Current ASoC has register function for platform/codec/dai/card,
but doesn't have for cpu.
It often produces confusion and fault on ASoC.

As result of ASoC community discussion,
we consider new struct snd_soc_component for CPU/CODEC,
and will switch over to use it.

This patch adds very basic struct snd_soc_component,
and register function for it.
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>
Reviewed-by: NStephen Warren <swarren@nvidia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This patch adds support for the Dialog DA7213 audio codec.
Signed-off-by: NAdam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

this add new API for sound compress to support gapless playback.
As noted in Documentation change, we add API to send metadata of encoder and
padding delay to DSP. Also add API for indicating EOF and switching to
subsequent track

Also bump the compress API version
Signed-off-by: NJeeja KP <jeeja.kp@intel.com>
Signed-off-by: NVinod Koul <vinod.koul@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Add a basic header file for the TI AESS IP block, located in the OMAP4
Audio Back-End subsystem.

Currently, this header file only contains a function to enable the
AESS internal clock auto-gating.  This will be used by a subsequent
patch to ensure that the AESS won't block the entire chip
low-power-idle mode.  We wish to be able to place the AESS into idle
even when no AESS driver has been compiled in.
Signed-off-by: NPaul Walmsley <paul@pwsan.com>
Cc: Liam Girdwood <lrg@ti.com>
Cc: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Péter Ujfalusi <peter.ujfalusi@ti.com>
Cc: Tony Lindgren <tony@atomide.com>
Acked-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Help avoid noise from the power up of the capture path propagating through
into the start of the recording (especially noise caused by the ramp of
microphone biases) by keeping the capture muted until after we've finished
powering things up with DAPM in the same manner we do for playback. This
allows us to take advantage of soft mute support in the hardware more
effectively and is more consistent.

The core code using the existing digital mute operation is updated to take
advantage of this. Some additional cases in the soc-pcm code and suspend
will need separate handling but these are less practically relevant than
the main runtime stream start/stop case.

Rather than refactor the digital mute function in every single driver a
new operation is added for drivers taking advantage of this functionality,
the old operation should be phased out over time.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
Acked-by Vinod Koul <vinod.koul@intel.com>
Acked-by: NLiam Girdwood <liam.r.girdwood@linux.intel.com>

This patch completes the replacement of the existing max98090 driver,
by installing a more complete driver.
Signed-off-by: NJerry Wong <jerry.wong@maximintegrated.com>
Tested-by: NMatthew Mowdy <matthew.mowdy@maximintegrated.com>
Reviewed-by: NRalph Birt <ralph.birt@maximintegrated.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Convert MicBias widgets to supply widget.

On tlv320aic3x, Mic bias power on/off shares the same register bits
with output mic bias voltage.  So, when power on mic bias, we need
reclaim it to voltage value.

Provide a new platform data so that the micbias voltage can be sent
according to board requirement. Now since tlv320aic3x codec driver
is DT aware, update dt files and functions to handle this new
"micbias-vg"  platform data.

Because of sharing of bits, when enabling the micbias, voltage also
needs to be updated. So use SND_SOC_DAPM_POST_PMU & SND_SOC_DAPM_PRE_PMD
macro to create an event to handle this.

Since micbias is converted to supply widget, updated machine drivers as
well.

This change is runtime tested on da850-evm with audio loopback
(arecord|aplay) for confirmation.
Signed-off-by: NHebbar Gururaja <gururaja.hebbar@ti.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Current soc-dai.h defines SND_SOC_DAIFMT_GATED as (2 << 4),
but gated clock should be default settings (= 0).
This patch fixup SND_SOC_DAIFMT_GATED as (0 << 4).
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Some audio drivers are calling snd_dma_continuous_data(GFP_KERNEL)
which makes "sparse" give a warning:

  $ make C=2 M=sound/usb modules
    ...
  sound/usb/6fire/pcm.c:625:25: warning: cast from restricted gfp_t
  sound/usb/caiaq/audio.c:845:41: warning: cast from restricted gfp_t
  sound/usb/usx2y/usbusx2yaudio.c:997:54: warning: cast from restricted gfp_t
  sound/usb/usx2y/usbusx2yaudio.c:1001:54: warning: cast from restricted gfp_t
  sound/usb/usx2y/usx2yhwdeppcm.c:774:54: warning: cast from restricted gfp_t
  sound/usb/usx2y/usx2yhwdeppcm.c:778:54: warning: cast from restricted gfp_t

Add __force to the cast to silence the warning.
Signed-off-by: NAntonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch adds snd_soc_of_parse_daifmt() and supports below style on DT.

        [prefix]format = "i2c";
        [prefix]clock-gating = "continuous";
        [prefix]bitclock-inversion;
        [prefix]bitclock-master;
        [prefix]frame-master;

Each driver can use specific [prefix]
(ex simple-card,cpu,dai,format = xxx;)

This sample will be
SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_CONT |
SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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>

Current soc-dai.h defines SND_SOC_DAIFMT_NB_NF as (1 << 8),
but normal bit clock / normal frame should be
default settings (= 0).
This patch fixup SND_SOC_DAIFMT_NB_NF as (0 << 8).
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The core does not modify these fields, so they can be made const. This allows
drivers to declare their op tables as const.
Signed-off-by: NLars-Peter Clausen <lars@metafoo.de>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Current simple-card driver calls asoc_simple_card_dai_init()
if platform had a asoc_simple_card_dai_init pointer.
And, this initialization function works only
when platform has an applicable initial value for each dai settings.
And basically, almost all sound card requires certain initialization.
This means that almost all platform has initialization settings,
and driver do nothing if it doesn't have settings.

And additionally, current simple-card supports sysclk settings but it was
only for codec.  In order to abolish deviation between cpu and codec,
and in order to simplify processing,
this patch adds asoc_simple_dai, and removed pointless
struct asoc_simple_dai_init_info which was trigger of
calling asoc_simple_card_dai_init().
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

All MXS users have been converted to device tree and the board files have been
removed.

No need to keep platform data in the driver.
Signed-off-by: NFabio Estevam <fabio.estevam@freescale.com>
Acked-by: NDong Aisheng <dong.aisheng@linaro.org>
Acked-by: NShawn Guo <shawn.guo@linaro.org>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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>

FSI driver's flag usage was changed/removed by
3449f5fa
(ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support)
ab6f6d85
(ASoC: fsi: add master clock control functions)

And unused flags had been removed on FSI driver,
but the definition had been kept to avoid compile error.

It is possible to cleanup sh_fsi.h now.
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

3449f5fa
(ASoC: fsi: add SND_SOC_DAIFMT_INV_xxx support)
added clock inversion support via snd_soc_dai_set_fmt().
Thus, this patch removed SH_FSI_xxx_INV and fsi_get_info()
from fsi driver, and modified platform settings to use new style.
Then, it cleaned up meaningless settings from platform.
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: NSimon Horman <horms+renesas@verge.net.au>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

ab6f6d85
(ASoC: fsi: add master clock control functions)
added driver level clock control functions.
And now, platform depended .set_rate() is no longer needed.
This patch removed unnecessary .set_rate() platform callback support.
Signed-off-by: NKuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The CS4271 requires its LRCLK and MCLK to be stable before its RESET
line is de-asserted. That also means that clocks cannot be changed
without putting the chip back into hardware reset, which also requires
a complete re-initialization of all registers.

One (undocumented) workaround is to assert and de-assert the PDN bit
in the MODE2 register.

This patch adds a new flag to both the DT bindings as well as to the
platform data to enable that workaround.
Signed-off-by: NDaniel Mack <zonque@gmail.com>
Acked-by: NAlexander Sverdlin <subaparts@yandex.ru>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Since we are now using the clock API integration to manage MCLK we can now
use clk_get_rate() to determine if we need to divide MCLK without relying
on platform data.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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>