Do not rewrite the whole register, but only update the needed
bits in set_dai_sysclk functions.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Driver for Texas Instruments TLV320DAC33 (SLAS546) low power stereo
audio DAC.

TLV320DAC33 is a stereo audio codec with integrated 24KB FIFO for low
power audio playback.

The digital interface can use I2S, DSP (A or B), Right and Left
justified formats.
DAC33 has stereo analog input, which can be bypassed to the analog
outputs.

Regarding to the internal 24KB FIFO the driver implements 'FIFO bypass'
mode (default) and nSample mode (FIFO is in use).
a) In 'FIFO bypass' mode the internal FIFO is not in use, the codec is
working synchronously as a normal codec (it needs constant stream of
data on the digital interface).

b) The nSample mode implementation uses one interrupt line from DAC33 to
the host:
Alarm threshold is set to 10ms of audio data (limit by the driver
implementation).
DAC33 will signal an interrupt, when the FIFO level goes under the
Alarm threshold.
The host will write to nSample register a value (number of stereo
samples), to tell DAC33 how many samples it should read in a burst from
the host. When the DAC33 received the number of samples, it disables the
clocks on the I2S bus. When the FIFO use again goes under the Alarm
threshold, DAC33 signals the host with an interrupt, and the process is
repeated.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NIgor Grinberg <grinberg@compulab.co.il>
Signed-off-by: NMike Rapoport <mike@compulab.co.il>
CC: Mark Brown <broonie@opensource.wolfsonmicro.com>
CC: alsa-devel@alsa-project.org
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The PM core will grow pm_link infrastructure in 2.6.33 which can be
used to implement the intended functionality of the ASoC-specific
device suspend and resume callbacks so drop them.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Remove the <plat/audio.h> include from arch/arm/plat-s3c/include/plat/audio.h
as it provides nothing to the current kernel and is not in any future plans
for the system.
Signed-off-by: NBen Dooks <ben@simtec.co.uk>
Signed-off-by: NSimtec Linux Team <linux@simtec.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Access to damp_power_widgets() is assumed to be single-threaded.
Concurrent accesses to dapm_power_widgets() may result in
unpredictable behavior.

Calls from:
close_delayed_work()
soc_codec_close()
soc_pcm_prepare()
soc_suspend()
soc_resume_deferred()
to snd_soc_dapm_stream_event() do not have the codec->mutex
taken to cover the call to dapm_power_widgets(). Thus, take
the mutex in these paths also to assure single-threaded use
of dapm_power_widgets().
Signed-off-by: NEero Nurkkala <ext-eero.nurkkala@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The power for the amplifier should be handled internally
by the tpa6130a2 driver.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

- Staticise ttpa6130a2_client.
- Remove unneeded cast from void.
- Use explict NULL rather than 0.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Driver for Texas Instruments TPA6130A2 stereo headphone
amplifier.

The driver provides playback gain control and also pre-defined
DAPM_HP widgets and DAPM routings for power management.

The DAPM_HP widget names are:
"TPA6130A2 Headphone Left"
"TPA6130A2 Headphone Right"

From soc machine drivers to use with the tpa6130a2 amplifier,
the tpa6130a2_add_controls has to be called, which adds the alsa
controls and the DAPM routing needed for the tpa6130a2.
After that the machine driver can connect the codec's output
with 'TPA6130A2 Left' and 'TPA6130A2 Right':

        {"TPA6130A2 Left", NULL, "CODEC LEFT OUT"},
        {"TPA6130A2 Right", NULL, "CODEC RIGHT OUT"},

Internally the left and right channels are powered separately.
When none of the channels are needed the amplifier is powered
down:
hard power: valid GPIO number is passed within platform data
soft power: Using the software shutdown of the amplifier
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The modified revision of at91sam9g20 Evaluation Kit rev. C and onwards share
with previous ones its audio connexion to Wolfson wm8731. Modify the SoC file
to extend the machine ID checking.
Signed-off-by: NNicolas Ferre <nicolas.ferre@atmel.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Cc: stable@kernel.org

These should be handled via set_tdm_slot() now and cause build
failures as-is.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Sometimes it is desirable to have a mux which does not reflect any
direct register configuration but which will instead only have an
effect implicitly (for example, as a result of changing which parts
of the device are powered up). Provide a virtual mux for this purpose.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Don't assume that enumerations are backed by registers when updating
mux power.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

We don't need to check for an event callback since we also check for
an appropriate event flag when applying mux status changes.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fix for typo in commit 8d50e447
ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs
Signed-off-by: NJonathan Cameron <jic23@cam.ac.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Change the way the debugfs entries are created:
If the codec->dev is valid, than use:
debugfs/asoc/{codec->name}.{dev_name(codec->dev)}/

if the codec->dev is NULL:
debugfs/asoc/{codec->name}/

as root for the debugfs entries.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Since the SND_SOC_DAPM_LINE can be input or output, additional check is
needed in order to determine if the widget is connected as input or
output.
When checking for connected outputs, if the widget is line, than check
if the sources list is not empty (line is connected as output)
For input endpoint check, when the widget is line, also check if the
sinks list is not empty (line is connected as input).
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

In order to support multiple codecs on the same system in the debugfs
the directory hierarchy need to be changed by adding directory per codec
under the asoc direcorty:

debugfs/asoc/{dev_name(socdev->dev)}-{codec->name}/codec_reg
                                                  /dapm_pop_time
                                                  /dapm/{widgets}

With the original implementation only the debugfs files are only
created for the first codec, other codecs loaded later would fail to
create the debugfs files (since they are already exist).
Furthermore in this situation any of the codecs has been removed, would
cause the debugfs entries to disappear, regardless if the codec, which
created them are still loaded (the one which loaded first).
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Some of the Blackfin options don't directly follow the kconfig options
they depend on, so kconfig is unable to display the proper tree.  So sort
the options such they expand/collapse properly.
Signed-off-by: NBarry Song <barry.song@analog.com>
Signed-off-by: NMike Frysinger <vapier@gentoo.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Fix the missing argument of snd_soc_dai_set_pll() in neo1973_*.c,
which was forgotten in the commit 85488037.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

wm8940 requires I2C.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Fix a couple of typos and a missing header file inclusion to build wm8711.c
properly with CONFIG_SPI_MASTER.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This is also shared with newer CODECs.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Don't use a static for WM8974 PLL factors - we don't support more than
one device so it won't happen but no sense in leaving the race condition
hanging around.  Also, pre_div is a single bit and it's a bit simpler if
we move the handling of the factor of 4 in the output into the
coefficient setup.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The DMA params for McASP with FIFO has been updated so that it works for
various FIFO levels. A member- 'fifo_level' has been added to the DMA
params data structure. The fifo_level can be adjusted by the tx[rx]_numevt
platform data. This is relevant only for DA8xx/OMAP-L1xx platforms. This
implementation has been tested for numevt values 1, 2, 4, 8.
Signed-off-by: NChaithrika U S <chaithrika@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

When running in TDM mode there can be more than 2 channels used. Datasheet has
figures for upto 8 channels so increase max_channels on all SSP interfaces to
this figure.
Signed-off-by: NGraeme Gregory <dp@xora.org.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

McASP write FIFO registers should be modified for playback and read FIFO
registers for capture. Check the PCM mode before manipulating the
FIFO registers. Currently, irrespective of playback/capture both the
FIFOs are enabled or disbaled. This resulted in errors in audio loopback
mode.
Signed-off-by: NChaithrika U S <chaithrika@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch removes references to cpu_dai->dma_data.
It makes struct davinci_pcm_dma_params part of
struct davinci_mcbsp_dev or struct davinci_audio_dev.

It removes the unused name variable from davinci_pcm_dma_params.
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

When both playback and capture stream were open
davinci_i2s_hw_params was setting parameters for
the wrong stream. The fix for davinci_i2s_hw_params
is sufficient, but it looks like a race still happens
in davici_pcm_open. This patch also makes the race smaller
but the next patch provides a better fix.
Signed-off-by: NTroy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NCliff Cai <cliff.cai@analog.com>
Signed-off-by: NBarry Song <barry.song@analog.com>
Signed-off-by: NMike Frysinger <vapier@gentoo.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Since the active field of the dai already tells us the stream activity,
the local counter variable is redundant and can be replaced.
Signed-off-by: NBarry Song <barry.song@analog.com>
Signed-off-by: NMike Frysinger <vapier@gentoo.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

When MONOMIX is set to Stereo, Left PGA was not powered on but should be.
Add a mapping from Capture Left Mux to Capture Left Mixer to fix the issue.
Signed-off-by: NPhil Vandry <vandry@TZoNE.ORG>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

1. delete redundant assignment to bus field in spi_driver structure
2. fix lost assignment to set_bias_level entry in ad1938 codec dai
3. change spi driver name of ad1836 from "ad1836-spi" to "ad1836"
Signed-off-by: NBarry Song <barry.song@analog.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NJoe Perches <joe@perches.com>
Signed-off-by: NJiri Kosina <jkosina@suse.cz>

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

New machine driver for WM8580 I2S i/f on SMDK64XX.
By default SoC-Slave is set and WM8580 is configured to use it's
PLLA to generate clocks from a 12MHz crystal attached to WM8580.

[Added dependency on BROKEN since the IISv4 interface hasn't been merged
yet, fixed the PLL API usage and removed the disabling of the PLL in the
hw_free function since that'll break simultaneous playback and record
 -- broonie.]
Signed-off-by: NJassi <jassi.brar@samsung.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Instead of always returnig pointer to the 'audio-bus' clock,
check which clock is used to generate internal clocks and
then return it's pointer.
Signed-off-by: NJassi <jassi.brar@samsung.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

McASP register settings are not correct for DSP mode of operation.
There is a channel swap initally. This patch provides fixes to
the register values for proper working.

Tested on DA830/OMAP-L137 EVM, DM6467 EVM.
Signed-off-by: NChaithrika U S <chaithrika@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

If the sound system hasn't been utilized yet and we suspend, then we
attempt to save/restore using state that doesn't exist.  So use a global
handle instead to reconfigure properly.
Signed-off-by: NCliff Cai <cliff.cai@analog.com>
Signed-off-by: NMike Frysinger <vapier@gentoo.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>