Since commit [e58aa3d2: genirq: Run irq handlers with interrupts disabled],
We run all interrupt handlers with interrupts disabled
and we even check and yell when an interrupt handler
returns with interrupts enabled (see commit [b738a50a:
genirq: Warn when handler enables interrupts]).

So now this flag is a NOOP and can be removed.
Signed-off-by: NYong Zhang <yong.zhang0@gmail.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The driver assumes that control_data points to the drivers i2c_client struct,
but this is no longer the case since the ASoC core has switched to regmap.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Call platform_device_put() instead of platform_device_unregister() if
platform_device_add() fails.
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This functionality is now subsumed within the bias management, using the
standard cache management functionality, without assuming the cache type.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

For digital only paths we need to make sure the bandgap is enabled prior
to starting the FLL which isn't tied into DAPM.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

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

Add the PCM rule to allow disabling the PCM playback SRC.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add the PCM rule to allow disabling the PCM playback SRC.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add the PCM rules to allow disabling the PCM playback and capture SRCs.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add a helper function to allow drivers to disable hardware resampling
when the application has specified the SNDRV_PCM_HW_PARAMS_NORESAMPLE
flag.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The installation of the minimum period size constraint in the PCM open
callbacks was not checked for errors.  Add this check, and move the call
to the beginning of the function to avoid having to do any cleanups in
the error case.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

twl6040 supports 5 playback, and 2 capture channels
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Only mono audio can be used for vibra (DL4 channel).
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Reset the twl6040_reg array to hold the chip default values.
The only changed values were for the microphone input selection.
Select no input for the microphones in the twl6040_init_chip function.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

There is no need to write to the vio registers at probe time, since most
them either read only, or shared with MFD or not used.
On the other hand it is a good idea to updated the ASICREV register in
the cache at this time.

After power up we need to restore some registers. Clean up the list to
contain only the registers we are going to restore.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

1. add different clkmux mode handling
SAIF can use two instances to implement full duplex (playback &
recording) and record saif may work on EXTMASTER mode which is
using other saif's BITCLK&LRCLK.

The clkmux mode could be set in pdata->init() in mach-specific code.
For generic saif driver, it only needs to know who is his master
and the master id is also provided in mach-specific code.

2. support playback and capture simutaneously however the sample
rates can not be different due to hw limitation.
Signed-off-by: NDong Aisheng <b29396@freescale.com>
Acked-by: NWolfram Sang <w.sang@pengutronix.de>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NVinod Koul <vinod.koul@linux.intel.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

The WM8996 can measure the impedance of accessories connected to the
headphone output. Implement initial support for this, measuring the
left channel impedance when an accessory is detected and using this
to distinguish between a line load and a headphone load.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

Rather than managing the bandgap in the bias level control use a supply
widget as we only actually need to enable it for analogue paths, not
fully digital ones.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

Useful for diagnostics.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

The WM8776 supports a continuous range of sample rates rather than
discrete values and supports a wider range of sample rates on the
playback path than is currently supported.  Update the constraints on
the DAIs to reflect this.
Signed-off-by: NTimur Tabi <timur@freescale.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The i2c core will clear the clientdata pointer automatically,
we don't have to set the `data' field to NULL in remove() or
if probe() failed anymore.
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The ambiguously named variable 'link' is used as a temporary throughout
davinci-pcm -- its presence makes grepping (and groking) the code
difficult.

Replace link with the value of link in almost all sites. The exception
is a couple places where the last-assigned link/chan needs to be
returned by a function -- in these cases, rename to last_link.
Signed-off-by: NBen Gardiner <bengardiner@nanometrics.ca>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: NMika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: NRyan Mallon <rmallon@gmail.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Current method for machine driver to register with the ASoC core is to
use snd_soc_register_card() instead of creating a "soc-audio" platform device.

In addition we use platform_device_register_simple() to create a platform
device for the codec. This function will handle putting and deleting the
device automatically which simplifies the error handling in the machine
driver.
Signed-off-by: NMika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: NRyan Mallon <rmallon@gmail.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

To get the PCM module loaded automatically by udev et al. we need to add a
proper MODULE_ALIAS.
Signed-off-by: NMika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: NRyan Mallon <rmallon@gmail.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Current method for machine driver to register with the ASoC core is to use
snd_soc_register_card() instead of creating a "soc-audio" platform device.
Signed-off-by: NMika Westerberg <mika.westerberg@iki.fi>
Reviewed-by: NRyan Mallon <rmallon@gmail.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Use snd_pcm_format_width() to determine the sample size, instead of
checking specify sample formats and assuming that those are the only
valid format.

This change adds support for big-endian architectures (which use the _BE
formats) and the packed 24-bit format (SNDRV_PCM_FORMAT_S24_3xE).

[Fixed single letter variable name legibility problem -- broonie]
Signed-off-by: NTimur Tabi <timur@freescale.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Roland UM-ONE midi usb interface differs from Roland UM-1.
Signed-off-by: NDaniele Guerrieri <d.guerrieri@gmail.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

We need to count the timeout down.
Reported-by: NAxel Lin <axel.lin@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

Fix below build warning:
sound/soc/blackfin/bf5xx-ad73311.c: warning: initialization from incompatible pointer type
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

clk_get() returns a pointer to the struct clk or an ERR_PTR().
This patch also use PTR_ERR() for return value.
Signed-off-by: NAxel Lin <axel.lin@gmail.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The Freescale SSI audio controller supports "synchronous" and "asynchronous"
modes.  In synchronous mode, playback and capture use the same input clock,
so sample rates must be the same during simultaneous playback and capture.
Unfortunately, the code which supports asynchronous mode is just broken in
various ways.  In particular, it was constraining sample sizes as well as
the sample rate.

The fix also allows us to simplify the code by eliminating the 'asynchronous',
'playback', and 'capture' variables that were used to keep track of playback
and capture streams.

Unfortunately, it turns out that simulataneous playback and record does not
actually work on the only platform that supports asynchronous mode: the
Freescale P1022DS reference board.  If a second stream is started, the SSI
grinds to halt for both streams.  This is true even if the P1022 is configured
for synchronous mode, so it's likely a hardware problem that needs to be
worked around.
Signed-off-by: NTimur Tabi <timur@freescale.com>
Acked-by: NLiam Girdwood <lrg@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

wait_for_avail() in pcm_lib.c has a race in it (observed in practice by an
Intel validation group).

The function is supposed to return once space in the buffer has become
available, or if some timeout happens.  The entity that creates space (irq
handler of sound driver and some such) will do a wake up on a waitqueue
that this function registers for.

However there are two races in the existing code

1) If space became available between the caller noticing there was no
   space and this function actually sleeping, the wakeup is missed and the
   timeout condition will happen instead

2) If a wakeup happened but not sufficient space became available, the
   code will loop again and wait for more space.  However, if the second
   wake comes in prior to hitting the schedule_timeout_interruptible(), it
   will be missed, and potentially you'll wait out until the timeout
   happens.

The fix consists of using more careful setting of the current state (so
that if a wakeup happens in the main loop window, the schedule_timeout()
falls through) and by checking for available space prior to going into the
schedule_timeout() loop, but after being on the waitqueue and having the
state set to interruptible.

[tiwai: the following changes have been added to Arjan's original patch:
 - merged akpm's fix for waitqueue adding order into a single patch
 - reduction of duplicated code of avail check
]
Signed-off-by: NArjan van de Ven <arjan@linux.intel.com>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Cc: <stable@kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

No code altered at this point, simply preparing for upcoming
refactorizations.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Move code from endpoint.c into a new file called stream.c and rename
functions so that their names actually reflect what they're doing.

This way, endpoint.c will be available to functions that hold all the
endpoint logic.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Sort its entries in alphabetical order.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Acked-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>