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}
This CL reduces the complexity and heap usage of the adaptive filter
in AEC3 by avoiding to compute these for the shadow
filter. In particular it
-Moves to compute the ERL, frequency response and impulse response
on an on-demand basis.
-Stores the ERL, frequency response and impulse response outside
of the adaptive filter.
All the changes have been tested for bitexactness on a sizeable
amount of recordings.
Bug: webrtc:10913
Change-Id: If83c236a6e3f2e489be129b9ebf6143a72f521d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151138
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29081}
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 CL propagates the number of render and capture channels into
the echo subtractor and the adaptive filters.
Bug: webrtc:10913
Change-Id: I5ffff24ff64b7cc0f262bf008b34b6dfca1e78f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151300
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29059}
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}
This CL removes and replaces the legacy fixed-point high-pass filter in
APM with the floating point high-pass filter in AEC3.
Bug: webrtc:10907
Change-Id: I88cf8f622ab139e4ffa97f89a72425aa3becfc58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150103
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28950}
This is a reland of 81c0cf287c
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
Bug: webrtc:10882
Change-Id: I2ddf327e80a03468c41662ae63c619ff34f2363a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150101
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28938}
This reverts commit 81c0cf287c.
Reason for revert: internal test failures
Original change's description:
> Simplification and refactoring of the AudioBuffer code
>
> This CL performs a major refactoring and simplification
> of the AudioBuffer code that.
> -Removes 7 of the 9 internal buffers of the AudioBuffer.
> -Avoids the implicit copying required to keep the
> internal buffers in sync.
> -Removes all code relating to handling of fixed-point
> sample data in the AudioBuffer.
> -Changes the naming of the class methods to reflect
> that only floating point is handled.
> -Corrects some bugs in the code.
> -Extends the handling of internal downmixing to be
> more generic.
>
> Bug: webrtc:10882
> Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28928}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: I2729e3ad24b3a9b40b368b84cb565c859e79b51e
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28931}
This CL performs a major refactoring and simplification
of the AudioBuffer code that.
-Removes 7 of the 9 internal buffers of the AudioBuffer.
-Avoids the implicit copying required to keep the
internal buffers in sync.
-Removes all code relating to handling of fixed-point
sample data in the AudioBuffer.
-Changes the naming of the class methods to reflect
that only floating point is handled.
-Corrects some bugs in the code.
-Extends the handling of internal downmixing to be
more generic.
Bug: webrtc:10882
Change-Id: I12c8af156fbe366b154744a0a1b3d926bf7be572
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149828
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28928}
This change fixes a bug where the initial delay could be set incorrectly.
Bug: webrtc:10896
Change-Id: I66b2234b69c46639488f4561e973384001230861
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149820
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28894}
This change reduces the level of several non-critical log messages in
order to reduce log spamming.
Bug: webrtc:8671
Change-Id: I6faae7a2ae4eeafd18c2770208485a75ad946e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148528
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28809}
This CL removes a long unused fallback behavior for the reverb
computation.
Bug: webrtc:8671
Change-Id: I4b57795a9bb33769237858f40392881ee235653e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148520
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28802}
As non-linear mode uses a suppressed version of y (not e) as output, this change
uses Y2, rather than E2, as nearend spectrum when computing the suppression gains.
E2 is still used in linear mode.
This change also affects how the minimum suppression gains are calculated. The
minimum gain is now min_echo_power / weighted_residual_echo.
Bug: webrtc:10550
Change-Id: I2904c5a09dd64b06bf25eb5a37c18dab50297794
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133023
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27629}
This CL corrects the minimum bound for the estimated
comfort noise level.
Bug: webrtc:10533
Change-Id: I473275ffbc7bb52572315849f30e13b764109d92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132003
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27567}
The configuration parameter filter.use_linear_filter can be used to
disable the linear filtering. Disabling the linear filter is equivalent
to runing in non-linear mode.
Bug: b/130016532
Change-Id: I8ffdf474822888b9915444bba6cc1c25ec1efe5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132552
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27566}
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 prevents FilterAnalyzer from accessing memory out-of-bounds
when the filter is resized.
Bug: chromium:946439
Change-Id: I7e2392c8b1ff0ff55566c663bf6d7a40d7754501
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129928
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27318}
In this CL bounds are added to the index used for accessing the filter frequency response. That vector has always a capacity in memory equal to the final number of blocks of the main filter. However, at the initial part of the call or after an echo path change, a transition phase is started and a filter with a lower number of blocks is used and, therefore, its size is lower than that capacity during that transition phase.
Bug: webrtc:10463
Change-Id: I6ebfdea43047a3fa993a27f2c52bb3024df84ffe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128777
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27264}
This CL removes parameters for AEC3 which are no longer used. To reflect
that change, one of the parameters also is renamed
Bug: chromium:941949,webrtc:8671
Change-Id: I26609b396fa14ecb7523eebfe531a1338718103b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127780
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27128}
This change reduces the risk of echo due to noise in the headroom
of the linear filter.
Changes:
- Use shorter delay headroom
- Delay headroom is specified in samples (not blocks)
- No hysteresis limit when delay is reduced
Bug: chromium:119942,webrtc:10341
Change-Id: I708e80e26d541dff8ca04b6da2d346a1d59cbfcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126420
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27126}
This change disables the ERLE estimation of onsets and instead assumes
minimum ERLE. This reduces the risk of echo leaks during onsets. The
estimated ERLE was sometimes incorrect due to:
- Not enough data to train on.
- Platform noise suppression can change the echo-path.
Bug: chromium:119942,webrtc:10341
Change-Id: I1dd1c0f160489e76eb784f07e99af02f44f387ec
Reviewed-on: https://webrtc-review.googlesource.com/c/123782
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26794}
This CL removes the legacy, no longer used, render buffering code. It
also removes four unused parameters from the AEC3 config. The change
is tested for bit-exactness.
Bug: webrtc:8671
Change-Id: I2bb6cb7a1097863f228767d757d551c00593bb00
Reviewed-on: https://webrtc-review.googlesource.com/c/119701
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26399}
In this CL a warning is avoided in the reverberation decay estimator code. The change is bitexact.
Bug: chromium:921582
Change-Id: I5a91f4b5970a21ba6da7254cf7fad8c2d0bcac4b
Reviewed-on: https://webrtc-review.googlesource.com/c/118441
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26342}
This reverts commit 74ba99062c.
Reason for revert: Breaks downstream project.
Original change's description:
> AEC3: Lockless transfer of render data to the capture thread
>
> This CL implements a lockless queue that replaces SwapQueue
> in the RenderWriter. This avoid stalls when the render and
> capture threads are accessing the queue at the same time.
>
> Bug: webrtc:10205
> Change-Id: Ie7d6fcf9c80fad957e2a90537658fb730ca2ed72
> Reviewed-on: https://webrtc-review.googlesource.com/c/117643
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26298}
TBR=gustaf@webrtc.org,peah@webrtc.org
Change-Id: Ie76ee8835da4e44982d181a152c9ffa19ff33e23
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10205
Reviewed-on: https://webrtc-review.googlesource.com/c/118142
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26304}
This CL implements a lockless queue that replaces SwapQueue
in the RenderWriter. This avoid stalls when the render and
capture threads are accessing the queue at the same time.
Bug: webrtc:10205
Change-Id: Ie7d6fcf9c80fad957e2a90537658fb730ca2ed72
Reviewed-on: https://webrtc-review.googlesource.com/c/117643
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26298}
Comfort noise was generated by picking random angles on the unit circle
for each frequency band and then obtaining points on the unit circle from
{cos(a), -sin(a)}.
In order to reduce complexity, this change introduces a randomly indexed
table of 32 elements over sin(a). cos(a) is obtained by adding an offset
corresponding to pi/2 to the index. The table is pre-scaled by sqrt(2) to
avoid later multiplications.
This change reduces the computational complexity of AEC3 by ~8% with no
audible degradation.
Bug: webrtc:10189
Change-Id: I8cfe2469022fb1fe910ab3f966e55d9d499b7161
Reviewed-on: https://webrtc-review.googlesource.com/c/116787
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26209}
Fixes the ENR threshold used in the dominant nearend detection when
the kill-switch WebRTC-Aec3UseLegacyNormalSuppressorTuning is pulled.
Bug: webrtc:8671,chromium:911141
Change-Id: I30ee58009633b3a9e12eff692226baada624a049
Reviewed-on: https://webrtc-review.googlesource.com/c/112903
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25880}
Correcting a mistake in the dominant nearend detection where
the meaning of the echo-to-nearend ratio was inversed.
Bug: webrtc:8671
Change-Id: I7f56369fad1784e256150c312b6b3dafcb9d0f71
Reviewed-on: https://webrtc-review.googlesource.com/c/112136
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25818}
In this CL the analysis of the impulse response that is done in the FilterAnalyzed class is changed in order to reduce its complexity. Instead of analyzing the whole impulse response in each Update call a smaller region is analyzed. That region is changed at each Update call which implies that several calls are needed in order to analyze the complete impulse response.
Bug: webrtc:10032,chromium:909007
Change-Id: Ic58be34ba18485311c63e0fed9b6e892f9cb864c
Reviewed-on: https://webrtc-review.googlesource.com/c/111602
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25817}
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}
In this CL a more precise estimation of the Erle is introduced. This is done by creating different estimators that are specialized in different regions of the linear filter. An estimation of which regions were used for generating the current echo estimate is performed and used for selecting the right Erle estimator.
Bug: webrtc:9961
Change-Id: Iba6eb24596c067c3c66d40df590be379d3e1bb7b
Reviewed-on: https://webrtc-review.googlesource.com/c/109400
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25707}
The analysis and synthesis windowing cause loss of power when
cross-fading the noise where frames are completely uncorrelated
(generated with random phase).
This CL also removes duplicate code and enables platform specific
optimizations for ARM in the comfort noise generation.
Bug: webrtc:9967,chromium:902262
Change-Id: Iffd59b301876442079d4a5f2c7fac55a3522397c
Reviewed-on: https://webrtc-review.googlesource.com/c/109581
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25526}
This change corrects the computation of the comfort noise gains.
Previously the comfort noise gain of band k, CG_k, was computed
from suppression gain of band k, SG_k, as:
CG_k = 1 - SG_k
But since the two signals are uncorrelated (the comfort noise
is randomly generated), the correct gain to maintain power is:
CG_k = sqrt(1 - SG_k^2).
Bug: webrtc:9967,chromium:902262
Change-Id: I393495742163d5e658bca4ab2f7a5067ab15af01
Reviewed-on: https://webrtc-review.googlesource.com/c/109580
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25525}
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 allows control over the dominant nearend functionality so that
it is not active during the initial phase, when estimates are less
certain.
Bug: webrtc:9906,chromium:898273
Change-Id: I5f61dac806ec3b1ebc1a3ec72f0a16d07a67f14a
Reviewed-on: https://webrtc-review.googlesource.com/c/107632
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25326}
This CL is the result of running include-what-you-use tool on part
of the code base (audio target and dependencies) plus manual fixes.
bug: webrtc:8311
Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604
Reviewed-on: https://webrtc-review.googlesource.com/c/106280
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25311}
This change makes the dominant nearend detection more accurate.
- The hangover is increased not leave nearend state between words.
- The SNR requirement is increased to not enter nearend state without
speech activity.
- An early exit mechanism has been added to leave nearend state quickly
when the echo is strong.
Bug: chromium:897701,webrtc:9897
Change-Id: I9e0f3e6ecb80eee1c0c917d4835f110555f74acf
Reviewed-on: https://webrtc-review.googlesource.com/c/107347
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25299}
In this CL we change the signal that controls the updates of the ERLE estimator. Until now, the render signal was used which is not optimum for reverberant signals. In this CL, a reverberation has been added to the the render signal and this new signal has been used for controlling when to update the ERLE estimator.
Bug: webrtc:9873
Change-Id: I0ebea3fc208f97aa237af015ba543015d49ed978
Reviewed-on: https://webrtc-review.googlesource.com/c/105660
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25285}
With this CL, the main and shadow filters are no longer fully reset to
0 as the delay changes. This allows for more robust echo removal for
some scenarios.
Bug: webrtc:9879,chromium:895838
Change-Id: I859aa3df3ae41648bc8efde01ec2e2a5cb392279
Reviewed-on: https://webrtc-review.googlesource.com/c/106345
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25251}
Removing code that has no audible effect.
Bug: webrtc:8671
Change-Id: Ibd7d0d19d760ae16b09285498c2ee09b42eb5968
Reviewed-on: https://webrtc-review.googlesource.com/c/106301
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25250}
This CL introduces two related changes
1) It changes the way that the AEC3 determines whether the linear
filter is sufficiently good for its output to be used. The new scheme
achieves this much earlier than what was done in the legacy scheme.
2) It changes the way that saturated echo is and handled so that the
impact of the nearend speech is lower.
Bug: webrtc:9835,webrtc:9843,chromium:895435,chromium:895431
Change-Id: I0b493676886e2134205e9992bbe4badac7e414cc
Reviewed-on: https://webrtc-review.googlesource.com/c/104380
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25208}
This CL simplifies the buffering of render data. Instead of making assumptions
about the worst possible platform, it leverages recent improvements in
the delay estimator to quickly adapt when the conditions change.
Pros:
- No capture delay, delay is found ~200 ms faster.
- Cleaner code that makes the concept of delay more clear.
- Allows for removal of one matched filter because of the jitter headroom
removal.
Cons:
- Delay estimator needs to re-adapt when the call jitter increases.
The code can be deactivated by a kill switch. When the kill switch is
pulled the CL is bit exact.
Bug: webrtc:9726,chromium:895338
Change-Id: Ie2f9c8c5ce5b5a4510b4bdb95db2b970b57cd5d0
Reviewed-on: https://webrtc-review.googlesource.com/c/96920
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25169}
In this CL the use of the stationarity properties at init is set to true by default.
Bug: webrtc:9865, chromium:894439
Change-Id: I716ce0d792a50616dc38cc0ba6f2c702549a81cc
Reviewed-on: https://webrtc-review.googlesource.com/c/105303
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25123}
"Perfection is achieved, not when there is nothing more to add,
but when there is nothing left to take away."
This CL removes the following kill-switches from AEC3
- WebRTC-Aec3DownSamplingFactor8KillSwitch
- WebRTC-Aec3NewSuppressionKillSwitch
- WebRTC-Aec3ShadowFilterJumpstartKillSwitch
- WebRTC-Aec3SlowFilterAdaptationKillSwitch
- WebRTC-Aec3SuppressorNearendAveragingKillSwitch
It also removes code paths and configuration parameters that are no
longer in use. The list of kill-switches in the audio processing
fuzzer test is updated.
The change has been tested for bit-exactness.
Bug: webrtc:8671
Change-Id: Ie0af86a14baf853548bf9c00b2b9b3bbc32c1aaa
Reviewed-on: https://webrtc-review.googlesource.com/c/105324
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25120}
This CL removes outdated code for testing of platforms with clock-drift
Bug: webrtc:8671
Change-Id: Ie202c514609d9f3d2357107b0daf895331275797
Reviewed-on: https://webrtc-review.googlesource.com/c/105183
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25105}
This CL ensures that the default setting for the usage of stationary signal
properties is not overridden by mistake.
Bug: chromium:894243
Change-Id: I85ab65383ee82b5f3153864da7a0cede7776c146
Reviewed-on: https://webrtc-review.googlesource.com/c/105181
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25104}
This CL changes the tuning of the echo suppressor for the case when
there is echo only. The resulting effect is a slight increase of
transparency
Bug: webrtc:9844,chromium:893744
Change-Id: I5e6a867e0d03dc3a468a8f5cfa64103e001baae1
Reviewed-on: https://webrtc-review.googlesource.com/c/104760
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25075}
This CL utilizes the AEC3 ability to tailor the suppressor during
situations when the nearend dominates over the residual echo. This is
done by increasing the thresholds for transparent echo suppressor
behavior when the nearend is strong compared to the residual echo.
Bug: webrtc:9836, chromium:893744
Change-Id: Ic06569eefc7f2557b401db43b3ac84b299071294
Reviewed-on: https://webrtc-review.googlesource.com/c/104460
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25071}
This is a new attempt to reduce the filter divergence
during double-talk without regressing in clock-drift
scenarios.
- The error_floor in decreased to allow for slow adaptation
when the filter performs well.
- The leakage_diverged is increased to allow for fast adaptation
when the shadow filter performs better.
- A new parameter, error_ceil, was added to stop the filter from
adapting too fast.
Bug: webrtc:9746,chromium:883264
Change-Id: Ie2868d2388b48412a192a004ec13f9eff34517b8
Reviewed-on: https://webrtc-review.googlesource.com/c/100460
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25063}
This CL lowers the default reverb decay to better match the standard
rooms where calls are made.
Bug: webrtc:9843
Change-Id: I46f1a629ecfdd72561829326d4fa58ede8107b6c
Reviewed-on: https://webrtc-review.googlesource.com/c/104740
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25061}
This CL introduces a major refactoring of AecState for the purpose of
simplifying further improvements to the logic in this code.
The changes have successfully been tested for bitexactness.
Bug: webrtc:8671
Change-Id: If98efde55a22c76b093089a11a0562daac7e16e6
Reviewed-on: https://webrtc-review.googlesource.com/c/102362
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24996}
In this CL the ERLE estimator is reset after a pre-amplifier gain change is communicated to APM.
Bug: webrtc:9805
Change-Id: I040f344e4607e862240250f9478d06de0d58a096
Reviewed-on: https://webrtc-review.googlesource.com/103222
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24933}
This CL removes some deprecated build targets (and their headers)
from system_wrappers:
- field_trial_api
- field_trial_default
- metrics_api
- metrics_default
It also refreshes all the dependencies on field_trial.h and metrics.h.
A nice side effect is that it is finally possible to remove 'nogncheck'
from the following files (when it was used with field_trial_default
and metrics_default):
- sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h
- sdk/android/src/jni/pc/peerconnectionfactory.cc
- sdk/objc/api/peerconnection/RTCFieldTrials.mm
Bug: webrtc:9631
Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/100524
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24878}
This CL removes killswitches for code that has been properly tested in
experiments and is to be considered to be permanent.
The changes have been tested for bitexactness.
Bug: webrtc:8671
Change-Id: I0f9db16f377390d9dd3779096da91f3abc0fb4a5
Reviewed-on: https://webrtc-review.googlesource.com/102360
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24877}
Specially for devices with high echo path gain, even low render signal can allow the linear filter of the AEC3 to converge. However, the conditions that were used for updating the ERLE avoided to update that estimation. In this commit, we allow adapting the ERLE estimator using even low render signal but the update of the ERLE is constraint in a way that decreases are not allowed.
Bug: webrtc:9776
Change-Id: Ic4331efcc47a0b05f394cdea9a88f336292de5a1
Reviewed-on: https://webrtc-review.googlesource.com/101641
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24859}
Speeds up adaptation of the matched filter of the delay estimator by
allowing the estimated echo and the error signal (microphone minus
estimated echo) to be saturated. Only microphone saturation pauses
the filter adaptation.
Bug: webrtc:9773
Change-Id: I8b8400539fde3ee821f36a95818bece02ddd626b
Reviewed-on: https://webrtc-review.googlesource.com/101341
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24802}
Right after a volume decrease, the echo path estimate is overestimated and, as a side effect, the nearend signal is also overestimated. Due to that, the suppression gains are kept high avoiding the suppression of echoes. In this CL the neared power spectrum estimation is limited to a level given by the power spectrum or the microphone input signal. Additionally, the minimum gain that is computed inside the suppressor is also modified. Instead of using the nearend power spectrum that is now bounded, the power spectrum of the signal after the linear echo canceler is used.
Bug: webrtc:9762
Change-Id: Ia24cd2ce248f2c2ba124711b75acff3b8c5cfa9f
Reviewed-on: https://webrtc-review.googlesource.com/100720
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24796}
Replace calls to .str() which copies with .Release which moves in cases where that's safe.
This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"
Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
This CL fixes a bug in the filter output transition when the 'from' input
points to the same array as the output. It also includes a slight
improvement to the transition by starting one sample earlier than
previously.
Bug: webrtc:9741,chromium:882789
Change-Id: Ifd5f16c1ac88a74d93499e7f4b4c0e5cb3e4976f
Reviewed-on: https://webrtc-review.googlesource.com/99540
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24683}
This CL adjusts the behavior of the AEC3 echo suppressor behavior
initially in the call, and when there has been delay changes. The
results is that short echo blips/bursts present in some such cases
no longer occur.
In particular this CL:
-Ensures that the suppressor back-off under stationary render
conditions does not occur until the linear filter has had the
ability to converge.
-Ensures that a previously converged filter behavior detection
is not sticky for stable and linear echo paths, which in turn
prevents echo leakage due to the more liberal echo suppressor
behavior applied on such platforms.
-Removes a bug that caused a random and jittery behavior for
the usage of the linear filter output initially in the calls
and after echo path changes
Bug: webrtc:9737, chromium:882396
Change-Id: Id2b46e366dc58ab8137f19ed59a2034c89ca3087
Reviewed-on: https://webrtc-review.googlesource.com/99063
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24656}
This CL introduces a new behavior for leveraging early information
about the delay that is acquired before the standard delay estimate
has been established.
To simplify the process of setting the parameters for that, the CL
also surfaces the delay estimator parameters to the config struct.
Bug: webrtc:9720,chromium: 880686
Change-Id: If886813f70cd805bd37752c63913d28398f1c6fe
Reviewed-on: https://webrtc-review.googlesource.com/97860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24614}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
This CL introduces the ability to control the usage of the shadow filter
output in the echo canceller output.
Bug: webrtc:9694,chromium:879451
Change-Id: I01f90de60de1799b32892051c176bda5e1a8d33e
Reviewed-on: https://webrtc-review.googlesource.com/97020
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24506}
During this work a parameter is added to the configuration file for the AEC3 that allows to enable or disable the use of a different ERLE estimation for the render onsets.
Bug: webrtc:9677
Change-Id: I467f2cd20683fee06b69c0ba51a90816c9e14f29
Reviewed-on: https://webrtc-review.googlesource.com/96082
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24470}
- Changes in the early reverberation estimation.
- Code optimization by avoiding squaring the whole impulse response.
Bug: webrtc:9651
Change-Id: Iefd4f5ad52a2584d21b20934db1fae5cb1bc81ed
Reviewed-on: https://webrtc-review.googlesource.com/95483
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24464}
Disables the faster filter adaptation in the event of
microphone gain changes as it sometimes impacted transparency
negatively.
Bug: webrtc:9526,chromium:863826
Change-Id: I48fb6dd45440518aaf94b6469d6bb891247ea4ab
Reviewed-on: https://webrtc-review.googlesource.com/95143
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24461}
Removing the some kill switches from the AEC3 codebase. CL is tested for
bit exactness.
Bug: webrtc:8671
Change-Id: I6ecdb1b5ccb05dca79bf0a0cd471f53d79d71d7e
Reviewed-on: https://webrtc-review.googlesource.com/96181
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24460}
This CL adds adaptive handling of platforms where the echo path has
a strong gain above 10 kHz. A configurable offset is adaptively applied
depending on the amount of echo and mode of the echo suppressor.
Bug: webrtc:9663
Change-Id: I27dde6dc23b04a76a3be8c49d7fc9e226b9137e6
Reviewed-on: https://webrtc-review.googlesource.com/95947
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24448}
Disable the use of ERLE uncertainty with a diverged filter as it has
been shown to make transparency worse.
Bug: webrtc:9668
Change-Id: I5e23665def187c0d1cf47a029c4ebc950e79bb44
Reviewed-on: https://webrtc-review.googlesource.com/96140
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24446}
This CL allows selecting an echo suppressor behavior which is specific
for whether the nearend is dominant, or the echo is dominant.
The changes in this CL are bitexact.
Bug: webrtc:9660
Change-Id: Ie32e65efe47e692de6d6a22a7ad3b469d745fd6b
Reviewed-on: https://webrtc-review.googlesource.com/95725
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24434}
This CL removes the UpdateGainIncrease code that is not used anymore.
The CL has been tested for bit exactness.
Bug: webrtc:8671
Change-Id: I4fcf26c3b4b5bba760ee139416ddefac86a36c2e
Reviewed-on: https://webrtc-review.googlesource.com/95940
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24425}
This CL removes some of the unused code in the suppressor. The CL has
been tested for bit exactness.
Bug: webrtc:8671
Change-Id: I960f9445dfd109cf1d5790debb8758872b5b8d0d
Reviewed-on: https://webrtc-review.googlesource.com/95682
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24417}
This CL removes the unused coherence computation from AEC3. This CL
only removes unused code, the output of AEC3 does not change.
Bug: webrtc:8671
Change-Id: Ie127c5ec64e29414f1e1570511d57a4d09fc9145
Reviewed-on: https://webrtc-review.googlesource.com/95650
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24403}
This CL adds functionality for applying an optional fixed delay in AEC3
to the capture signal
Bug: webrtc:9647
Change-Id: Id3b3f896bcf203e6611298dc804c3c80da9f1883
Reviewed-on: https://webrtc-review.googlesource.com/95142
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24399}
This CL:
-Adds another config parameter that controls the duration of the initial
state.
-Adds reading of that parameter in audioproc_f from the json settings file.
-Adds missing reading of another parameter in audioproc_f from the json
settings file.
Bug: webrtc:8671
Change-Id: Ie6164c360492de5e6b0ade8838bbabe214560b5e
Reviewed-on: https://webrtc-review.googlesource.com/94621
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24360}
This CL lowers the margins in the AEC3 conservative mode to increase
the transparency when there are audio buffer issues, and during call
startup.
In particular, this CL adjusts the parameters and thresholds to
-Make the requirements for filter divergence more strict, to minimize
the transparency loss during minor filter divergence.
-Decrease the echo power uncertainty used during initial filter
convergence, to increase transparency after audio buffer issues.
-Deactivate the enforcement of conservative suppressor gain after
audio buffer.
Bug: webrtc:9641,chromium:875611
Change-Id: Ie171bb411f17a1e8661c291118debd334f65c74f
Reviewed-on: https://webrtc-review.googlesource.com/94776
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24333}
This CL ensures that the linear echo prediction mode is not used
when the transparent mode is active.
TBR: saza@webrtc.org,gustaf@webrtc.org
Bug: webrtc:9612,chromium:873074
Change-Id: I25cda5226251df769b6524594ea8a2b78532aaec
Reviewed-on: https://webrtc-review.googlesource.com/93740
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24268}
This CL changes the AEC3 code to allow the main and shadow filters
to have different lengths.
Bug: webrtc:9614,chromium:873100
Change-Id: I3ec2861d496986610d5a73db5771bbe9b8bf7dcd
Reviewed-on: https://webrtc-review.googlesource.com/93465
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24265}
This CL makes the jump-starting of the shadow filter more extreme.
It furthermore utilizes this to allow the AEC to rely further, and
more quickly on its linear filter estimates.
The result is mainly increased transparency but also some
cases of fewer echo blips.
Bug: webrtc:9612,chromium:873074
Change-Id: I90f7cfbff9acb9d0c36409593afbf476e7a830d3
Reviewed-on: https://webrtc-review.googlesource.com/93461
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24264}
This CL ensures that the shadow filter is adapted at each block, which
avoids that a temporary filter length mismatch can occur between the
main and shadow filters.
Bug: webrtc:9602,chromium:872201
Change-Id: I651812b4e3b134c6c5e1fe3df5ab78dbdb5c1fb4
Reviewed-on: https://webrtc-review.googlesource.com/93000
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24253}
This CL adds functionality to use the shadow filter output instead
of the main filter output for cases when the former is better than
the latter. One case when that happens is when there have been an
echo path change, either in the acoustic path, in the audio buffers
or due to some active audio processing effects being applied on
the device.
The CL causes less echo leaks, in particular on devices with
active render processing.
Bug: webrtc:9581,chromium:869821
Change-Id: Icb8df1b94141598da82dc188051ac59e43338938
Reviewed-on: https://webrtc-review.googlesource.com/91820
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24166}
This CL introduces an adaptive estimation of the early reverb
in the estimation for the room reverberation. The benefits of
this is that for room with long early reflections there is
a lower risk of underestimating the reverberation.
This CL is for a landing the code in
https://webrtc-review.googlesource.com/c/src/+/87420,
and the review of the code was done in that CL. The author of
code is devicentepena@webrtc.org
Bug: webrtc:9479, chromium:865397
Change-Id: Id6f57e2a684664aef96e8c502e66775f37da59da
Reviewed-on: https://webrtc-review.googlesource.com/91162
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24146}
This CL increases the AEC3 transparency during call
startup and after echo path delay changes in 3 ways:
1. The exit requirements for the initial mode is
made less strict.
2. The requirements for using the linear echo model
are made less strict.
3. The duplicated reverb modelling in the linear mode
removed.
Bug: webrtc:9572,chromium:868329
Change-Id: I79ea0796ed26408e35576bb39eaae4e4848b4f83
Reviewed-on: https://webrtc-review.googlesource.com/90868
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24132}
This CL adds a functionality that jump-starts the
AEC3 shadow filter whenever it performs consistently
worse than the main filter.
The jump-start is done such that the shadow filter
is re-initialized using the main filter coefficients.
The effects of this is a significantly more accurate
main linear filter which leads to less echo leakage
and better transparency
Bug: webrtc:9565, chromium:867873
Change-Id: Ie0b23cd536adc7ce96fc3ed2a7db112aec7437f1
Reviewed-on: https://webrtc-review.googlesource.com/90413
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24117}
This CL corrects the way that the echo subtractor output is
adjusted during the adjustment of the adaptive filter when the
analog AGC gain changes.
The CL also ensures that the main adaptive filter is not updated
when this occurs.
Bug: webrtc:9561,chromium:867373
Change-Id: I636f936128f7d9f0d82ca4140b59f148eb35d6a4
Reviewed-on: https://webrtc-review.googlesource.com/90401
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24101}
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 CL refactors the code in AEC3 that analyzes how
well the adaptive filter performs. The purpose of this
is both to simplify code that is more complex than needed
and also to pave the wave for the upcoming CLs that
softens the echo suppression during doubletalk.
The main changes are that:
-The shadow adaptive filter is now never analyzed. This
turned out to never affect the output in the recordings
it was tested on.
-The convergence analysis was moved to the aec state
code.
The changes are bitexact on all testcases where they
have been tested on.
Bug: webrtc:8671
Change-Id: If76b669565325c8eb4d11d1178a7e20306da9a26
Reviewed-on: https://webrtc-review.googlesource.com/87430
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23958}
This reverts commit e90879097c.
Reason for revert: breaking downstream projects
Original change's description:
> IWYU: Add <cmath> for fabsf() and powf(), switch to C++ versions
>
> Fix the build with libstdc++ after 496cedfe5 ("AEC3: Reverberation model:
> Changes on the decay estimation") by including the missing header:
>
> ../../modules/audio_processing/aec3/reverb_model_estimator.cc: In constructor ‘webrtc::ReverbModelEstimator::ReverbModelEstimator(const webrtc::EchoCanceller3Config&)’:
> ../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: error: ‘fabsf’ was not declared in this scope
> reverb_decay_(fabsf(config.ep_strength.default_len)),
> ^~~~~
> ../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: note: suggested alternative: ‘labs’
> reverb_decay_(fabsf(config.ep_strength.default_len)),
> ^~~~~
> labs
> ../../modules/audio_processing/aec3/reverb_model_estimator.cc: In member function ‘void webrtc::ReverbModelEstimator::UpdateReverbDecay(const std::vector<float>&)’:
> ../../modules/audio_processing/aec3/reverb_model_estimator.cc:206:15: error: ‘powf’ was not declared in this scope
> decay = powf(2.0f, -exp_candidate * kFftLengthBy2);
> ^~~~
>
> While here, also switch to the C++ versions of those functions: std::fabs()
> and std::pow() respectively.
>
> Spotted by Jose Dapena Paz <jose.dapena@lge.com>.
>
> Bug: chromium:819294
> Change-Id: Id803243be8dd17eac95c70a88a37ee2fe1505a5a
> Reviewed-on: https://webrtc-review.googlesource.com/87421
> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23870}
TBR=gustaf@webrtc.org,alessiob@webrtc.org,raphael.kubo.da.costa@intel.com,devicentepena@webrtc.org
Change-Id: I22423a2d4201183f70ae084e0e21930367824f1c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:819294
Reviewed-on: https://webrtc-review.googlesource.com/87401
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23871}
Fix the build with libstdc++ after 496cedfe5 ("AEC3: Reverberation model:
Changes on the decay estimation") by including the missing header:
../../modules/audio_processing/aec3/reverb_model_estimator.cc: In constructor ‘webrtc::ReverbModelEstimator::ReverbModelEstimator(const webrtc::EchoCanceller3Config&)’:
../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: error: ‘fabsf’ was not declared in this scope
reverb_decay_(fabsf(config.ep_strength.default_len)),
^~~~~
../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: note: suggested alternative: ‘labs’
reverb_decay_(fabsf(config.ep_strength.default_len)),
^~~~~
labs
../../modules/audio_processing/aec3/reverb_model_estimator.cc: In member function ‘void webrtc::ReverbModelEstimator::UpdateReverbDecay(const std::vector<float>&)’:
../../modules/audio_processing/aec3/reverb_model_estimator.cc:206:15: error: ‘powf’ was not declared in this scope
decay = powf(2.0f, -exp_candidate * kFftLengthBy2);
^~~~
While here, also switch to the C++ versions of those functions: std::fabs()
and std::pow() respectively.
Spotted by Jose Dapena Paz <jose.dapena@lge.com>.
Bug: chromium:819294
Change-Id: Id803243be8dd17eac95c70a88a37ee2fe1505a5a
Reviewed-on: https://webrtc-review.googlesource.com/87421
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23870}
Fix the build with libstdc++ after 496cedfe5 ("AEC3: Reverberation model:
Changes on the decay estimation") by including the missing header:
../../modules/audio_processing/aec3/reverb_model_estimator.cc: In constructor ‘webrtc::ReverbModelEstimator::ReverbModelEstimator(const webrtc::EchoCanceller3Config&)’:
../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: error: ‘fabsf’ was not declared in this scope
reverb_decay_(fabsf(config.ep_strength.default_len)),
^~~~~
../../modules/audio_processing/aec3/reverb_model_estimator.cc:61:21: note: suggested alternative: ‘labs’
reverb_decay_(fabsf(config.ep_strength.default_len)),
^~~~~
labs
../../modules/audio_processing/aec3/reverb_model_estimator.cc: In member function ‘void webrtc::ReverbModelEstimator::UpdateReverbDecay(const std::vector<float>&)’:
../../modules/audio_processing/aec3/reverb_model_estimator.cc:206:15: error: ‘powf’ was not declared in this scope
decay = powf(2.0f, -exp_candidate * kFftLengthBy2);
^~~~
Spotted by Jose Dapena Paz <jose.dapena@lge.com>.
Bug: chromium:819294
Change-Id: If992e5e473b9d4d0c1b3c1006c3816b7c4eee296
Reviewed-on: https://webrtc-review.googlesource.com/87241
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#23863}
This clarifies the dependencies of utility/ a lot (spoiler:
there are very few) and makes it easier to separate the build
targets for aecm and aec2.
Bug: webrtc:9488
Change-Id: If916f86e80c19d1b650d0908fbe8343ea7c47bd7
Reviewed-on: https://webrtc-review.googlesource.com/87141
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23860}
The main filter is adapted at a lower rate which reduces the risk of
diverging during double talk. The change yields notable transparency
improvements.
Bug: webrtc:9497
Change-Id: Ib23b7a4055d313dede535d2b65dc7e023a2db042
Reviewed-on: https://webrtc-review.googlesource.com/87300
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23858}
In this CL we have introduced changes on the estimation of the decay involved in the exponential modeling of the reverberation. Specifically, the instantaneous ERLE has been tracked and used for adapting faster in the regions when the linear filter is performing well. Furthermore, the adaptation is just perform during render activity.
Change-Id: I974fd60e4e1a40a879660efaa24457ed940f77b4
Bug: webrtc:9479
Reviewed-on: https://webrtc-review.googlesource.com/86680
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23836}
This change simplifies the calculation of the suppression gains.
It also contains a new tuning of the suppressor.
The suppressor behavior is tuned by setting echo-to-nearend ratios
for when the suppressor is to be fully transparent and for when to
fully suppress. An echo-to-masker value determines when the signal
is masked by noise. These three values are specified for low and
high frequencies.
Change-Id: I108e83c8f2a35462085a3fabaebcc02fa3103607
Bug: webrtc:9482
Reviewed-on: https://webrtc-review.googlesource.com/86021
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23830}
In this work the performance of the linear filter is
estimated. The estimation aims at capture situations when the linear
filter is largely over-estimating the echo. In those circumstances,
the linear filter is scaled with the purpose of accelerating its
convergence.
Change-Id: I05ea3739d82838a6f08673432da92125c47943e0
Bug: webrtc:9466,chromium:857426
Reviewed-on: https://webrtc-review.googlesource.com/86133
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23789}
This CL removes the constraint that freezes the filter adaptation
whenever the estimated echo or the prediction error is saturated. This
allows for much more rapid filter recovery in cases where the echo path
gain for some reason changes, such as when the analog AGC gain is
adjusted or the loudspeaker volume is changed.
TBR: devicentepena@webrtc.org
Bug: webrtc:9466,chromium:857426
Change-Id: Ic0b3b03f41f12e9a607aaadd2ee91cbaa16cac52
Reviewed-on: https://webrtc-review.googlesource.com/86124
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23775}
This CL changes the behavior when the main filter diverges.
Instead of entering non-linear mode, the AEC continues to operate in
linear mode but estimates the residual echo differently. R2 is S2
scaled by a factor of 10.
Bug: chromium:857018,webrtc:9462
Change-Id: I41212efe164ad319cf38a163cdf9d3ea151e0997
Reviewed-on: https://webrtc-review.googlesource.com/85981
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23772}
The frequency shape of the echo path has been included in the reverberation model.
Bug: webrtc:9454,chromium:856636
Change-Id: Id2bc3096df31e29328936f94fe965ed1883d70f7
Reviewed-on: https://webrtc-review.googlesource.com/85370
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23746}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
Reduce noise of the nearend spectrum estimation by averaging multiple
frames.
Bug: webrtc:9420,chromium:853699
Change-Id: Iad7e68b1209a369e263b2d892791943e42bfbb3f
Reviewed-on: https://webrtc-review.googlesource.com/83960
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23655}
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}
In this work we introduce some changes on how the reverberation model for AEC3 is applied. Currently, the exponential modelling of the tails is applied over the linear echo estimates. That might result in an overestimation of the reverberation tails under certain conditions. In this work, the reverberation model is instead applied over an estimate of the energies at the tails of the linear estimate.
Additionally, the stationary estimator is changed so it does not disable the aec immediately after a burst of activity.
Bug: webrtc:9384,webrtc:9400,chromium:852257
Change-Id: Ia486694ed326cfe231fc688877c0b9b6e2c450ff
Reviewed-on: https://webrtc-review.googlesource.com/82161
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23599}
This CL ensures that the linear-filter based refined delay is chosen to
match the delay that was detected by the delay estimator during the time
it takes for the linear filter to converge.
Bug: webrtc:9371,chromium:850451
Change-Id: Ib9cf532df0577ceca10a260d9d2deba5306f88bb
Reviewed-on: https://webrtc-review.googlesource.com/81682
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23537}
This filter preprocess the time domain representation of the adaptive
linear filter to avoid low-frequency components causing issues in
the filter analysis.
Bug: webrtc:9343, chromium:848231
Change-Id: I40494959f1b76242a7c9f2a2fc85c2ad4af9e164
Reviewed-on: https://webrtc-review.googlesource.com/79142
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23536}
This change contains a new anti-aliasing filter for the delay estimator
for down-sampling factor 4. The new (elliptic) filter has a much wider
main lobe allowing for faster convergence.
Bug: webrtc:9288,chromium:846615
Change-Id: Id109974a59fe6f48c5e0ccc4f4e06c0d94c8bd03
Reviewed-on: https://webrtc-review.googlesource.com/81680
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23534}
Preparing for changing the filters of the decimator by moving the old
filters to the new zero, pole, gain notation.
Bug: webrtc:9288,chromium:846615
Change-Id: I2b01a2555d34617e0bf251c782703753f72cd56f
Reviewed-on: https://webrtc-review.googlesource.com/81189
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23528}
The signal used for delay estimation at downsampling factor 8 is bandpass
filtered and contains less energy than for other downsampling factors.
This CL adjusts the energy threshold used for determining if there is enough
farend activity to update the matched filters in the delay estimator.
Only downsampling factor 8 is affected.
Bug: webrtc:9288,chromium:846615
Change-Id: I6f38f5609a31e7a08e60571ac75ea75c9962e026
Reviewed-on: https://webrtc-review.googlesource.com/80443
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23486}
CascadedBiQuadFilter can run identical filters multiple times. This CL
allows the use of different filters in each step. This enables the use
of more elaborate filters. The filters are defined by zeros, poles and
gains.
The 'old' way of initializing CascadedBiQuadFilter with a transfer
function and number of filters is left intact.
Bug: webrtc:9288,chromium:846615
Change-Id: Ie4a5b98eba044415571cdcac087b20870a0b5d33
Reviewed-on: https://webrtc-review.googlesource.com/80060
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23473}
In this work, we change the behavior of the gain limiter so it also looks at the energy
on farend around the default delay for deciding the suppression gain
that should be applied at the initial portion of the call.
Bug: webrtc:9311,chromium:846724
Change-Id: I0b777cedbbd7fd689e72070f72237296ce120d3c
Reviewed-on: https://webrtc-review.googlesource.com/78960
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23400}
Letting the delay estimator operate at a sampling frequency of 2 kHz
with audio between 0 and 1 kHz makes it sensitive to noisy environments.
This CL bandpass filters the 16 kHz signal before downsampling to 2 kHz
in a way that the downsampled 2 kHz signal contains audio between 1 and
2 kHz. It also sets downsampling factor 8 as default which significantly
reduces computational complexity.
Bug: webrtc:9288,chromium:846615
Change-Id: Iaf67898a1a14326cd61bb7f81c14d3c12a697c8d
Reviewed-on: https://webrtc-review.googlesource.com/78703
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23395}
This CL fixes the rounding of the estimated average call skew. Before it
was rounded down (toward INT_MIN). Now it is rounded to the nearest integer.
This avoids unnecessary fluctuations of the estimated call skew (and
unnecessary resets).
Bug: webrtc:9283,chromium:888042
Change-Id: Id5b3c593f812f5f9fd3dcdafb7e388a6ef1ac153
Reviewed-on: https://webrtc-review.googlesource.com/77684
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23338}
The ERLE computation was improved by two means:
- The update function was always called and just parts of the internal code reacts to the converged filter flag
- When computing the ERLE, the ratio of energies is now computed using more points and, therefore, a more robust estimation is achieved.
Bug: webrtc:9284
Change-Id: Ie4f871f19cfad1a13741352ddd7b0a27ad6c3fb6
Reviewed-on: https://webrtc-review.googlesource.com/77767
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23329}
This CL applies a high pass filter to the delay estimator signals which
improves the adaptation of the matched filters in noisy environments.
This results in faster delay estimation.
Bug: webrtc:9288
Change-Id: I8ffe5442eab7ac2f10a7ba236b08a0f07ec90645
Reviewed-on: https://webrtc-review.googlesource.com/77725
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23308}
This CL increases the allowed variations in the API call skew limit in
AEC3.
Bug: webrtc:9283,chromium:888042
Change-Id: Ib5e784c6f3dcf1bf3a2cbfe2b1559953db9227a8
Reviewed-on: https://webrtc-review.googlesource.com/77430
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23305}
This CL turns on the previously implemented AEC3 audibility
improvements, which before has been off by default.
Bug: webrtc:9193,chromium:836790
Change-Id: Ibcd057ba5dd002718d62fd83db33d01d9563b8ea
Reviewed-on: https://webrtc-review.googlesource.com/77123
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23265}
This CL adds robustness to avoid the AEC3 transparent mode to be
incorrectly activated when
-there is strong near-end noise
-there is only low-level nearend activity.
Bug: webrtc:9256,chromium:841193
Change-Id: I26c2759d163914eb85dc3d863da8acbf28cbb88d
Reviewed-on: https://webrtc-review.googlesource.com/75511
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23191}
This CL corrects the usage of the estimated echo path gain to not be
hardcoded to 1. In order to retain the tuned behavior, the CL for now
maintains the former behavior in the code.
Bug: webrtc:9255,chromium:851187
Change-Id: I7f91c72e476680a8a854c22b74b1771fae446110
Reviewed-on: https://webrtc-review.googlesource.com/75510
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23190}
This CL ensures that the external audio buffer delay is correctly used
by removing the applied headroom and avoiding that the delay estimation
feedback fromt the echo remover overrules the external delay
information.
Bug: webrtc:9241,chromium:839860
Change-Id: I53cc78ace34a71994ab24a3b552f29979e2aae78
Reviewed-on: https://webrtc-review.googlesource.com/75513
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23189}
This CL softens the effect of the AEC3 transparent mode to also handle
headsets that leak low-level echoes in a nonlinear way.
This is handled by reintroducing the limit in the echo path gain for the
nonlinear mode. Due to recent improvements in echo suppressor behavior
this is now possible to do with a limited impact on the near-end speech.
Bug: webrtc:9246,chromium:840347
Change-Id: I0ca5157160d1884ba93b962323b56016756986d3
Reviewed-on: https://webrtc-review.googlesource.com/74703
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23145}
Fixes a confusion of time units (milliseconds vs blocks) of externally
reported audio delay. This fix reduces the risk of echo in the beginning
of a call.
Bug: webrtc:9241,chromium:839860
Change-Id: I534cc15d6b215a5881ae46759f573a56871170a3
Reviewed-on: https://webrtc-review.googlesource.com/74589
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23128}
This CL contains changes to the echo suppressor that improves the
transparency of AEC3.
- The comfort noise level is used as masker and the masking threshold is
increased.
- Suppression gains are allowed to increase more rapidly.
- Suppression gains decrease slower in the lower frequencies after strong
nearend.
Change-Id: I7adf31ed90b0e007072191f40439f27c3b0bccf2
Bug: webrtc:9230,chromium:839379
Reviewed-on: https://webrtc-review.googlesource.com/73680
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23115}
The audio processing code is using parts of the iSAC codec to do voice
activity detection (VAD), but it's undesirable for it to pull in the
entire iSAC codec as a dependency. So this CL factors out the parts of
iSAC that's needed for VAD to a separate build target.
Bug: webrtc:8396
Change-Id: I884e25d8fd0bc815fca664352b0573b4b173880e
Reviewed-on: https://webrtc-review.googlesource.com/69640
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23110}
This CL removes the updating of the buffered data used to to pad the
64 sample blocks to 128 samples FFTs. As that padding was used
incorrectly in one place this resolves an important issue.
Bug: webrtc:9159,chromium:833801,webrtc:9206
Change-Id: Ie6830878ebec6130b61d4e7e3169357f2e253073
Reviewed-on: https://webrtc-review.googlesource.com/73240
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23059}
This CL changes the way the suppressor gain is computed in AEC3 in that
the FFTs used are padded with data and windowed with a Hanning-style
window.
This gives better FFT accuracy, an behavior matching the suppressor
gain application, and also results in one less FFT operation.
Bug: webrtc:9204,chromium:837563
Change-Id: I612676c389cb76a3130966a9b596ff3f44d21863
Reviewed-on: https://webrtc-review.googlesource.com/73141
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23057}
This reverts commit b04e5cae08.
Reason for revert: The reason for the revert is that some scenarios were detected where this caused the delay estimation to occur too slowly.
Original change's description:
> Making the delay estimator more robust to noisy nearends and low echoes
>
> This CL reduces the delay estimator step size to make it react better in
> scenarios where the environment is noisy, or the echo level is fairly
> low.
>
> Bug: webrtc:9177,chromium:835281
> Change-Id: I482d898c91eddc497e1284ee500d26df21a0574a
> Reviewed-on: https://webrtc-review.googlesource.com/71486
> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22990}
TBR=gustaf@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9177, chromium:835281
Change-Id: I33e09ebfed8ad8330419e554f482c956608befce
Reviewed-on: https://webrtc-review.googlesource.com/72843
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23042}
This CL makes sure that the coherence-based gains are affected by the
upper gain limit during call start-up and after resets.
Bug: webrtc:9159,chromium:833801
Change-Id: I93fdd173b6e11ea861d0e01e12c048ec0a91db70
Reviewed-on: https://webrtc-review.googlesource.com/72841
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23039}
This CL is created from a work initiated at https://webrtc-review.googlesource.com/c/src/+/61160
The purpose of this work is to improve the performance of the echo canceler (AEC3) when the farend signal contains stationary noises:
- An stationarity estimator of the farend signal has been added for detecting the portions of the farend signal that are pure noise.
- When the echo canceler deals with a portion of the signal that contains basically noise, the echo suppressor is able to back-off and avoid the fading of the nearend speech.
Change-Id: Id4b87fc59f4765bf1fca36d1cab39a49aabe104a
Bug: webrtc:9193,chromium:836790
Reviewed-on: https://webrtc-review.googlesource.com/64141
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23024}
The code that attenuates narrow banded echo peaks in low frequencies
is removed as it affects transparency negatively.
Bug: webrtc:9192,chromium:836729
Change-Id: Ib90ce6a3db0a75e8d69bdca432e1f8f8bfbbd988
Reviewed-on: https://webrtc-review.googlesource.com/72380
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23022}
This CL overrides the power-based suppressor gain decision with
a coherence based descision for the cases when that indicates a
higher suppressor gain.
Bug: webrtc:9159,chromium:833801
Change-Id: I0e7d82ac1b8c70ffe9d45907559bb14b1b849d71
Reviewed-on: https://webrtc-review.googlesource.com/71660
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22997}
This CL reduces the delay estimator step size to make it react better in
scenarios where the environment is noisy, or the echo level is fairly
low.
Bug: webrtc:9177,chromium:835281
Change-Id: I482d898c91eddc497e1284ee500d26df21a0574a
Reviewed-on: https://webrtc-review.googlesource.com/71486
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22990}
Only specially taggged targets may transitively depend on poisonous
targets. We first apply it to audio codecs.
This makes it much clearer exactly what parts of the code still have
dependencies on the audio codecs (and we want to eventually get rid of
pretty much all of them).
Bug: webrtc:8396, webrtc:9121
Change-Id: Iba5c2e806c702b5cfe881022674705f647896d43
Reviewed-on: https://webrtc-review.googlesource.com/69520
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22979}
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 increases the transparency in AEC3 during regions of low level
echo. What is done is:
-Low-level echoes are smoothly weighted so as to be deemed less
disturbing.
-The time-domain masking effect of the nearend speech is increased for
all frequencies.
-A separate, even more increased, time-domain masking effect is
introduced for lower frequencies.
-The intra-band masking is reduced to reduce the risk of echo leakage.
-The limiting of maximum gain due to filter-bank dynamics is removed
as the usecase for it could no longer be identified.
Bug: webrtc:9159,cromium:833801
Change-Id: I289b92919763124d6c5e5ede19e9a5917877c654
Reviewed-on: https://webrtc-review.googlesource.com/70421
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22915}
This CL changes the handling of saturated microphone signals in AEC3.
Some of the changes included are
-Make the detection of saturated echoes depend on the echo path gain
estimate.
-Remove redundant code related to echo saturation.
-Correct the computation of residual echoes when the echo is saturated.
-Soften the echo removal during echo saturation.
Bug: webrtc:9119
Change-Id: I5cb11cd449de552ab670beeb24ed8112f8beb734
Reviewed-on: https://webrtc-review.googlesource.com/67220
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22809}
instead of relying on optional.h to included these 2 headers.
Bug: webrtc:9078
Change-Id: I7a4b3facd81690b8f107640487e129986c1f5ff6
Reviewed-on: https://webrtc-review.googlesource.com/68602
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22803}
This CL adds a timeout for the detection of the headset mode that
allows it to be entered also for the cases where a headset is
inserted during the call.
Bug: chromium:826720,webrtc:9083
Change-Id: Ic3cb4cc0258997a74eccd1bcdf65765e44016ad8
Reviewed-on: https://webrtc-review.googlesource.com/65240
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22658}
