in tlv320aic23_set_bias_level, for the case SND_SOC_BIAS_ON, the
comment says "vref/mid, osc on, dac unmute" but the code doesn't
clear the corresponding bits, thus when resuming, several bits are
not cleared preventing the codec from working.

in tlv320aic23_suspend, clearing the active register is not needed
as it will be done by tlv320aic23_set_bias_level, when setting
bias to SND_SOC_BIAS_OFF
Signed-off-by: NEric Bénard <eric@eukrea.com>
Cc: broonie@opensource.wolfsonmicro.com
Cc: anuj.aggarwal@ti.com
Cc: lrg@slimlogic.co.uk
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

After mass production started it was found that several boards exhibit
noise problems during sound playback. After some investigation it was
determined that CLKX polarity is set incorrectly, and even if most boards
can tolerate the wrong setting, there are some that don't.

Fix polarity setup in the board file. As the clock settings for input and
output now match, merge in and out functions and structures to simplify
code.
Signed-off-by: NGrazvydas Ignotas <notasas@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The config option SND_FSI_AK4642 selects SND_SOC_AK4642 which in turn
enables the compilation of ak4642.c - however this codec uses I2C to
communicate with the HW.
Same applies to DA7210.

Consequently when I2C is not set, the compilation fails [1]

This patch fixes this issues, by adding a depencdency on the related HW-
controller.
Signed-off-by: NPeter Huewe <peterhuewe@gmx.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The most useful configuration for the WM2000 is to enable the ANC so turn
that on by default, and since we're not reflecting chip default state also
enable the speaker output by default.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral
has FIFO support. This FIFO provides additional data buffering. It
also provides tolerance to variation in host/DMA controller response
times. More details of the FIFO operation can be found at

http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber=sprufm1&fileType=pdf

Existing sequence of steps for audio playback/capture are:
a. DMA configuration
b. McASP configuration (configures and enables FIFO)
c. Start DMA
d. Start McASP (enables FIFO)

During McASP configuration, while FIFO was being configured, FIFO
was being enabled in davinci_hw_common_param() function of
sound/soc/davinci/davinci-mcasp.c file. This generated a transmit
DMA event, which gets serviced when DMA is started.

https://patchwork.kernel.org/patch/84611/ patch clears the DMA
events before starting DMA, which is the right thing to do. But
this resulted in a state where DMA was waiting for an event from
McASP (after step c above), but the event which was already there,
has got cleared (because of step b above).

The fix is not to enable the FIFO during McASP configuration as
FIFO was being enabled as part of McASP start.
Signed-off-by: NSudhakar Rajashekhara <sudhakar.raj@ti.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The header contains an extern that isn't used by anything.  Remove.
Signed-off-by: NGrant Likely <grant.likely@secretlab.ca>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch is to change 'auido.h' to 'audio.h' for
fixing nuc900 alsa driver build error.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add support for i2s audio on Bluewater Systems Snapper CL15 module
Signed-off-by: NRyan Mallon <ryan@bluewatersys.com>
Acked-by: NH Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Trivial add/add fixup required in the clock table.

Conflicts:
	arch/arm/mach-ep93xx/clock.c

Add core support for EP93xx i2s audio
Signed-off-by: NRyan Mallon <ryan@bluewatersys.com>
Acked-by: NH Hartley Sweeten <hsweeten@visionengravers.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.36' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc

This patch to remove the 'break;', when the 'switch' jumps to
the 'default' branch, the 'return -EINVAL' will be return with
a error number, so the 'break;' code never be run, it is unuseful
and should be removed here.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Remove break after return, it is not needed.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Signed-off-by: NNicolas Ferre <nicolas.ferre@atmel.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add ep93xx i2s audio driver
Signed-off-by: NRyan Mallon <ryan@bluewatersys.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Upper threshold is used in mode7 of DAC33.
Instead of hard wired UTHR, add control to change the upper threshold
value.
Changing upper threshold is not allowed when the playback is already
running, since wrongly timed change in the UTHR can cause problems
with the codec.
With this control the length of the burst in mode7 can be changed.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Add the necessary files to support the TLV320AIC23B wired in I2S
on our i.MX platforms.
Signed-off-by: NEric Bénard <eric@eukrea.com>
Acked-by: NSascha Hauer <s.hauer@pengutronix.de>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>

* introduce 3 new flags to allow a more detailed configuration
of the SSI link :
	IMX_SSI_NET : enable Network Mode
	IMX_SSI_SYN : enable Synchronous Mode
	IMX_SSI_USE_I2S_SLAVE : enable I2S Slave Mode
* new platform can use these settings without breaking actual
platforms.
Signed-off-by: NEric Bénard <eric@eukrea.com>
Acked-by: NSascha Hauer <s.hauer@pengutronix.de>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The variable 'state' of structure 's3c_ac97_info' seems no use here,
so this patch is to remove the unnecessary variable.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The variable 'periods' of structure 'atmel_runtime_data'
seems no use in whole atmel alsa driver,so I make a patch
to remove the unnecessary variable.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Use the resource_size function instead of manually calculating the
resource size.This patch can reduce the chance of introducing off-by-one
errors.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NManuel Lauss <manuel.lauss@googlemail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

OMAP McBSP FIFO is word structured:
McBSP2 has 1024 + 256 = 1280 word long buffer,
McBSP1,3,4,5 has 128 word long buffer

This means, that the size of the FIFO
depends on the McBSP word size configuration.
For example on McBSP3:
16bit samples: size is 128 * 2 = 256 bytes
32bit samples: size is 128 * 4 = 512 bytes
It is simpler to place constraint for buffer and period based on channels.
McBSP3 as example again (16 or 32 bit samples):
1 channel (mono): size is 128 frames (128 words)
2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
4 channels: size is 128 / 4 = 32 frames (4 * 32 words)

Use the second method to place hw_rule on buffer size, and in threshold
mode to period size.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: NTony Lindgren <tony@atomide.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Save the word length configuration of McBSP, and use that information
to calculate, and configure the threshold in McBSP.
Previously the calculation was only correct when the stream had 16bit
audio.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: NTony Lindgren <tony@atomide.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Sicne the platform data's buffer_size now holds the full size
of the FIFO, there is no longer need to handle the ports
differently.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: NTony Lindgren <tony@atomide.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Use the actual FIFO size in words as buffer_size on OMAP3.
Change the threshold configuration to use 1 based numbering, when
specifying the allowed threshold maximum or the McBSP threshold value.
Set the default maximum threshold to (buffer_size - 0x10) intialy.
>From users of McBSP, now it is expected to use this method.
Asking for threshold 1 means that the value written to threshold registers
are going to be 0, which means 1 word threshold.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: NTony Lindgren <tony@atomide.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

Users of McBSP can use the omap_mcbsp_get_fifo_size function to query
the size of the McBSP FIFO.
The function will return the FIFO size in words (the HW maximum).
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Acked-by: NJarkko Nikula <jhnikula@gmail.com>
Acked-by: NMark Brown <broonie@opensource.wolsfonmicro.com>
Acked-by: NTony Lindgren <tony@atomide.com>
Signed-off-by: NLiam Girdwood <lrg@slimlogic.co.uk>

The size calculation is end - start + 1. But,sometimes, the '1' can
be forgotten carelessly, witch will have potential risk, so use resource_size
for {request/release}_mem_region and ioremap here should be good habit.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NDaniel Glöckner <dg@emlix.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Merge branch 'for-2.6.35' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc

This patch is to modify the ac97 delays to minimum, all these 1000 micro
seconds delays seem over spec for the AC97 interface.

I deleted some unnecessary delays here and changed the AC97 cold and warm reset
delays from 1000us to 100us.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fix a '#include "nuc900-audio.h' typo, I think it should be 'audio'.
At the same time, this patch renames the 'nuc900-auido.h' file to
'nuc900-audio.h'.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The current implement meant ACTL_ACCON was only accessed once when read or write
proceeding, which is not right, if so,we have to wait the 'timeout=0x10000' to end
every times.

We need to polling the bit AC_R_FINISH and AC_W_FINISH of ACTL_ACCON
register to identify whether read or write is finished or not,so I make
the patch to fix the issue.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch is to remove the 'SUBSTREAM_TYPE','PCM_TX' and 'PCM_RX' definition.

There is no need to redefine SNDRV_PCM_STREAM_PLAYBACK as PCM_TX,
the SUBSTREAM_TYPE(substream) can be deleted too, the playback or record can be
judged by 'if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)' directly rather
than 'if (PCM_TX == stype)', which makes the codes easy to read.
Signed-off-by: NWan ZongShun <mcuos.com@gmail.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NSascha Hauer <s.hauer@pengutronix.de>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>