The tdfb_controls.m4 is changed for better 15 degree control enum
spacing for beam direction. The file is with this change dedicated
for line arrays with physical limitation to -90 to +90 degrees that
is just achievable with 16 max. enum values.
The new tdfb_controls360.m4 uses the previous 30 degree spacing for
360 degrees arrays such as circular. The pipeline macros for 360
degrees are added to pipelines macros. The difference is only include
of 360 degrees controls version.
The topology development CMakeList.txt is changed to build topologies
with TDFB with single dual mono -90 to +90 degree angle steerable
beam. The stereo dual beam setup files are preserved and updated but
currently used only for testbench test for more complex configuration.
Signed-off-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
In order to avoid inter core communication while running
GOOGLE_RTC_AUDIO_PROCESSING, this change ensure that the speaker
processing and the amplifier feedbacks are running on the same core as
the AEC.
Signed-off-by: Lionel Koenig <lionelk@google.com>
Remove the setup config info from all initial config data in the
codec_adapter kcontrol's. This is no longer needed.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Remove the setup config info from all initial config data in the
codec_adapter kcontrol's. This is no longer needed.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Add support for boards without speaker amplifier. Just simply remove
all speaker related components and configuration.
Signed-off-by: Brent Lu <brent.lu@intel.com>
detect pipelines require PCM_ID to be define which is the case only
when they get inserted using PIPELINE_PCM_ADD.
Note that PIPELINE_PCM_ADD and PIPELINE_ADD have different signature.
Signed-off-by: Lionel Koenig <lionelk@google.com>
Pipeline extects PIPELINE_FORMAT to be defined. When including a
pipeline using DAI_ADD, DAI_FORMAT is defined but no PIPELINE_FORMAT.
Prior c687815, the last previously defined PIPELINE_FORMAT was used.
This change ensure the requested DAI_FORMAT is used in the pipeline.
This addresses bug #5193.
Signed-off-by: Lionel Koenig <lionelk@google.com>
So far we support apl, glk, cml, jsl, tgl.
SSP0 or SSP2 can be used.
DMICS may or may not be present, the simplifying assumption is that we
have 0, 2 or 4channels.
the existing topology names are kept but will not be used with kernel
updates, the -ssp<N> and -dmic<N> ch suffixes will be added based on
NHLT information to avoid a need to override topology files.
A better solution would be a DAI index that can be overridden in a
topology file, but we have no interface to do so at the moment, so
brute-force combinatorial handling of all possible solutions it is.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
In order to ease comprehension and avoid side-effect of constants being
defined after calling a macro, this change undefined the locally define
constants for pipelines and dai macros.
Signed-off-by: Lionel Koenig <lionelk@google.com>
All .tplg output files have been compared and are strictly identical
after the change.
The deprecation warnings were added more than one year ago in
https://github.com/alsa-project/alsa-lib/commits//706192341d1d0bbb906
Now that we just upgraded our Docker image to ALSA 1.2.6
(https://hub.docker.com/r/thesofproject/sof/tags) so #5153 can enable
topology v2, the volume of warnings has became unbearable. For instance
good luck trying to find the actual error messages for the build
failures of #5155 - they're totally drowned in these deprecation
warnings.
Signed-off-by: Marc Herbert <marc.herbert@intel.com>
1.Default support 1ms period capture pipeline to update host position
more precisely.
2.Revise pipeline 1 to playback and pipeline 4 to capture in comment.
Signed-off-by: YC Hung <yc.hung@mediatek.com>
Introduce a new component to perform acoustic echo cancellation
on capture path using a buffer from the playback as a reference.
1. Put the google_rtc_audio_processing at
$SOF_REPO/third_party_libraries
- libgoogle_rtc_audio_processing.a
- libgoogle_rtc_audio_processing_tuning.a
- libc++.a (Corresponding stdlibc++)
- libc++abi.a
2. Put the header in $SOF_REPO/third_party_includes
- google_rtc_audio_processing.h
3. Build firmware and tool with xcc
4. To verify it works:
- aplay some speech
- At the same time arecord the mic which uses AEC
The mic signal should not exhibit any echo from the playback.
Signed-off-by: Lionel Koenig <lionelk@google.com>
1. Default support 1ms period playback pipeline to update host position
more precisely.
2. Revise pcm node number description
Signed-off-by: YC Hung <yc.hung@mediatek.com>
Add support for four max98357a speaker amplifiers running in TDM mode
which format is 8 slots with 32 bit slot/sample width on ADL boards.
The only difference between this one and sof-adl-max98360a-rt5682-4ch
is the SSP port for speaker amplifiers; this one is using SSP2 while
max98360a's topology is using SSP1.
This topology implements a 4-channel pipeline directly to speaker
amplifiers so audio effects need to be done in user space.
Signed-off-by: Brent Lu <brent.lu@intel.com>
In a previous commit DTS changed the period of this topology to 1000 when `DTS` was defined. This
commit removes this behaviour as DTS codec now supports the original period for this topology
Signed-off-by: Mark Barton <mark.barton@xperi.com>
The loudness EQ is changed to 250 taps to create large IPC
message type. It also improves subjective quality of this
effect with better bass response. The earlier version was
made very short to fit the that time IPC size limit.
Signed-off-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Convert playback pipeline on analog output to have a mixer so adding a
deep-buffer pipeline will be simpler.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
The same volume information is present twice, remove the one that
seems out of place in the pipeline.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
mix/merge of parts coming from pipe-mixer-volume-dai-playback.m4 and
pipe-eq-iir-eq-fir-volume-playback.m4
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
This topology was used initially on UpExtreme but isn't shipped in any
commercial device.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
To include deep buffer for HDaudio topologies, we need the ability to
replace volume by some other component (e.g. eq-iir-eq-fir-volume)
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Add support for four max98360a speaker amplifiers running in TDM mode
which format is 8 slots with 32 bit slot/sample width on ADL boards.
This topology implements a 4-channel pipeline directly to speaker
amplifiers so audio effects need to be done in user space.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Move generated *.conf and *.tplg v1 files down from:
build_tools/topology/topology1/*.{conf,tplg}
_to:
build_tools/topology/topology1/production/*.{conf,tplg}
... then copy/"install" the production/* subdirectory two levels up.
This fixes the race condition #5067 that also copied a random number of
development/ and dsp_enhancement/ topologies, sometimes even truncating
these.
In other words, this commit REMOVES the following two copies:
build_tools/topology/development/ # randomly corrupted copy, removed
build_tools/topology/dsp_enhancement/ # randomly corrupted copy, removed
build_tools/topology/topology1/development/ # real build dir, unchanged
build_tools/topology/topology1/dsp_enhancement # real build dir, unchanged
Production topologies are copied only to help with the v1->v2
transition. That duplication makes the build directory confusing enough,
no need to extend that copy to development topologies. A single instance
of development topologies in the build directory is enough.
This removal may break some CI script(s): this is perfect because such
CI script(s) were copying randomly corrupted data.
Signed-off-by: Marc Herbert <marc.herbert@intel.com>
The recent changes to the string parser in alsa-lib cause the topology
builds to break for some topologies. Avoid adding a newline for the bytes
data for the MUXDEMUX config by introducing a new macro for creating lists
without new lines between items.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
When the string fields left empty, the alsa topology parser treats
SND_CONFIG_TYPE_STRING type config as -EINVAL, so set defauls values.
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
Add support for four max98360a speaker amplifiers running in TDM mode
which format is 8 slots with 32 bit slot/sample width on ADL boards.
To implement the 2-way woofer/tweeter speaker function in SOF, there
is a demux to create 4-channel audio data with EQ on each channel for
band-split function.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Topology file could use PIPELINE_FILTER1 macro to include a m4 file
with eq coefficient. If macro is not defined, eq_iir_coef_bandsplit.m4
will be included.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Change the default slot mapping from L_lo-L_hi-R_lo-R_hi to
L_lo-R_lo-L_hi-R_hi so slot 0/1 is for woofer and slot 2/3 is for
tweeter.
Signed-off-by: Brent Lu <brent.lu@intel.com>
The demux routing matrix and config are removed for two reasons: 1.
the config 'demux_priv_1' is hardcoded for pipeline 1 only. 2. other
m4 file with demux compoenet like pipe-volume-demux-playback has the
matrix and config defined in the topology file.
The topology which implement this pipeline should define the routing
matrx and config named as 'demux_priv_<pipeline id>' before including
this m4 file.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Modify RTNR to work with KWD on TGL and ADL.
Remove unused RTNR 16kHz topology.
Add RTNR support to max98390 on ADL.
Signed-off-by: Ming Jen Tai <mingjen_tai@realtek.com>
No functional change, only fix an ASCII-art topology representation
in a comment.
Signed-off-by: Guennadi Liakhovetski <guennadi.liakhovetski@linux.intel.com>
Fix the documentation to match actual topology definition. The comments
did not include addition of mixers nor the capture PCMs. Also many
comments reflect a fixed core id that is no longer accurate.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
The DTS Codec can be conditionally added to the sof-mt8195-mt6359-rt1019-rt5682 topology on those pipelines intended for headphone and speaker output.
To enable processing and configure settings, additional bytes must be sent to the control via ALSA TLV. The intention is that endusers will use ALSA UCM to achieve this.
Signed-off-by: Mark Barton <mark.barton@xperi.com>
Commit 771db86de2 ("topology1: codec_adapter: Add 'codec_adapter' pipeline configuration")
in an attempt to support PCM + compr mixer scenarios broke simple codec adapter
pipelines.
So, similar with PCM case (see sof/pipe-host-volume-playback.m4 vs
sof/pipe-volume-playback.m4) we introduce two separate configuration
files.
One for Passthrough codec-adapter pipeline and one standalone Host +
codec-adapter pipeline that can be independently scheduled in a scenario
with a mixer for example.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
As opposed to pipe-codec-adapter-playback this has its own scheduling
task and can be used in more complex scenarios.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Support SKU 0B29 product, the audio hardware configuration is rt711
on link2, two rt1316s on link0 and link1, rt714 on link3.
Signed-off-by: Gongjun Song <gongjun.song@intel.com>
Support SKU 0AF3 product. The audio hardware configuration is two
rt1316s on link1 and link2, and rt714 on link0.
Signed-off-by: Gongjun Song <gongjun.song@intel.com>
This patch adds support for three configurations:
SSP1 connects max98390 2/4 speakers
SSP2 connects max98390 2 speakers
The SSP TDM configuration uses 4 slots for playback and
4 slots for the echo reference capture - regardless of the number of speakers.
UCM files in userspace specify which channels needs to be used on the specific platform.
There is no information reported by the topology/firmware related to valid channels.
Max98390 uses following channels mapping for playback and EchoRef capture
Chan 0 = Left
Chan 1 = Right
Chan 2 = Tweeter Left
Chan 3 = Tweeter Right
(Chan 0 and 1 with regular speakers;Chan 2 and 3 with tweeter speakers)
Addition:
Add SSP2 BT_OFFLOAD support
Signed-off-by: Mac Chiang <mac.chiang@intel.com>
Add support for cs42l42 with max98360a running on JSL boards. The
cs42l42 needs to enable bclk earlier in prepare stage and disable bclk
at hw_free statge so we add the SSP_CC_BCLK_ES flag for it.
Signed-off-by: Brent Lu <brent.lu@intel.com>
This topologies are for i.MX8QXP/i.MX8QM, i.MX8MP demonstrating
mixing for 1 PCM stream and 1 Compr stream.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Add support for max98360a running on ADL boards. We choose SSP1 for
speaker amplifier for BT offload compatibility. Also increase the
sample depth to 32 bits for more dynamic range and avoid using m/n
counter.
We add the flag SSP_CC_BCLK_ES to SSP0 for the compatibility with
CS42L42 in the future.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Add m4 and headers to support basic topology with passthrough
pipeline for ACPSP and DMIC use cases support on renoir platform.
Signed-off-by: Ajit Kumar Pandey <AjitKumar.Pandey@amd.com>
Signed-off-by: balapati <balakishore.pati@amd.com>
Comments for calls to PIPELINE_PCM_ADD and DAI_ADD describe parameters
in the same order they are passed to the macros. The only exception is
order of "priority" and "core", and this can be very misleading. In
most cases the actual current values for the two parameters are 0,
making it even easier to make a mistake when modifying them.
Fix the order in the comments to match the actual order in which the
parameters should be passed to PIPELINE_PCM_ADD and DAI_ADD.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Updated guidance for BT hardware is to use 38.4Mhz MCLK for both
SCO and A2DP mode. This applies for all Intel platforms supporting
Bluetooth offload.
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
This adds 'codec_adapter' pipeline configuration allowing this
pipeline to be part of a more complex topology.
Important configuration here is the 'scheduler' widget. Without this,
'codec_adapter' would work in a more complex topology. This is because
each individual pipeline needs to have a 'scheduling' component.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
We plan to test the CI test topology on ADL-P nocodec device, so
add the topology build support in cmake list.
Signed-off-by: Zhang Keqiao <keqiao.zhang@intel.com>
Not sure why they were intertwined, let's separate them out to avoid
git conflicts with integration.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
This topology is used on GLK and TGL. variations include:
- DMIC supported on TGL and not on GLK
- TGL uses SSP0 instead of SSP2
APL support was added only based on user reports. This assumes the
same topology as GLK but was not tested.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
This patch adds topologies sof-apl-nocodec-demux-eq-2ch4ch.tplg and
sof-apl-nocodec-demux-eq-4ch4ch.tplg. Playback of 2ch creates 4ch
output in format L_lo, L_hi, R_lo, and R_hi. An example band-split
at 2 kHz is configured for EQ processing. The low band contains
an additional 80 Hz high-pass.
The pipeline was tested in UP2 device. The nocodec topology enables
an useful SPP loopback mode. The capture PCM is connected to DAI
loopback so this pipeline can be tested with simultaneous 2ch aplay
and 4ch arecord.
Signed-off-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
This PR adds RTNR Noise Reduction/Suppression(NR/NS) component by
Realtek Semiconductor Corp. This feature links to proprietary libraries.
Please contact antz0525@realtek.com for any question about the library.
Signed-off-by: Ming Jen Tai <mingjen_tai@realtek.com>
Without enclosing ENDPOINT_NAME with `', the generated tplg has
pcm id 0 named to "Headphones" instead of "Speakers".
$ ./tplgtool.py ~/work/sof/tools/build_tools/topology/sof-tgl-max98357a-rt5682.tplg
pcm=Headphones;id=0;type=playback;fmt=S16_LE;rate_min=48000;rate_max=48000;ch_min=2;ch_max=2;
pcm=Headset;id=1;type=both;fmt=S16_LE;rate_min=48000;rate_max=48000;ch_min=2;ch_max=2;
pcm=HDMI1;id=2;type=playback;fmt=S16_LE;rate_min=48000;rate_max=48000;ch_min=2;ch_max=2;
pcm=HDMI2;id=3;type=playback;fmt=S16_LE;rate_min=48000;rate_max=48000;ch_min=2;ch_max=2;
pcm=HDMI3;id=4;type=playback;fmt=S16_LE;rate_min=48000;rate_max=48000;ch_min=2;ch_max=2;
pcm=HDMI4;id=5;type=playback;fmt=S16_LE;rate_min=48000;rate_max=48000;ch_min=2;ch_max=2;
pcm=EchoRef;id=6;type=capture;fmt=S16_LE;rate_min=48000;rate_max=48000;ch_min=2;ch_max=2;
pcm=DMIC;id=99;type=capture;fmt=S32_LE;rate_min=48000;rate_max=48000;ch_min=4;ch_max=4;
pcm=DMIC16kHz;id=100;type=capture;fmt=S16_LE;rate_min=16000;rate_max=16000;ch_min=2;ch_max=2;
$ grep Headphones topology/sof-tgl-max98357a-rt5682.conf -n -B5
7071-# PCM Low Latency, id 0
7072:SectionPCM."Headphones" {
7073-
7074- # used for binding to the PCM
7075- id "0"
7076-
7077: dai."Headphones 0" {
7078- id "0"
7079- }
Signed-off-by: Yong Zhi <yong.zhi@intel.com>
For speakers SSP2 link is used.
For Headset SSP0 link is used.
Adding required macros to select SSP and Platform.
Signed-off-by: Vamshi Krishna Gopal <vamshi.krishna.gopal@intel.com>
Modify sof-jsl-rt5682 topology so Waves codec can be added
to playback in case 'WAVES' is defined.
Small refactoring
Signed-off-by: Oleh Titov <Otitov@waves.com>
The patch to fix CHANNELS_MIN side effects had a bug in which caused
LOCAL_CHANNELS_MIN to slip into conf output. Example from
sof-adl-sdw-max98373-rt5682:
---cut--
SectionPCMCapabilities."Passthrough Capture 14" {
» formats "S16_LE"
» rate_min "8000"
» rate_max "16000"
» channels_min "LOCAL_CHANNELS_MIN"
--cut--
Fixes: 659266685b ("topology: remove side effects from macro definitions")
BugLink: https://github.com/thesofproject/sof/issues/4621
Signed-off-by: Kai Vehmanen <kai.vehmanen@linux.intel.com>
Dynamic range compression (DRC) or simply compression is an
audio signal processing operation that reduces the volume of
loud sounds or amplifies quiet sounds, thus reducing or
compressing an audio signal's dynamic range.
This enables the DRC component for i.MX8QM/i.MX8QXP/i.MX8MP
with wm8960 codec.
Signed-off-by: Iuliana Prodan <iuliana.prodan@nxp.com>
This commit fixes the issue mentioned in #4583.
A temporary definition is introduced that takes it's value from
"CHANNELS_MIN" if it is already defined, otherwise it is set to
a default value. This temporary is used instead of "CHANNELS_MIN"
for the rest of the file. This avoids having the macro defined in
files where it shouldn't be.
In order to be completely sure it avoids side effects this temporary
is undefined after it is no longer needed.
Signed-off-by: Bud Liviu-Alexandru <budliviu@gmail.com>
ControlBytes section name is given by DEF_EQFIR_COEF macro.
Otherwise, using pipe-eq-fir-volume-playback.m4 in a topology results in
a compilation error.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Add macro to allow configuration of min channels in PCM_CAPABILITIES.
The default behavior is not changed.
So far min channels was hardcoded to 2 for pipe-volume-playback but we
need mono for i.MX8ULP configuration.
Notice that we need to use the local macro TCHANNELS_MIN because we
don't want to modify the value of CHANNELS_MIN macro outside
of sof/pipe-volume-playback.m4 file.
Doing so will cause unpredictable behavior for next pipeline
definitions.
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Convert all playback pipelines to have a mixer so adding a deep-buffer
pipeline will be simpler.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
This enables EQ component with IIR filter for i.MX8QM/i.MX8QXP/i.MX8MP
with wm8960 codec.
Users can try various configurations for EQ parameters using sof-ctl tool:
$ amixer -Dhw:1 controls | grep EQ
numid=44,iface=MIXER,name='EQIIR1.0 eqiir_coef_1'
$ ./sof-ctl -Dhw:1 -n 44 -s sof/tools/ctl/eq_iir_bassboost.txt
Signed-off-by: Daniel Baluta <daniel.baluta@nxp.com>
Add support for reuse of sof-tgl-rt711-rt1308 on Google Volteer. This
is not intended for production but removes both hotwording and
smart-amp support - this simpler topology is intended for
SoundWire/DMIC tests only.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
The topologies used for sof-tgl-rt711-rt1308 and
sof-tgl-max98357a-rt5682 are almost equivalent.
Conditionally add support for the amplifier reference dailink and BT
offload, so that this topology can be used as is on Google
Volteer. This will help root-cause suspend/resume issues we see only
on TGL_RVP_SDW
Related kernel PR: https://github.com/thesofproject/linux/pull/3006
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
This patch contains updates to add beam on/off and direction controls.
The common m4 tasks to add controls are placed to include files
tdfb_controls.m4, tdfb_defines.m4, tdfb_undefines.m4.
The single beam examples are replaced by blobs with 0, +/-30, and +/-90
degree angled beams. The Arrays are 50m, 68mm spaced for typical
notebook microphones. The 28mm example is for four microphones. A four
microphones for notebook with microphones at 0, 36, 146, and 182mm
line locations.
Signed-off-by: Seppo Ingalsuo <seppo.ingalsuo@linux.intel.com>
Add a deepbuffer pipelines connected to the the mixer-dai
pipeline for SSP0 and SSP2. The pipeline deadlines are left at
1ms for now and will be changed later after the mixer pipelines
are validated.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
capture
S32_LE without PGA is not supported in alsabat, switch to use
pipe-volume-capture and unblock the CI alsabat test case.
Signed-off-by: Iris Wu <xiaoyun.wu@intel.com>
The use of s32le did not expose any problems on APL, but alsa-bat was
previously reported as failing on JSL. Now that this test was extended
to CML_NOCODEC, we see the same issue. Manual tests with s24le show no
issues.
Let's just use s24le across the board and move on.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Enable the bclk clock control for SSP2.
Note that this impacts existing GLK-based chromebooks as well as newer
hardware.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Set clks_control to (SOF_DAI_INTEL_SSP_CLKCTRL_MCLK_ES |
SOF_DAI_INTEL_SSP_CLKCTRL_BCLK_ES) to enable MCLK/BCLK early start
feature.
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
Define the SSP_CC_MCLK/BCLK_ES bit to be used in SSP_CONFIG_DATA macro
to enable mclk/bclk on hw_params and disable malk/bclk on hw_free.
Signed-off-by: Brent Lu <brent.lu@intel.com>
Signed-off-by: Bard Liao <yung-chuan.liao@linux.intel.com>
For some reason s32le does not work for DAI definitions on JSL-NOCODEC
platforms. alsa-bat tests fail, but they work fine on APL, CNL,
TGL. This is likely to be an ICL platform issue. The root cause is
still TBD
Use s24le for now to unlock Intel daily tests.
All other platforms remain with s32le.
BugLink: https://github.com/thesofproject/sof/issues/4427
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Move the existing sof-apl-nocodec to the development folder, in case
SOF CI still wants to use it.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Not all devices have 4 cores, some only have 2 and even APL/GLK is
currently limited to a single core.
For now we still use a single core for all topologies, we will enable
multi-core in a follow-up patch.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Prepare for reuse across all platforms. For now this still uses
single-core.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
We first want to enable the simplified topologies, then multi-core
then dynamic pipelines. The latter two cases will be handled in
follow-up patches.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
There is no rt1308 on the TGL-H-RVP platform, need to add a topology
file with only rt711 to enable soundwire on the TGL-H-RVP platform.
The patch of enable soundwire on the TGL-H-RVP platform has been
merged into the thesofproject/linux.
Signed-off-by: Gongjun Song <gongjun.song@intel.com>
Remove the addition of DAPM routes for virtual widgets.
These are not needed for suppressing errors with legacy
machine drivers. Just adding the virtual widget is enough.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
The UpExtreme HAT connector with the SoundWire,
I2S, DMIC mixed mode exposes the following pins
DMIC_DATA0: input: HAT PIN 8
DMIC_CLK: output: HAT PIN 26
IS2_MCLK: output: HAT PIN 16
I2S1_SCLK: output: HAT PIN 32
I2S1_SFRM: output: HAT PIN 10
I2S1_TXD: output: HAT PIN 24
I2S1_RXD: input: HAT PIN 18
Let's use core3 for DMIC, core2 for SSP0, core1 for SSP1 and core0 for
SSP2.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
In preparation for Topology2.0, move the current topology files
to the topology1 folder and once the 1.0 topologies are
built copy them to the /sof/tools/build-tools/topology folder.
When Topology2.0 topologies come along, they will be built into
the topology2 folder and the 2.0 binaries will be copied over
the 1.0 binaries.
Signed-off-by: Ranjani Sridharan <ranjani.sridharan@linux.intel.com>
Seppo Ingalsuo