This change adds a Block class to reduce the need for std::vector<std::vector<std::vector<float>>>. This make the code
easier to read and less error prone.
It also enables future changes to the underlying data structure of a
block. For instance, the data of all bands and channels could be stored
in a single vector.
The change has been verified to be bit-exact.
Bug: webrtc:14089
Change-Id: Ied9a78124c0bbafe0e912017aef91f7c311de2ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262252
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36968}
This CL adds functionality in AEC3 that allows the computational
complexity to be reduced when the output of APM is not used.
Bug: b/177830919
Change-Id: I08121364bf966f34311f54ffa5affbfd8b4db1e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/211341
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33476}
Internal counters in the RenderDelayBuffer can slip out of sync with external counters, leading to buffer misalignment.
This CL gives the RenderDelayBuffer an opportunity to update its counters.
Tested:
Passes: modules_unittests --gtest_filter=BlockProcessor.*
Fails as expected due to new unit test: modules_unittests --gtest_filter=BlockProcessor.* --force_fieldtrials="WebRTC-Aec3RenderBufferCallCounterUpdateKillSwitch/Enabled/"
audioproc_f with default AEC settings has been verified to be bit-exact on a large number of aecdumps.
Bug: webrtc:11803
Change-Id: I9363b834c8c8c934add0335013df60bf131da4bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180126
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31795}
This CL enables extracting the linear AEC output,
allowing for more straightforward
testing/development.
Bug: b/140823178
Change-Id: I14f7934008d87066b35500466cb6e6d96f811688
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153672
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29789}
- RenderBuffer::Spectrum() loses its channel argument, allowing for
greater flexibility in passing the multi-channel spectrum data into
functions.
- The FFT spectra lengths are made compile-time constant, rendering
some DCHECKs obsolete.
Bug: webrtc:10913
Change-Id: Ied0c50cf72d974cfef7279fd2b9c572d049b8b16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157104
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29528}
Multichannel signals are downmixed to mono before decimation and
delay estimation. This is useful when not all channels play
audio content. The feature can be toggled in the AEC3 configuration.
Bug: webrtc:10913
Change-Id: I7d40edf7732bb51fec69e7f3ca063d821c5069c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151762
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29126}
All channels are populated by RenderDelayBuffer. but all other
dependent modules are hardcoded to do their regular mono processing
on the first channel.
Bug: webrtc:10913
Tested: Bitexactness on a large set of aecdumps
Change-Id: I11d11aa0ad3da0f244c0ec020d2c9f0f4a735834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151640
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29079}
The VectorBuffer and MatrixBuffer names are too generic for their use case.
Bug: webrtc:10913
Change-Id: Ideecd0d27e07487a85a61dc6474e69733d07dcd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151602
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29076}
All dependent modules are hardcoded to do their regular mono processing on the first channel.
This _almost_ makes RenderBuffer multi-channel: FftData is still only mono.
Bug: webrtc:10913
Change-Id: Id5cc34dbabfe59e1cc72a9675dc7979794e870ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151139
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29074}
This is a reland of a66395e72f
Original change's description:
> Reland "Add core multi-channel pipeline in AEC3 This CL adds basic the basic pipeline to support multi-channel processing in AEC3."
>
> This is a reland of f3a197e553
>
> Original change's description:
> > Add core multi-channel pipeline in AEC3
> > This CL adds basic the basic pipeline to support multi-channel
> > processing in AEC3.
> >
> > Apart from that, it removes the 8 kHz processing support in several
> > places of the AEC3 code.
> >
> > Bug: webrtc:10913
> > Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> > Commit-Queue: Per Åhgren <peah@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29017}
>
> Bug: webrtc:10913
> Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29034}
Bug: webrtc:10913
Change-Id: Id8da5666df8c86f290c73ad5dc9958199f1a7ebe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151127
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29042}
This is a reland of f3a197e553
Original change's description:
> Add core multi-channel pipeline in AEC3
> This CL adds basic the basic pipeline to support multi-channel
> processing in AEC3.
>
> Apart from that, it removes the 8 kHz processing support in several
> places of the AEC3 code.
>
> Bug: webrtc:10913
> Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29017}
Bug: webrtc:10913
Change-Id: Ifc4b13bd994cfd22dca8f8755fa5700617cc379d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151124
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29034}
This reverts commit f3a197e553.
Reason for revert: Speculative revert, as this may'be broken some build bots
Original change's description:
> Add core multi-channel pipeline in AEC3
> This CL adds basic the basic pipeline to support multi-channel
> processing in AEC3.
>
> Apart from that, it removes the 8 kHz processing support in several
> places of the AEC3 code.
>
> Bug: webrtc:10913
> Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29017}
TBR=saza@webrtc.org,peah@webrtc.org
Change-Id: I877d2993b9ccf024bd1d57bca1513c3e24d0bed3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10913
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150940
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29022}
This CL adds basic the basic pipeline to support multi-channel
processing in AEC3.
Apart from that, it removes the 8 kHz processing support in several
places of the AEC3 code.
Bug: webrtc:10913
Change-Id: If5b75fa325ed0071deea94a7546cb4a7adf22137
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150332
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29017}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
This change makes it possible to disable AEC3's render delay
controller and delay estimator, and instead rely on an external
delay estimator. The delay is communicated via SetAudioBufferDelay.
When the feature is enabled, no echo removal will be performed
until the first delay is provided.
The delay is
Bug: b/130016532
Change-Id: I16643109d78d770ff1d2713cf247b0b9cce1bc1c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131327
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27467}
This change removes the (unused) ability of EchoRemover overriding
the delay of the RenderDelayController. The change is tested for
bit-exactness.
Bug: webrtc:8671
Change-Id: I188ef740f1437de64ffe236d07a7dcd4128192c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130518
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27414}
This change introduces a clockdrift detector operating on the estimated
delay of the echo path delay estimator. Each time the delay estimate
changes it is compared to previous estimates. If the estimates are
slowly increasing or decreasing, clockdrift is detected.
Four different patterns are considered clockdrift:
- k, k+1, k+2, k+3
- k, k+2, k+1, k+3
- k, k-1, k-2, k-3
- k, k-2, k-1, k-3
A delay estimate history matching the three last elements in one of the
patterns is considered probable clockdrift. Matching all four elements
is considered verified clockdrift.
If the delay is constant for some time after clockdrift is detected the
clockdrift detector will revert to no detected clockdrift.
The level of clockdrift is reported via an UMA histogram.
Bug: webrtc:10014
Change-Id: I1cce4d593e101a8b3fa99df6935e59b4243cb97a
Reviewed-on: https://webrtc-review.googlesource.com/c/111381
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25758}
This CL utilizes the existing, but unused, ability to set
different histogram thresholds for early and late delay
estimation. It does so by tuning the parameters for these.
On top of that, some corrections are added to correctly
handle resets and the use of the hysteresis thresholds.
Bug: webrtc:19886,chromium:896334
Change-Id: I950ac107c124541af8f02b4403f477dda71cc1a1
Reviewed-on: https://webrtc-review.googlesource.com/c/106706
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25443}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251
Change-Id: Ibdafc0bb08de1be7189af7053a67a24e3a26bd6b
Reviewed-on: https://webrtc-review.googlesource.com/89001
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23997}
This CL re-activates the explicit handling of microphone
gain changes in the AEC3 code. The implementation is done
beneath a kill-switch so that when that switch is active
the changes in this CL are bitexact.
Bug: webrtc:9526,chromium:863826
Change-Id: I58e93d8bc0bce7bec91e102de9891ad48ebc55d8
Reviewed-on: https://webrtc-review.googlesource.com/88620
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23986}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameter 'modules/audio_processing'
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: Id29f8de59dba704787c2c38a3d05c60827c181b0
Reviewed-on: https://webrtc-review.googlesource.com/83982
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23653}
This CL adds support for using any externally reported audio buffer
delay to set the initial alignment in AEC3 which is used before the
AEC has been able to detect the delay.
Bug: chromium:834182,webrtc:9163
Change-Id: Ic71355f69b7c4d5815b78e49987043441e7908fb
Reviewed-on: https://webrtc-review.googlesource.com/70580
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22917}
This CL robustifies the echo removal behavior when headsets are used.
In particular it:
-Introduces a secondary, more refined alignment when no alignment can
be found using the delay estimator.
-Changes decision logic for when to use the linear filter output.
-Changes the decision logic for when to be transparent.
-Changes the way that the transparent mode works.
-Makes the nonlinear mode less aggressive.
-Removes the detector for non-audible echoes.
-Makes the attenuation when there are signals with strong narrowband
characteristics more mild in scenarios with low render.
Furthermore the CL:
-Removes the input of external echo leakage information.
Bug: webrtc:9047,chromium:824111,webrtc:8314,webrtc:8671,webrtc:5201,webrtc:5919
Change-Id: Ied1fe0c0a35d3c31b47606ed2db319a73644d406
Reviewed-on: https://webrtc-review.googlesource.com/60866
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22548}
This CL adds functionality for passing the information about the
estimated delay to the echo remover in AEC3.
The CL also adds information about how long ago the delay changed,
and how long ago the delay estimate was updated.
Bug: webrtc:8671
Change-Id: If274ffe0465eb550f3e186d0599c6dc6fef7f5e8
Reviewed-on: https://webrtc-review.googlesource.com/55261
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22137}
This CL resets the AEC3 realignment functionality when a significant
and persistent skew in the number of render and capture API calls is
detected.
Bug: chromium:811658,webrtc:8879
Change-Id: Ib5c727b38f427da2a7d25eac7c939a17bdaabe74
Reviewed-on: https://webrtc-review.googlesource.com/52260
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21997}
The faster AEC3 alignment introduced recently may in
cases cause the alignment (and the AEC3) to repeatedly
reset. This CL avoids these resets by handling buffer
issues (which are triggering the resets) separately
during the initial coarse alignment phase.
Change-Id: Idf5e2ffda2591906da8060d03ec8ca73cdaedf53
Bug: webrtc:8798,chromium:805815
Reviewed-on: https://webrtc-review.googlesource.com/43480
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21758}
This CL modifies the refactored render buffering scheme in AEC3
so that:
-A non-causal state can never occur which means that situations with
nonrecoverable echo should not occur.
-For a stable audio pipeline with a predefined API call jitter,
render overruns and underruns can never occur.
Bug: webrtc:8629,chromium:793305
Change-Id: I06ba1c368f92db95274090b08475dd02dbb85145
Reviewed-on: https://webrtc-review.googlesource.com/29861
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21215}
This CL centralizes the render buffering in AEC3 so that all render
buffers are updated and synchronized/aligned with the render alignment
buffer.
Bug: webrtc:8597, chromium:790905
Change-Id: I8a94e5c1f27316b6100b420eec9652ea31c1a91d
Reviewed-on: https://webrtc-review.googlesource.com/25680
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20989}
The audio processing module reports the metrics 'echo return loss'
and 'echo return loss enhancement' for AEC3.
Bug: webrtc:8533
Change-Id: I166c504adf013d6cb5d6d3c9717d0622c3454bb7
Reviewed-on: https://webrtc-review.googlesource.com/24880
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20835}
This CL refactors the delay estimator in AEC3.
Furthermore, it adds:
1. Allow for a customized delay estimator behavior to simplify
development.
2. Exposes that behavior to clear configuration settings.
3. Adds logging of the delay range supported by the delay
estimator.
Bug: webrtc:8519
Change-Id: I1764a090519a78b021b2e7de565c52a6c02c848e
Reviewed-on: https://webrtc-review.googlesource.com/21166
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20733}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
