The WM8960 is a low power, high quality stereo codec designed for
portable digital audio applications.

Stereo class D speaker drivers provide 1W per channel into 8W loads.
Guaranteed low leakage, excellent PSRR and pop/click suppression
mechanisms enable direct battery connection for the speaker supply.

The device also integrates a complete microphone interface and a stereo
headphone driver. External component requirements are drastically
reduced as no separate microphone, speaker or headphone amplifiers are
required. Advanced on-chip digital signal processing performs automatic
level control for the microphone or line input.

Stereo 24-bit sigma-delta ADCs and DACs are used with low power
over-sampling digital interpolation and decimation filters and a
flexible digital audio interface.

The master clock can be input directly or generated internally by an
onboard PLL, supporting most commonly-used clocking schemes.

This driver was originally written by Liam Girdwood, with substantial
subsequent additions and updates for feature completeness and changes in
the ASoC framework from me.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

SCMODE(0): Data Driven (Falling), Data Sampled (Rising), Idle State (Low)
SCMODE(1): Data Driven (Rising), Data Sampled (Falling), Idle State (Low)
SCMODE(2): Data Driven (Rising), Data Sampled (Falling), Idle State (High)
SCMODE(3): Data Driven (Falling), Data Sampled (Rising), Idle State (High)

SCMODE(3) does not invert the clock polarity compared to the default SCMODE(0).

This patch also adds all possible NF/IF, NB/IB combinations to the DSP_A and
DSP_B modes.
Signed-off-by: NDaniel Ribeiro <drwyrm@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

This ensures that we sync with the DAPM powerdown sequencing properly
and don't need to bounce the power on the voice DAC so often.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

This is simple code motion, intended to support future refactoring of
the DAPM algorithms and (more immediately) the additon of events for
DACs and ADCs.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Signed-off-by: NAlexander Beregalov <a.beregalov@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Due to the process and communications issues with the 2.6.30 S3C
platform merges none of the underlying arch/arm code for S3C64xx audio
support made it into mainline, rendering the drivers useless.  Disable
them in Kconfig to avoid user confusion - users patching in the required
support can always reenable this too.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add DSP_A interface format support by setting the LRP bit in
DSP mode.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Signed-off-by: NEric Miao <eric.miao@marvell.com>
Cc: Philipp Zabel <philipp.zabel@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

The WM8988 is a low power, high quality stereo CODEC designed for
portable digital audio applications.

The device integrates complete interfaces to 2 stereo headphone or line
out ports. External component requirements are drastically reduced as no
separate headphone amplifiers are required. Advanced on-chip digital
signal processing performs graphic equaliser, 3-D sound enhancement and
automatic level control for the microphone or line input.

The WM8988 can operate as a master or a slave, with various master clock
frequencies including 12 or 24MHz for USB devices, or standard 256fs
rates like 12.288MHz and 24.576MHz. Different audio sample rates such as
96kHz, 48kHz, 44.1kHz are generated directly from the master clock
without the need for an external PLL.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Many devices require symmetric configurations of capture and playback
data formats, often due to shared clocking but sometimes also due to
other shared playback and record configuration in the device. Start
providing core support for this by allowing the DAIs or the machine
to specify that the sample rates used should be kept symmetric.

A flag symmetric_rates is provided in the snd_soc_dai and
snd_soc_dai_link structures. If this is set in either of the DAIs or in
the machine then a constraint will be applied when a stream is already
open preventing any changes in sample rate.
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

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

Replace all DMA_24BIT_MASK macro with DMA_BIT_MASK(24)

Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

Replace all DMA_28BIT_MASK macro with DMA_BIT_MASK(28)

Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

Replace all DMA_30BIT_MASK macro with DMA_BIT_MASK(30)

Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

Replace all DMA_31BIT_MASK macro with DMA_BIT_MASK(31)

Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

Replace all DMA_32BIT_MASK macro with DMA_BIT_MASK(32)

Signed-off-by: Yang Hongyang<yanghy@cn.fujitsu.com>
Signed-off-by: NAndrew Morton <akpm@linux-foundation.org>
Signed-off-by: NLinus Torvalds <torvalds@linux-foundation.org>

Fix for compillation error introduced by the constrain patch.
Signed-off-by: NPeter Ujfalusi <peter.ujfalusi@nokia.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

Add powerdown sequence for VREF using a shared jack when the headphone
is present and the microphone isn't on.
Signed-off-by: NMatthew Ranostay <mranostay@embeddedalley.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Added an else part to check
SNDRV_MIXER_OSS_PRESENT_CVOLUME for MIC (slot 7)
in commit 36c7b833

Similarly, checks and volume control is required for
SNDRV_MIXER_OSS_PRESENT_CSWITCH and SNDRV_MIXER_OSS_PRESENT_CROUTE
as well.
Signed-off-by: NDeepika Makhija <deepika.makhija@einfochips.com>
Signed-off-by: NViral Mehta <viral.mehta@einfochips.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NAkinobu Mita <akinobu.mita@gmail.com>
Cc: <stable@kernel.org>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

To enable periods shorter than 1 ms, we have to make sure that short
periods are only available for alternate settings that have a small
enough data packet interval.  Furthermore, the code that aligns URBs to
USB frames is now superfluous.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The data packet interval needs to be available in the audioformat
structure, together with the other audio format parameters, so that it
can be used to influence ALSA hardware parameters.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This removes the check_hw_params_convention() function because
1) it is not necessary, as the hw_rule_* functions also work correctly
   (i.e., as no-ops) when the device supports all combinations of the
   audio format parameters; and
2) it would become too complex when adding a fourth altsetting-dependent
   hardware parameter, as this would require another three loops to
   check dependecies with rate/channels/format.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

When listing the device's sample formats in the stream? proc file, the
sample format number itself is rather obscure, so we better show the
format width, too.
Signed-off-by: NClemens Ladisch <clemens@ladisch.de>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

The driver should pass a device that specifies internal DMA ops, but
substream->pcm is just a logical device, and thus doesn't have arch-
specific dma callbacks, therefore following bug appears:

  Freescale Synchronous Serial Interface (SSI) ASoC Driver
  ------------[ cut here ]------------
  kernel BUG at arch/powerpc/include/asm/dma-mapping.h:237!
  Oops: Exception in kernel mode, sig: 5 [#1]
  ...
  NIP [c02259c4] snd_malloc_dev_pages+0x58/0xac
  LR [c0225c74] snd_dma_alloc_pages+0xf8/0x108
  Call Trace:
  [df02bde0] [df02be2c] 0xdf02be2c (unreliable)
  [df02bdf0] [c0225c74] snd_dma_alloc_pages+0xf8/0x108
  [df02be10] [c023a100] fsl_dma_new+0x68/0x124
  [df02be20] [c02342ac] soc_new_pcm+0x1bc/0x234
  [df02bea0] [c02343dc] snd_soc_new_pcms+0xb8/0x148
  [df02bed0] [c023824c] cs4270_probe+0x34/0x124
  [df02bef0] [c0232fe8] snd_soc_instantiate_card+0x1a4/0x2f4
  [df02bf20] [c0233164] snd_soc_instantiate_cards+0x2c/0x68
  [df02bf30] [c0234704] snd_soc_register_platform+0x60/0x80
  [df02bf50] [c03d5664] fsl_soc_platform_init+0x18/0x28
  ...

This patch fixes the issue by using card's device instead.
Signed-off-by: NAnton Vorontsov <avorontsov@ru.mvista.com>
Acked-by: NTimur Tabi <timur@freescale.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

According to the data sheet data is clocked out on the falling edge
and latched on the rising edge of the bit clock. While the left sample
is transmitted the word clock line is low.
Signed-off-by: NDaniel Glöckner <dg@emlix.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

ak4535_remove() from sound/soc/codecs/ak4535.c calls
i2c_unregister_device() with a possibly null pointer.

This bug was found by smatch (http://repo.or.cz/w/smatch.git/).
Signed-off-by: NDan Carpenter <error27@gmail.com>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>

ALC889 has two SPDIF outputs: 0x06, 0x10. Board vendors can use either or both.
DX58SO uses 0x10, but the driver assumes 0x06. The safe solution is to add
0x10 as slave output to the existing 0x06.
Reported-by: NJeroen Van Breedam <jeroen.vanbreedam@sgr5.be>
Tested-by: NJeroen Van Breedam <jeroen.vanbreedam@sgr5.be>
Signed-off-by: NWu Fengguang <fengguang.wu@intel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch increases periods_min to 6 from 4, this will remove any
hickups where the buffer is not filled fast enough from user space.
Signed-off-by: NHans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch replaces the references to bus_id to the new dev_name() API.
Signed-off-by: NHans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch replaces the references to bus_id to the new dev_name() API.
Signed-off-by: NHans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch will set the channel A and control channel mode register to
zero before disabling the AC97C peripheral.
Signed-off-by: NHans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch will enable the AC97C before resetting the external codec,
leaving the AC97C disabled will result in floating I/O lines that can
affect the reset procedure.
Signed-off-by: NHans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch will enable interrupts from AC97C and report about error
conditions that occurs.

On channel A both overrun and underrun will be enabled depending if
playback and/or capture are enabled. On the control channel the overrun
interrupt is enabled.
Signed-off-by: NHans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch will set a proper maximum bytes for the buffer, which is:
channels * bytes per sample * maximum periods * maximum bytes per period.

It also sets the minimum periods to 6, a value chosen from testing, with
a minimum of 6 periods the system has good time to fill in new audio
data without skipping.
Signed-off-by: NHans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch will take care not to overwrite OCA and ICA registers when
assigning input and output channels. It will also make sure the
registers are at a known state when enabling a channel and clean up
properly in case of an error.
Signed-off-by: NHans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

Signed-off-by: NHans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>

This patch will remove traces of channel B registers, since they are not
used by the AC97C driver. Channel B might be used for other purposes.

The driver also adds channel status bits TXEMPTY and OVRUN and a
AC97C_CH_MASK macro to ease clearing a channel settings.
Signed-off-by: NHans-Christian Egtvedt <hans-christian.egtvedt@atmel.com>
Signed-off-by: NTakashi Iwai <tiwai@suse.de>