072 MHz < 3.125 MHz (I2S max?).

Tested on Nokia N810 with TVL320AIC33 from rates 8 - 96 kHz and on Texas
Instruments Beagle with TWL4030 from rates 8 - 48 kHz.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

TWL4030 currently supports rates between 8 kHz and 48 kHz and sets the codec
mode register accordingly in twl4030_hw_params. Expose this info so that
ASoC can match other rates than 44.1 kHz or 48 kHz as well.
Signed-off-by: Jarkko Nikula <jarkko.nikula@nokia.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Fixes swapping of channels at start of stereo playback.

Channel swap can be observed while playing left-only or right-only audio data. The channel
swap is fixed by handling the XSYNCERR condition.
Signed-off-by: Naresh Medisetty <naresh@ti.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The TI DVEVM board uses the SND_SOC_DAIFMT_CBM_CFM & I2S formats, but the
Lyrtech SFFSDR board uses the SND_SOC_DAIFMT_CBM_CFS & RIGHT-JUSTIFIED formats.
Signed-off-by: Hugo Villeneuve <hugo.villeneuve@lyrtech.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

fixes playing/recording of 8 bit audio files.

Generated on  20081108  against v2.6.27
Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Add support for more sample rates, different crystals
and split playback/capture rates.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Arun KS <arunks@mistralsolutions.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

According to TRM, 256*Fs clock output should be enabled
when TWL4030 is in slave mode, not master.
This allows sound to work on OMAP3 Pandora, which uses
256*Fs clock.
Signed-off-by: Grazvydas Ignotas <notasas@gmail.com>
Acked-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The S3C24xx dma does not allow more than one buffer to be enqueue prior to
the dma transfers starting. This patch adds an additional parameter to
s3c24xx_pcm_enqueue() to allow for passing an initial dma maximum load
value.

Signed-off-by: David Anders <danders at amltd.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Rather than try to remember to keep the core version number updated
(which hasn't been happening) just remove it.  It was much more useful
when ASoC was out of tree.
Signed-off-by: Mark brown <broonie@opensource.wolfsonmicro.com>

this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for
palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test).
I sent it here some time ago, but now I got to fixing bugs in it. It should
be somehow mostly ok and ready for applying.

[Converted to use snd_soc_dapm_nc_pin() and bool Kconfig -- broonie]
Signed-off-by: Marek Vasut <marek.vasut@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Takashi Iwai <tiwai@suse.de>

The file(s) below do not use LINUX_VERSION_CODE nor KERNEL_VERSION.
  sound/soc/codecs/ad73311.c

This patch removes the said #include <version.h>.
Signed-off-by: Huang Weiyi <weiyi.huang@gmail.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Call device_create_file only once in snd_soc_dapm_sys_add function.
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Add audio support for the Atmel AT91SAM9G20ek board(uing wolfson 8731).
It is based on the former eti_b1_wm8731.c file, using the atmel scc API.
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The Ateml AT91 and AVR32 SoC share common IP for audio and can share the
same driver code using the atmel-ssc API provided for both architectures.
Do this, creating a new unified atmel ASoC architecture to replace the
previous at32 and at91 ones.

[This was contributed as a patch series for reviewability but has been
squashed down to a single commit to help preserve both the history and
bisectability.  A small bugfix from Jukka is included.]
Tested-by: Jukka Hynninen <ext-jukka.hynninen@vaisala.com>
Signed-off-by: Sedji Gaouaou <sedji.gaouaou@atmel.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Steve Sakoman <steve@sakoman.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

sound/soc/soc-dapm.c: In function 'snd_soc_dapm_sys_add':
sound/soc/soc-dapm.c:828: error: 'ret' undeclared (first use in this function)
Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

The last change to Kconfig ca53fb24
added a wrong item SND_SOC_AC97, which must be SND_SOC_AC97_CODEC.
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Disable the automatic volume control feature of the CS4270 audio codec.  This
feature, which is enabled by default, causes volume change commands to be
delayed.  Sometimes the volume change happens after playback is started.
Signed-off-by: Timur Tabi <timur@freescale.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Move the bus dependencies in SND_SOC_ALL_CODECS into the individual
codec options rather than have them centrally. This allows the
inclusion of AC97 codecs when testing on platforms with AC97 support
and will also handle codecs on multi-function devices more gracefully.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The WM9713 comes out of cold reset in low power mode so always requires
a warm reset to bring up the AC97 link after a cold reset.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The SSP ports PXA series processors can be used to implement a variety of
audio interface formats. This patch implements support for I2S, DSP A and
DSP B modes on these ports.

This patch is based on the previous out of tree pxa2xx-ssp driver (which
was originally written by Liam Girdwood with updates from Philipp Zabel
and Nicola Perrino) and pxa3xx-ssp driver (originally written by Seth
Forsee based on the pxa2xx-ssp driver). Testing coverage is not complete
currently.
Tested-by: Daniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

As well as ensuring that UI-relevant parts of control names don't get
truncated in the DAPM code this avoids conflicts in long control names
that differ only at the end of a long string.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

Since we can query the playback stream power state directly we do not
need to infer if it is powered up from the timer being scheduled.  Doing
this avoids problems that previously existed with streams being
incorrectly determined to be powered up caused when the timer is
scheduled when streams are closed after being partially set up.
Reported-by: Nobin Mathew <nobin.mathew@gmail.com>
Reported-by: Jukka Hynninen <ext-jukka.hynninen@vaisala.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg
will set register 0x06 to a value of 0x59.
Also, pop_time debugfs interface setup is moved so that it
is setup in the same function as codec_reg
Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>

The control had an extra space at the end of the name.
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

Fix missing unsigned for irqsave flags in psc i2s driver
Make attribute visiblity static
Collect all sysfs errors before checking status

[Word wrapped DEVICE_ATTR() lines for 80 columns -- broonie]
Signed-off-by: Jon Smirl <jonsmirl@gmail.com>
Acked-by: Grant Likely <grant.likely@secretlab.ca>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

When ASoC was converted to support full int width masks SOC_SINGLE_VALUE()
omitted the assignment of rshift, causing the control operatins to report
some mono controls as stereo. This happened to work some of the time due
to a confusion between shift and min in snd_soc_info_volsw().
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

- Setting the TFS pin selector for SPORT 0 based on whether the selected
  port id F or G. If the port is F then no conflict should exist for the
  TFS. When Port G is selected and EMAC then there is a conflict between
  the PHY interrupt line and TFS.  Current settings prevent the conflict
  by ignoring the TFS pin when Port G is selected. This allows both
  ssm2602 using Port G and EMAC concurrently.

 - some code cleanup
Signed-off-by: Cliff Cai <cliff.cai@analog.com>
Signed-off-by: Bryan Wu <cooloney@kernel.org>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>

.. fix all the worst problems in -rc1

If CONFIG_AMIGA_BUILTIN_SERIAL=m, I get the following warnings:

| drivers/char/amiserial.c: At top level:
| drivers/char/amiserial.c:2138: warning: data definition has no type or storage class
| drivers/char/amiserial.c:2138: warning: type defaults to 'int' in declaration of 'console_initcall'
| drivers/char/amiserial.c:2138: warning: parameter names (without types) in function declaration
| drivers/char/amiserial.c:2134: warning: 'amiserial_console_init' defined but not used

because console_initcall() is not defined (nor really sensible) in the
modular case.

So disable serial console support if the driver is modular.
Signed-off-by: Geert Uytterhoeven <geert@linux-m68k.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>

In commit f337b9c5 ("epoll: drop
unnecessary test") Thomas found that there is an unnecessary (always
true) test in ep_send_events().  The callback never inserts into
->rdllink while the send loop is performed, and also does the
~EP_PRIVATE_BITS test.  Given we're holding the mutex during this time,
the conditions tested inside the loop are always true.

HOWEVER.

The test "!ep_is_linked(&epi->rdllink)" wasn't there because we insert
into ->rdllink, but because the send-events loop might terminate before
the whole list is scanned (-EFAULT).

In such cases, when the loop terminates early, and when a (leftover)
file received an event while we're performing the lockless loop, we need
such test to avoid to double insert the epoll items.  The list_splice()
done a few steps below, will correctly re-insert the ones that were left
on "txlist".

This should fix the kenrel.org bugzilla entry 11831.
Signed-off-by: Davide Libenzi <davidel@xmailserver.org>
Signed-off-by: Linus Torvalds <torvalds@linux-foundation.org>