The SSP DAI now handles the clocking setup itself, all it needs is the
master clock frequency. Remove the code from Zylonite and Magician
platforms.
Signed-off-by: NDaniel Mack <daniel@zonque.org>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add the axg sound card to handle the specifities of the axg audio
sub system.

This card is required to:
 * setup the dpcm links specific to the AXG (with a cpu sound dai)
 * handle the 4 lanes masks of the tdm interfaces
 * add the loopback link when a tdm pad interface has a playback
   stream
 * handle multi-codec links
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add the DT bindings documentation for axg sound card
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Amlogic's axg card driver can't use snd_soc_of_parse_tdm_slot()
directly because it needs to handle 4 mask for each direction.
Yet the parsing of each mask is the same, so export
snd_soc_of_get_slot_mask() to reuse the the existing code.
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add Amlogic's axg TDM input driver which take the TDM signal of 4 input
lanes and push the decoded audio samples to TODDR fifo
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add Amlogic's axg tdm output driver which pulls data from FRDDR fifo
and produce the TDM signals for 4 output lanes.
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add Amlogic's axg TDM interface driver. This driver manages the format
and clocks provided on the pads.

On this SoC, each stream direction provides 4 serial lanes. This makes
a maximum of 8 channels in i2s modes and 128 channels in DSP modes.

While each lanes operate on the same slot number (same bit clock), they
may have different TDM masks. This requires to provide a function to let
the card set the 4 masks, in lieu of the usual set_tdm_slots() callback
of the dai driver.
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add Amlogic's axg TDM core driver. On this SoC, tdm is bit more
complex than usual, mainly because the different TDM input decoders can
be attached to any of TDM pad interface, including the output pads.

For the this, TDM on this SoC is modeled like this:
- TDM interface provides the DAIs the codecs will be attached to.
  The main responsibility of this driver is to manage the pad format
  and the TDM clock rates.
- TDM Formatters: These are the entities which are actually dealing with
  the TDM signal. TDMOUT produce a TDM signal from the audio sample
  provided by FRDDR using the clocks provided the TDM interface. TDMIN
  feeds TODDR with audio sample using the clocks and TDM signal provided
  by the TDM Interface.
- TDM Streams: This provides the link between 1 DAI stream of the TDM
  interface and one (or more) TDM formatters.

This driver provides the TDM formatter and TDM stream operations.
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add the DT bindings documentation for axg's TDM interfaces
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

While the two error labels "err" and "err_clk_put" goto the same place
it is rather confusing that the earlier one is certainly used later
again.
Signed-off-by: NMarcel Ziswiler <marcel.ziswiler@toradex.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add sound/soc/stm drivers entry for STM32 audio drivers from
ST Microelectronics.
Signed-off-by: NArnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add sound/soc/sti drivers entry for STI audio drivers from
ST Microelectronics.
Signed-off-by: NArnaud Pouliquen <arnaud.pouliquen@st.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

This fixes a spelling mistake.
Signed-off-by: NMarcel Ziswiler <marcel.ziswiler@toradex.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add a quirk for the "Connect Tablet 9" tablet, this tablet has a
mono-speaker. Otherwise it works fine with the defaults.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add quirk table entries for the following tablets:

ITWorks TW701
Ployer Momo7w
Trekstor win7
Yours 8"

These all use the default settings, except that they only have a single
speaker and thus need the mono-speaker quirk.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

During my initial round of bytcr_rt5651 long-name patches I did not include
a difference for mono vs stereo speaker setups in the longname because it
seems that all 5651 devices with only a single speaker do some mixing of
left + right on the PCB.

However further testing has shown that while this works great when only
playing audio on the left or right channel, the output becomes garbled
when using both channels at once. Something which does not happen when
using the Stereo DAC MIXL / MIXR switches to mix the channels together
inside the codec and then only outputting on a single channel.

So we need to have separate UCM profiles and thus separate long-names
for devices with a mono speaker vs stereo speakers. Just as we already
have for the bytcr_rt5640 case.

This commit adds a new BYT_RT5651_MONO_SPEAKER quirk and adds "stereo-spk"
or "mono-spk" to the long-name based on this and enables this mapping on
devices with a mono speaker.

Changing the long-name like this is ok for now, since I'm still working
on the UCM profiles, so they are not in upstream alsa-lib yet.

This brings the long-name naming scheme fully in sync with the bytcr_rt5640
case, which is good from a consistency pov.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

During the recent cleanup series 3 of the 6 input mappings where removed
from the bytcr_rt5651 machine driver because testing showed that none of
them were used.

However some devices do actually have their internal mic on IN2 (and
only IN2, not IN1 and IN2), this did not show during previous tests
due to a bug in the userspace UCM input device switching code.

This commit re-adds the IN2 mapping for devices with the internal mic.
on IN2 and the headser mic on IN3 and enables this mapping on devices
with their internal mic on IN2.

This commit also changes the default internal mic input to IN2, because
all my 7 test devices have their mic there.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

With the default over current detect limit of 1500uA headsets on often
get detected as headphones on the VIOS LTH17 and even when detected as
headset the OVCD current triggers often while plugged in, resulting in
false-positive button press detection.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Some boards have I2cSerialBusV2, GpioIo, GpioInt as ACPI resources, other
boards may  have I2cSerialBusV2, GpioInt, GpioIo instead. We want the
GpioIo one for the ext-amp-enable-gpio.

So far we've been assuming that the GpioIo one always comes first, this
commit adds code to detect which one comes first and to add the right
gpio-mapping.

This fixes sound not working on the Vios LTH17 laptop.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add a mixer control for the IN3 Boost volume, IN3 is used for the headset
mic on most devices, so this is necessary to control the headset mic
volume.
Signed-off-by: NHans de Goede <hdegoede@redhat.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Currently the compressed streams in DSP firmwares are
identified essentially by looking at a fixed location inside
the firmware. This is fragile and also limits things to a
single compressed stream.

Here a new form of firmware parameter is added, the HOST_BUFFER
which identifies a compressed stream from meta-data in the
firmware file. This is more robust and allows for the possiblity
of using multiple streams per core in the future. Currently the
implementation is still limited to a single stream and will
use the first HOST_BUFFER parameter encountered. If there aren't
any HOST_BUFFER parameters it will fall back to the legacy way
of finding the host buffer.
Signed-off-by: NRichard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: NCharles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Newer voice control firmwares can capture multiple audio channels.
Allow up to 8 channels for future-proofing.
Signed-off-by: NRichard Fitzgerald <rf@opensource.cirrus.com>
Signed-off-by: NCharles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Currently when creating ALSA control names for the DSP the length of any
prefix applied to the CODEC is not taken into account. Whilst this is
mostly harmless it does result in ALSA doing the truncation of the
control names and printing a warning. It is better to have the driver do
the truncation so it can truncate from the start of parameter name
itself to give a greater chance of the result maintain a unique name.
Signed-off-by: NCharles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Commit 6396bb22 ("treewide: kzalloc() -> kcalloc()") was
overlooked when doing some refactoring to the algorithm list
handling, which lead to twice as much buffer being allocated
as required for reading the algorithm list. A kcalloc is no
longer appropriate since the allocation size is now in bytes
not registers, as such change back to kzalloc.

Fixes: 7f7cca08 ("ASoC: wm_adsp: Simplify handling of alg offset and length")
Signed-off-by: NCharles Keepax <ckeepax@opensource.cirrus.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add the DT bindings documentation for axg's TDM formatters: TDMIN
and TDMOUT.
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add support for the spdif output serializer of the axg SoC family
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add the DT bindings documentation for axg's SPDIF output.
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add the capture memory interface of Amlogic's axg SoCs.
TDM, SPDIF or PDM input devices place audio samples inside this FIFO.
The FIFO content is then pushed to DDR
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add the playback memory interface of Amlogic's axg SoCs.
This device pulls data from DDR to an internal FIFO.
This FIFO is then used to feed TDM and SPDIF Output devices.
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Amlogic's axg SoCs have two types of fifos which are the memory
interfaces of the audio subsystem. FRDDR provides the playback
interface while TODDR provides the capture interface.

The way these fifos operate is very similar. Only a few settings
are specific to each.

They implement the same pcm driver here and the specifics of each
will be dealt with the related DAI driver.
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add the DT bindings documentation for axg's FIFOs: TODDR and FRDDR.
Signed-off-by: NJerome Brunet <jbrunet@baylibre.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Add documentation for power management of HDAC HDMI codec device for
various scenarios such as S0/S3, probe and playback use case.
Signed-off-by: NSriram Periyasamy <sriramx.periyasamy@intel.com>
Signed-off-by: NSanyog Kale <sanyog.r.kale@intel.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: NSrinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: NVinod Koul <vkoul@kernel.org>
Signed-off-by: NMark Brown <broonie@kernel.org>

Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: NSrinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: NVinod Koul <vkoul@kernel.org>
Signed-off-by: NMark Brown <broonie@kernel.org>

Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: NSrinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: NVinod Koul <vkoul@kernel.org>
Signed-off-by: NMark Brown <broonie@kernel.org>

Now that the component framework is integrated into the ASoC core,
remove any redundant code in this driver.
Signed-off-by: NSrinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: NVinod Koul <vkoul@kernel.org>
Signed-off-by: NMark Brown <broonie@kernel.org>

This patch aims at achieving dynamic behaviour of audio card when
the dependent components disappear and reappear.

With this patch the card is removed if any of the dependent component
is removed and card is added back if the dependent component comes back.
All this is done using component framework and matching based on
component name.
Signed-off-by: NSrinivas Kandagatla <srinivas.kandagatla@linaro.org>
Reviewed-by: NVinod Koul <vkoul@kernel.org>
Signed-off-by: NMark Brown <broonie@kernel.org>

Having interrupts enabled for ACP<->SYSMEM DMA transfer, we are in
for an interrupt storm.
For both playback and capture interrupts should be enabled for
I2S<->ACP DMA.
Signed-off-by: NAkshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Earlier, ch1 was used to define ACP-SYSMEM transfer and ch2 for
ACP-I2S transfer. With recent patches ch1 is used to define channel
order number 1 and ch2 as channel order number 2. Thus,
Playback:
ch1:SYSMEM->ACP
ch2:ACP->I2S
Capture:
ch1:I2S->ACP
ch1:ACP->SYSMEM
Signed-off-by: NAkshu Agrawal <akshu.agrawal@amd.com>
Signed-off-by: NMark Brown <broonie@kernel.org>

Patch adds Geminilake I2S machine driver which uses following codecs:
RT5682 and MAX98357A.
Signed-off-by: NNaveen Manohar <naveen.m@intel.com>
Signed-off-by: NHarsha Priya <harshapriya.n@intel.com>
Acked-by: NPierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Signed-off-by: NMark Brown <broonie@kernel.org>