Commit b2047e99 ("ASoC: hdac_hdmi: add link management")
introuduced the following build warning:

sound/soc/codecs/hdac_hdmi.c:1721:16: warning: unused variable 'timeout' [-Wunused-variable]

Remove the unused 'timeout' variable.
Signed-off-by: Fabio Estevam <fabio.estevam@nxp.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge remote-tracking branches 'asoc/topic/es8328', 'asoc/topic/find-dai', 'asoc/topic/fsl', 'asoc/topic/fsl-sai' and 'asoc/topic/fsl-ssi' into asoc-next

Merge remote-tracking branches 'asoc/topic/bcm2835', 'asoc/topic/cs42l56', 'asoc/topic/da7213', 'asoc/topic/da7218' and 'asoc/topic/da7219' into asoc-next

Merge remote-tracking branches 'asoc/topic/adsp', 'asoc/topic/ak4624', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next

Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/fsl-ssi', 'asoc/fix/rockchip' and 'asoc/fix/rt286' into asoc-linus

Happened when the Playback (or Capture) is running continuously
and Capture (or Playback) is restarted (xrun, manual stop/start...)

Since the RX (or TX) FIFO are only reset when the whole SSI is disabled,
pending samples from previous capture (or playback) session may still
be present. They must be erased to not introduce channel slipping.

FIFO Clear register fields are documented in IMX51, IMX35 reference manual.
They are not documented in IMX50 or IMX6 RM, despite they are
working as expected on IMX6SL and IMX6solo.
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

Previously, SCR.SSIEN and SCR.TE were enabled at once if no capture
stream was also running.
This may not give a chance for the DMA to write the first sample in
TX FIFO before the streaming starts on the PCM bus, inserting void
samples first.
Those void samples are then responsible for slipping the channels.
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

If the capture is already running while playback is started, it is highly
probable (>80% in a 8 channels scenario) that samples are lost between
the DMA and TX fifo.

The reason is that SIER.TDMAE is set before STCR.TFEN0, leaving a time
window where the FIFO doesn't receive the samples written by the DMA.

This particular case happened only if capture is already enabled as
SCR.SSIEN is already set at the playback startup instant.
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

Most of functions only receive the ssi_private reference and don't have
a knowledge of 'dev' pointer, even for debug purpose.
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

im6sl reference manual 47.7.4:
"
Bit clock - Used to serially clock the data bits in and out of the SSI port.
This clock is either generated internally (from SSI's sys clock) or taken
from external clock source (through the Tx/Rx clock ports).
[...]
Care should be taken to ensure that the bit clock frequency (either
internally generated by dividing the SSI's sys clock or sourced from
external device through Tx/Rx clock ports) is never greater than 1/5
of the ipg_clk (from CCM) frequency.
"

Since, in master mode, the sysclk is a multiple of bitclk, we can
easily reach a high sysclk value, whereas keeping a reasonable bitclk.

ex: 8ch x 16bit x 48kHz = 6144000, requires a 24576000 sysclk (PM=1)
    yet ipg_clk/5 = 66Mhz/5 = 13.2
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

The max number of slots in TDM mode is 32:
- Frame Rate Divider Control is a 5bit value
- Time slot mask registers control 32 slots.
Signed-off-by: Arnaud Mouiche <arnaud.mouiche@invoxia.com>
Reviewed-by: Fabio Estevam <fabio.estevam@nxp.com>
Tested-by: Caleb Crome <caleb@crome.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

Some definitions to support the PCM5102A codec
by Texas Instruments.
Signed-off-by: Florian Meier <florian.meier@koalo.de>

Changes to original patch by Florian Meier:
* rebased (Makefile and Kconfig
* fixed checkpath errors (spaces, newlines)
* added dt-binding documentation
Signed-off-by: Martin Sperl <kernel@martin.sperl.org>
Signed-off-by: Mark Brown <broonie@kernel.org>

Manage the hda idisp link using shiny new link APIs.  We need to
keep link On while we probe and also hold the reference in runtime
resume and drop in suspend
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Use shiny new link APIs to manage the links. Also remove old link
configuration logic from driver.

We need to keep link and cmd dma to off during active suspend
to allow system to enter low power state and turn it on if
the link and cmd dma was on before active suspend in active
resume.
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

The HDA links can be switched off when not is use, similarly
command DMA can be stopped as well. This calls for a reference
counting mechanism on the link by it's users to manage the link
power. The DMA can be turned off when all links are off

For this we add two APIs
	snd_hdac_ext_bus_link_get
	snd_hdac_ext_bus_link_put

They help users to turn up/down link and manage the DMA as well
Signed-off-by: Jeeja KP <jeeja.kp@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Takashi Iwai <tiwai@suse.de>
Signed-off-by: Mark Brown <broonie@kernel.org>

Broxton-P reference platform also uses combo jack for audio
connector so we need to set codec pdata to use this based on DMI
match for this board.
Signed-off-by: Ramesh Babu <ramesh.babu@intel.com>
Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

ACPI driver data can be NULL so we need to check that before
dereference the driver data.
Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Acked-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Commit 086d7f80 ("ASoC: Convert WM8962 MICBIAS to a supply widget",
2011-09-23) says:

	A supply widget is generally clearer than a MICBIAS widget and a
	mic bias is just a type of supply so use a supply widget for the
	MICBIAS. This also avoids confusion with the routing when
	connected to multiple inputs.

but this has never been documented as a policy.  Add some comments to
make it clear.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Signed-off-by: Stephen Rothwell <sfr@canb.auug.org.au>
Acked-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge branch 'topic/dai-link' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into asoc-intel

Format number after 0x in hex.

Cc: Jie Yang <yang.jie@linux.intel.com>
Cc: Liam Girdwood <lgirdwood@gmail.com>
Cc: Mark Brown <broonie@kernel.org>
Cc: Jaroslav Kysela <perex@perex.cz>
Cc: Takashi Iwai <tiwai@suse.com>
Signed-off-by: Joonas Lahtinen <joonas.lahtinen@linux.intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Current checking for PLL 32KHz mode fails in driver code when
bypassing the PLL. This is due to an incorrect check of PLL
source type when 32KHz clock is provided. Removal of this check
resolves the issue.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

This update changes the dividers used for ranges of input MCLK
frequencies, to improve PLL locking for a corner case when at edge
of MCLK frequency input divider range.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Currently PC counter is always synchronised to DAI which means that
when the DAI is disabled, features such as ALC calibration cannot
be executed successfully. This patch makes sure that when the DAI
is disabled, PC counter is set to free-running.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

When the codec is DAI clk slave, and the SRM feature of the PLL
is being used, the enabling of the DAI should occur only after
the PLL has locked to the incoming WCLK. This update adds checking
to the the DAI widget event, so it waits for SRM to lock. There is
also a timeout if that lock doesn't occur within a given time.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Currently, when Codec is I2S master DAI clocks are continuously
generated even if all audio streams have stopped. To improve
efficiency, control of the DAI clocks for master mode have been
moved to a DAPM widget event so they're only enabled as required.
Signed-off-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

This patch adds the Broxton-P machine driver for Intel Broxton-P
reference boards. This machine uses the RT298 codec
Signed-off-by: Ramesh Babu <ramesh.babu@intel.com>
Signed-off-by: Senthilnathan Veppur <senthilnathanx.veppur@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

The Broxton-P platform has 6 SSPs so we need to add ssp2 thru
ssp5 to DAI list for the driver.
Signed-off-by: Pardha Saradhi K <pardha.saradhi.kesapragada@intel.com>
Signed-off-by: Ramesh Babu <ramesh.babu@intel.com>
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Although the ES8328 does support different rates for capture and
playback, only very limited combinations are supported (8kHz and 48kHz
or 8.0182kHz and 44.1kHz) with most rates required to be symmetric.

Instead of adding a lot of complexity for little gain, let's enforce
symmetric rates.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

The values are the same for the DAC and ADC so remove the specific
values and use values with shifts.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

This is a refactor in preparation for supporting more sample sizes which
has no functional change.
Signed-off-by: John Keeping <john@metanate.com>
Signed-off-by: Mark Brown <broonie@kernel.org>