patch suppresses the warning message "control load not supported"
as this is a debug information to help debug issues in topology.
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Some BIOS have inconsistent dev_type value for DMIC link type.
Since there is only one device type for DMIC link type, remove device
type check if link type is NHLT_LINK_DMIC.
Signed-off-by: Guneshwor Singh <guneshwor.o.singh@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Patch fixes wrong path in commit 0b06122f ("ASoC: Intel: kbl: Add
map for new DAIs for Multi-Playback & Echo Ref") which resulted in pop
noise.
Current topology for Headset results in unwanted pop noise, while
switching from spk->hs at the start of Headset Playback.
Hence re-introduced mixin-mixout dsp module in topology for headset
playback pipe to fix the regression.
And the corresponding modification for headset route is updated here.

Fixes: 0b06122f ("ASoC: Intel: kbl: Add map for new DAIs for
Multi-Playback & Echo Ref")
Signed-off-by: Naveen Manohar <naveen.m@intel.com>
Signed-off-by: Sathya Prakash M R <sathya.prakash.m.r@intel.com>
Acked-By: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Symbol SND_SOC_INTEL_SST_TOPLEVEL is user selectable so add the
help text for this symbol.
Signed-off-by: Vinod Koul <vinod.koul@intel.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Merge remote-tracking branches 'asoc/topic/tfa9879', 'asoc/topic/ts3a277e', 'asoc/topic/wm8741', 'asoc/topic/wm97xx' and 'asoc/topic/zte' into asoc-next

Merge remote-tracking branches 'asoc/topic/samsung', 'asoc/topic/sgtl5000', 'asoc/topic/stm32', 'asoc/topic/sunxi' and 'asoc/topic/tas571x' into asoc-next

Merge remote-tracking branches 'asoc/topic/rt5645', 'asoc/topic/rt5651', 'asoc/topic/rt5659', 'asoc/topic/rt5663' and 'asoc/topic/rt5670' into asoc-next

Merge remote-tracking branches 'asoc/topic/max98925', 'asoc/topic/max98927', 'asoc/topic/msm8916' and 'asoc/topic/omap' into asoc-next

Merge remote-tracking branches 'asoc/topic/hdmi', 'asoc/topic/img' and 'asoc/topic/kirkwood' into asoc-next

Merge remote-tracking branches 'asoc/topic/dwc', 'asoc/topic/fallthrough', 'asoc/topic/fsi', 'asoc/topic/fsl' and 'asoc/topic/graph' into asoc-next

Merge remote-tracking branches 'asoc/topic/cygnus', 'asoc/topic/da7213', 'asoc/topic/davinci' and 'asoc/topic/doc' into asoc-next

Merge remote-tracking branches 'asoc/topic/bcm2835', 'asoc/topic/cs42l56' and 'asoc/topic/cs43130' into asoc-next

Merge remote-tracking branches 'asoc/topic/ac97', 'asoc/topic/ac97-mfd', 'asoc/topic/amd' and 'asoc/topic/arizona-mfd' into asoc-next

ASoC: Fixes for v4.14

I've been quite lax in sending these due to conference season but here's
a fairly large collection of ASoC updates.  The one thing that's not
device specific is Takashi's fix for races between delayed work and PCM
destruction, otherwise everything is specific to an individual device.

# gpg: Signature made Thu 26 Oct 2017 15:11:23 BST
# gpg:                using RSA key ADE668AA675718B59FE29FEA24D68B725D5487D0
# gpg:                issuer "broonie@kernel.org"
# gpg: Good signature from "Mark Brown <broonie@sirena.org.uk>" [unknown]
# gpg:                 aka "Mark Brown <broonie@debian.org>" [unknown]
# gpg:                 aka "Mark Brown <broonie@kernel.org>" [unknown]
# gpg:                 aka "Mark Brown <broonie@tardis.ed.ac.uk>" [unknown]
# gpg:                 aka "Mark Brown <broonie@linaro.org>" [unknown]
# gpg:                 aka "Mark Brown <Mark.Brown@linaro.org>" [unknown]
# gpg: WARNING: This key is not certified with a trusted signature!
# gpg:          There is no indication that the signature belongs to the owner.
# Primary key fingerprint: 3F25 68AA C269 98F9 E813  A1C5 C3F4 36CA 30F5 D8EB
#      Subkey fingerprint: ADE6 68AA 6757 18B5 9FE2  9FEA 24D6 8B72 5D54 87D0

DSP modes and left/right justified modes can be supported
on bcm2835 by configuring the frame sync polarity and
frame sync length registers and by adjusting the
channel data position registers.

Clock and frame sync polarity handling in hw_params has
been refactored to make the interaction between logical
rising/falling edge frame start and physical configuration
(changed by normal/inverted polarity modes) clearer.

Modes where the first active data bit is transmitted immediately
after frame start (eg DSP mode B with slot 0 active)
only work reliable if bcm2835 is configured as frame master.
In frame slave mode channel swap (or shift, this isn't quite
clear yet) can occur.

Currently the driver only warns if an unstable configuration
is detected but doensn't prevent using them.
Signed-off-by: Matthias Reichl <hias@horus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

bcm2835's configuration registers can't be changed when a stream
is running, which means asymmetric configurations aren't supported.

Channel and rate symmetry are already enforced by constraints
but samplebits had been missed.

As hw_params doesn't check for symmetry constraints by itself
and just returns success if a stream is running this led to
situations where asymmetric configurations were seeming to
succeed but of course didn't work because the hardware wasn't
configured at all.

Fix this by adding the missing samplerate symmetry constraint.
Signed-off-by: Matthias Reichl <hias@horus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Sample rates are only restricted by the capabilities of the
clock driver, so use SNDRV_PCM_RATE_CONTINUOUS instead of
SNDRV_PCM_RATE_8000_192000.

Tests (eg with pcm5122) have shown that bcm2835 works fine
in 384kHz/32bit stereo mode, so change the maximum allowed
rate from 192kHz to 384kHz.
Signed-off-by: Matthias Reichl <hias@horus.com>
Reviewed-by: Eric Anholt <eric@anholt.net>
Signed-off-by: Mark Brown <broonie@kernel.org>

bcm2835 supports arbitrary positioning of channel data within
a frame and thus is capable of supporting TDM modes. Since
the driver is limited to 2-channel operations only TDM setups
with exactly 2 active slots are supported.

Logical TDM slot numbering follows the usual convention:

For I2S-like modes, with a 50% duty-cycle frame clock,
slots 0, 2, ... are transmitted in the first half of a frame,
slots 1, 3, ... are transmitted in the second half.

For DSP modes slot numbering is ascending: 0, 1, 2, 3, ...

Channel position calculation has been refactored to use
TDM info and moved out of hw_params.

set_tdm_slot, set_bclk_ratio and hw_params now check more
strictly if the configuration is valid. Illegal configurations
like odd number of slots in I2S mode, data lengths exceeding
slot width or frame sizes larger than the hardware limit of
1024 are rejected. Also hw_params now properly checks for
errors from clk_set_rate.

Allowed PCM formats are already guarded by stream constraints,
thus the formats check in hw_params has been removed and
data_length is now retrieved via params_width().

Also standard functions like snd_soc_params_to_bclk are now
being used instead of manual calculations to make the code
more readable.

Special care has been taken to ensure that set_bclk_ratio works
as before. The bclk ratio is mapped to a 2-channel TDM config
with a slot width of half the ratio. In order to support odd ratios,
which can't be expressed via a TDM config, the ratio (frame length)
is stored and used by hw_params.
Signed-off-by: Matthias Reichl <hias@horus.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Add mclk-fs support to audio graph card
as it was previously implemented in simple card.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Add mclk-fs support to audio graph card
as initially supported in simple card.
Signed-off-by: Olivier Moysan <olivier.moysan@st.com>
Acked-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

The main rt5514 driver optionally calls into the SPI back-end to load
the firmware. This causes a link error when one driver selects rt5514
as built-in and another driver selects rt5514-spi as a loadable module:

sound/soc/codecs/rt5514.o: In function `rt5514_dsp_voice_wake_up_put':
rt5514.c:(.text+0xac8): undefined reference to `rt5514_spi_burst_write'

As a workaround, this adds another silent symbol, to force rt5514-spi
to be built-in for that configuration. I'm not overly happy with
that solution, but couldn't come up with anything better. Using
'IS_REACHABLE()' would break the case that relies on the loadable
module, and all other ideas would result in more complexity.
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>

The new functions are only used when CONFIG_PM is enabled,
leading to a harmless warning:

sound/soc/codecs/rt5514-spi.c:474:12: error: 'rt5514_resume' defined but not used [-Werror=unused-function]
sound/soc/codecs/rt5514-spi.c:464:12: error: 'rt5514_suspend' defined but not used [-Werror=unused-function]

This marks them as __maybe_unused to make the build silent
again.

Fixes: 58f1c07d ("ASoC: rt5514: Voice wakeup support.")
Signed-off-by: Arnd Bergmann <arnd@arndb.de>
Signed-off-by: Mark Brown <broonie@kernel.org>

Check the JD status in the button pushing to prevent the IRQ that is locked
by button pushing event while the jack unpluging.
Signed-off-by: Oder Chiou <oder_chiou@realtek.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Before rendering starts, DMA driver copies full buffer valid data
to ACP SRAM for the first time, after that ACP SRAM to I2S
FIFO DMA will be initiated. After rendering first half of ACP SRAM,
IOC will be raised then Audio data will be copied from first half of
System Memory to first half of ACP SRAM. Similarly after rendering
second half of ACP SRAM, IOC will be raised then Audio Data will be
copied from second half of the System Memory to second half of the
ACP SRAM in ping-pong way till rendering stops.

Old design introducing latency issues resulting stutter sound observed
during playback.
Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda@amd.com>
Signed-off-by: Akshu Agrawal <Akshu.Agrawal@amd.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

Minimum time required between power On of codec and read
of RT5645_VENDOR_ID2 is 400msec. We should wait that long
before reading the value.

TEST=Cold boot the device and check for sound device.
Signed-off-by: Akshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: Bard Liao <bardliao@realtek.com>
Signed-off-by: Alex Deucher <alexander.deucher@amd.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

32bit and 24bit audio capture formats for H3/H2+ are broken because the
RX_SAMPLE_BITS and the RX_FIFO_MODE bits of AC_ADC_FIFOC register of the audio
codec are not set to operate in 24bit mode but in 16bit mode only.
The following patch sets the H3 audio codec registers and the DMA bus width
properly when a 24/32bit capture is requested.
Signed-off-by: Andrea Bondavalli <andrea.bondavalli74@gmail.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

DSP modes are documented in the data sheet but not enabled in the driver.
The work-around already implemented for DA7218/9 is also required to
make sure the bit clock handling in DSP modes follows ASoC conventions.

Tested with ARD-AUDIO-DA7212 and Minnowmax Turbot boards
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Acked-by: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

The current code might be a bit intriguing without having experienced the
issue before, and might come up as a mistake.

Make explicit what's going on by adding a comment.
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>

While the current code was reporting to be able to work in master mode, it
failed to do so because the BCLK divider wasn't programmed, meaning that
the BCLK would run at the PLL's frequency no matter the sample rate.

It was obviously a bit too fast.

Add support to retrieve the divider to use, and set it. Since our PLL is
not always able to generate a perfect multiple of the sample rate, we'll
have to choose the closest divider that matches our setup.

Fixes: 36c68493 ("ASoC: Add sun8i digital audio codec")
Reviewed-by: Chen-Yu Tsai <wens@csie.org>
Signed-off-by: Maxime Ripard <maxime.ripard@free-electrons.com>
Signed-off-by: Mark Brown <broonie@kernel.org>
Cc: <stable@vger.kernel.org>