Because currently snd_printd() and snd_printdd() macros are expanded
to empty when CONFIG_SND_DEBUG=n, a compile warning like below
appears sometimes, and we had to covert it by ugly ifdefs:
  sound/pci/hda/patch_sigmatel.c: In function ‘stac92hd71bxx_fixup_hp’:
  sound/pci/hda/patch_sigmatel.c:2434:24: warning: unused variable ‘spec’ [-Wunused-variable]

For "fixing" these issues better, this patch replaces snd_printd() and
snd_printdd() definitions with empty inline functions instead of
macros.  This should have the same effect but shut up warnings like
above.

But since we had already put ifdefs, changing to inline functions
would trigger compile errors.  So, such ifdefs is removed in this
patch.

In addition, snd_pci_quirk name field is defined only when
CONFIG_SND_DEBUG_VERBOSE is set, and the reference to it in
snd_printdd() argument triggers the build errors, too.  For avoiding
these errors, introduce a new macro snd_pci_quirk_name() that is
defined no matter how the debug option is set.
Reported-by: NStratos Karafotis <stratosk@semaphore.gr>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Empty files can get deleted by the patch program, so remove empty Kbuild
files and their links from the parent Kbuilds.
Signed-off-by: NDavid Howells <dhowells@redhat.com>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

Although we've had macros defining double _RANGE controls for a while now
they've not actually been backed up properly by the implementation, it's
treated everything as mono. Fix that by implementing the handling in the
stereo controls, ensuring that the mono controls don't mistakenly get
treated as stereo.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

pop_wait is used to determine if a deferred playback close
needs to be cancelled when the a PCM is open or if after
the power-down delay expires it needs to run. pop_wait is
associated with the CODEC DAI, so the CODEC DAI must be
unique. This holds true for most CODECs, except for the
dummy CODEC and its DAI.

In DAI links with non-unique dummy CODECs (e.g. front-ends),
pop_wait can be overwritten by another DAI link using also a
dummy CODEC. Failure to cancel a deferred close can cause
mute due to the DAPM STOP event sent in the deferred work.

One scenario where pop_wait is overwritten and causing mute
is below (where hw:0,0 and hw:0,1 are two front-ends with
default pmdown_time = 5 secs):

aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE -d 1
sleep 1
aplay /dev/urandom -D hw:0,1 -c 2 -r 48000 -f S16_LE -d 3 &
aplay /dev/urandom -D hw:0,0 -c 2 -r 48000 -f S16_LE

Since CODECs may not be unique, pop_wait is moved to the PCM
runtime structure. Creating separate dummy CODECs for each
DAI link can also solve the problem, but at this point it's
only pop_wait variable in the CODEC DAI that has negative
effects by not being unique.
Signed-off-by: NMisael Lopez Cruz <misael.lopez@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Make the flag in the pdata of type bool to fix a sparse warning.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Reported-by: NFengguang Wu <fengguang.wu@intel.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add a flag to suppress the update in emu1010_firmware_thread() during
suspend/resume.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Instead of calling request_firmware() at each time, keep the obtained
firmware internally and reuse it.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Yet again like previous two commits, drop the old hwdep user-space
firmware code from vx driver (snd-vxpocket and snd-vx222).
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Current FSI driver is using platform information pointer,
but it is not good design for DT support.
This patch makes master clock selection
independent from platform information pointer.
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Current FSI driver required set_rate() platform callback function
to set audio clock if it was master mode,
because it seemed that CPG/FSI-DIV clocks calculation depend on
platform/board/cpu.
But it was calculable regardless of platform.
This patch supports audio clock calculation method,
but the sampling rate under 32kHz is not supported at this point.
Old type set_rate() is still supported now,
but it will be deleted on next version
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add the possibility to specify a gpio through platform data
so that a HW reset can be issued to the codec.
Signed-off-by: NJavier Martin <javier.martin@vista-silicon.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

For more strict protection for wild disconnections, a refcount is
introduced to the card instance, and let it up/down when an object is
referred via snd_lookup_*() in the open ops.

The free-after-last-close check is also changed to check this refcount
instead of the empty list, too.
Reported-by: NMatthieu CASTET <matthieu.castet@parrot.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

ALSA did not provide any direct means to infer the audio time for A/V
sync and system/audio time correlations (eg. PulseAudio).
Applications had to track the number of samples read/written and
add/subtract the number of samples queued in the ring buffer.  This
accounting led to small errors, typically several samples, due to the
two-step process.  Computing the audio time in the kernel is more
direct, as all the information is available in the same routines.

Also add new .audio_wallclock routine to enable fine-grain synchronization
between monotonic system time and audio hardware time.
Using the wallclock, if supported in hardware, allows for a
much better sub-microsecond precision and a common drift tracking for
all devices sharing the same wall clock (master clock).
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Keep track of boundary crossing when hw_ptr
exceeds boundary limit and wraps-around. This
will help keep track of total number
of frames played/received at the kernel level
Signed-off-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The CS4271 has a feature to sync its analog mute flags, so one mute
circuitry can be used for both channels.

Give users access to this feature with a new DT property and a flag in
the platform data.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NDavid Howells <dhowells@redhat.com>
Acked-by: NArnd Bergmann <arnd@arndb.de>
Acked-by: NThomas Gleixner <tglx@linutronix.de>
Acked-by: NMichael Kerrisk <mtk.manpages@gmail.com>
Acked-by: NPaul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: NDave Jones <davej@redhat.com>

Convert #include "..." to #include <path/...> in kernel system headers.
Signed-off-by: NDavid Howells <dhowells@redhat.com>
Acked-by: NArnd Bergmann <arnd@arndb.de>
Acked-by: NThomas Gleixner <tglx@linutronix.de>
Acked-by: NPaul E. McKenney <paulmck@linux.vnet.ibm.com>
Acked-by: NDave Jones <davej@redhat.com>

This patch adds support for Dialog semiconductor's DA9055 audio codec.

This has been tested on DA9055 EVB with Samsung SMDK6410 board.
Signed-off-by: NAshish Chavan <ashish.chavan@kpitcummins.com>
Signed-off-by: NDavid Dajun Chen <david.chen@diasemi.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Allow regulators managed via DAPM to make use of the bypass support that
has recently been added to the regulator API by setting a flag
SND_SOC_DAPM_REGULATOR_BYPASS. When this flag is set the regulator will
be put into bypass mode before being disabled, allowing the regulator to
fall into bypass mode if it can't be disabled due to other users.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Passing struct snd_dma_buffer pointer instead, so that they work no
matter whether real SG buffer is used or not.

This is a preliminary work for the HD-audio DSP loader code.
Signed-off-by: NIan Minett <ian_minett@creativelabs.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The 'dres' field (discharge resistance for headphone outputs) is no longer
used in the driver, so remove it.

It was used in the original version of the driver when entering standby
from off, but we stopped using it when we switched from having a single
startup sequence to having separate cap and capless sequences.
Signed-off-by: NTimur Tabi <timur@freescale.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

If the LRCLK is shared and the WM8960 is clock master then we should
enable the LRCM bit to tell the device that it should drive LRCLK when
either ADC or DAC is enabled rather than separately driving the two
LRCLKs.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

For ENUM controls the bitmask is calculated based on the number of items.
Currently this is done each time the control is accessed. And while the
performance impact of this should be negligible we can easily do better. The
roundup_pow_of_two macro performs the same calculation which is currently done
manually, but it is also possible to use this macro with compile time constants
and so it can be used to initialize static data. So we can use it to initialize
the mask field of a ENUM control during its declaration.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Acked-by: NPeter Ujfalusi <peter.ujfalusi@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NVinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Signed-off-by: NMauro Carvalho Chehab <mchehab@redhat.com>

Add support for tuning AM (on devices with the necessary additional
hardware components), and advertise the available bands using the new
VIDIOC_ENUM_FREQ_BANDS ioctl.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
CC: Ondrej Zary <linux@rainbow-software.org>
Signed-off-by: NMauro Carvalho Chehab <mchehab@redhat.com>

For following the standard, define more channel map positions and
shuffle the items a bit:

- As both PulseAudio and gstreamer define MONO channel position
  explicitly, we should follow that, too.  The mono streams point to
  this channel position unless they are explicitly assigned to certain
  channel positions.

- Top-front-* and Top-rear-* positions are added, carried from
  PulseAudio's definitions.

- Move NA and MONO definitions at the top of table right after
  UNKNOWN, since these are more abstract in comparison with other
  practical positions.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This will be used to enable additional control of the regulators.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

There is already a set of channel position definitions in alsa-lib
mixer.h, and it'd be more practical to keep the same order for the
PCM channel map, too.  The value is shifted with 1 to keep zero for
UNKNOWN.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

ALC650 has a channel swap option between surround and CLFE channels,
so we need to tweak the channel maps dynamically depending on the
register bit.

Now struct snd_ac97 can contain chmap pointers for playback and
capture.  The driver may store these and let ac97 driver changing the
channel mapping dynamically.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch implements the basic data types for the standard channel
mapping API handling.

- The definitions of the channel positions and the new TLV types are
  added in sound/asound.h and sound/tlv.h, so that they can be
  referred from user-space.

- Introduced a new helper function snd_pcm_add_chmap_ctls() to create
  control elements representing the channel maps for each PCM
  (sub)stream.

- Some standard pre-defined channel maps are provided for
  convenience.
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Since bypass paths aren't part of DAPM streams and we may not have any
DAPM streams there may not be anything that triggers a DAPM sync for
them. Mark all input and output widgets as dirty and then sync to do so
at the end of suspend and resume.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by: NLiam Girdwood <lrg@ti.com>

The only user was removed over two years ago in commit a6c65736 ("ASoC: Remove
current PGA control handling").
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Move the Tegra+WM8903 ASoC platform data header out of
arch/arm/mach-tegra, as a pre-requisite of single zImage.
Signed-off-by: NStephen Warren <swarren@nvidia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Remove the main ALSA version number from the kernel ALSA driver.
The ALSA driver package release diverges from the upstream. This may
confuse users to see the same ALSA version for many kernel releases
and this version lost it's original purpose and connection.

The "ioctl" APIs have own version numbers, so the user space may check
for specific API changes only.
Signed-off-by: NJaroslav Kysela <perex@perex.cz>

The WM0010 is a compact digital signal processor that has been
highly optimised for low-power audio applications.  Extensive memory
resources and core optimisation allow the device to manage all audio
processing algorithms efficiently and autonomously, while the host
processor sleeps or performs other tasks.
Signed-off-by: NDimitris Papastamos <dp@opensource.wolfsonmicro.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Sometimes the analogue circuitry connected to the microphone needs some
time to settle after power up. Allow systems to configure this delay in
the platform data, the driver will then insert the required delay during
power up of paths that involve the microphone.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Here we update the asoc structures to add compress stream definations
First the struct snd_soc_dai_driver adds a new member to indicate if the dai is
compressed or pcm. Next we add a new structre the struct snd_soc_compr_ops in
the struct snd_soc_dai_link. This is to be used for machine driver to perform
any opertaions required for setting up compressed audio streams

next is the compressed data operations, they are added using struct
snd_compr_ops in the struct snd_soc_platform_driver.
Signed-off-by: NNamarta Kohli <namartax.kohli@intel.com>
Signed-off-by: NRamesh Babu K V <ramesh.babu@intel.com>
Signed-off-by: NVinod Koul <vinod.koul@linux.intel.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Implement suspend/resume support for AD1816 chips.
Tested with Terratec SoundSystem Base-1.
Signed-off-by: NOndrej Zary <linux@rainbow-software.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

struct snd_card_ad1816a is only set but the values are never used then.
Removing it allows struct snd_card's private_data to be used for
struct snd_ad1816a, simplifying the code.
Signed-off-by: NOndrej Zary <linux@rainbow-software.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>