Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
The type rtc::scoped_refptr<T> is now part of api/. Please include it from
api/scoped_refptr.h.
More info: See: https://groups.google.com/forum/#!topic/discuss-webrtc/Mme2MSz4z4o.
Bug: webrtc:9887, webrtc:8205
No-Try: True
Change-Id: Ic6c7c81e226e59f12f7933e472f573ae097b55bf
Reviewed-on: https://webrtc-review.googlesource.com/c/119041
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26414}
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 is a reland of 80b95de765
Original change's description:
> Trim down FileWrapper class to be merely a wrapper owning a FILE*
>
> Bug: webrtc:6463
> Change-Id: If71e2f3a75dc1863bc805ab71de1e2d33294f805
> Reviewed-on: https://webrtc-review.googlesource.com/c/117881
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26311}
Bug: webrtc:6463
Change-Id: I12154ef65744c1b7811974a1d871e05ed3fbbc27
Reviewed-on: https://webrtc-review.googlesource.com/c/118660
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26337}
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}
The pitch gain type in ComputePitchGainThreshold() is wrong
(size_t instead of float).
The pitch period is an unsigned integer type, but it is safer to
switch to a signed type and add checks on the sign.
Bug: webrtc:9076
Change-Id: If69d182071edab9750a320f0fbfac24aa8052ee0
Reviewed-on: https://webrtc-review.googlesource.com/c/117302
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26259}
This Config configuration will eventually replace the AudioProcessing::noise_suppression() interface.
This also introduces a proxy NoiseSuppression, returned by AudioProcessing::noise_suppression.
Without this proxy, ApplyConfig could overwrite NS settings for clients who currently use noise_suppression(). For example, the following code will not preserve the noise suppression level:
apm->noise_suppression()->set_level(NoiseSuppression::kHigh);
auto cfg = apm->GetConfig();
apm->ApplyConfig(cfg);
The NoiseSuppression instance returned by noise_suppression() has no way to update the config inside APM, so GetConfig() will return an out-of-date config which is then re-applied. This CL adds a proxy that makes this update, by forwarding Enable() and set_level() calls to ApplyConfig().
Drive-by change: AudioProcessing::Config substructs are reordered to mirror the capture processing pipeline.
Tested: Ran ToT and this CL builds of audioproc_f and verified identical settings/aecdumps.
Bug: webrtc:9947
Change-Id: I823eade894be115c254d656562564108b2b63b1f
Reviewed-on: https://webrtc-review.googlesource.com/c/116521
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26248}
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}
This CL adds two unit tests to make sure that, when an echo path gain
change occurs, the echo canceller is notified.
Such a change can be caused by (i) a pre-amplifier gain change or
(ii) an analog gain change.
Bug: webrtc:7494
Change-Id: Ia47cfbbc5694340cd3e760d8d3c3393f79897a9d
Reviewed-on: https://webrtc-review.googlesource.com/c/111780
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26190}
This updates some tests to use AudioProcesing::Config() and
AudioProcessing::GetStatistics() instead.
Some tests are left with voice_detection() because
a) not all tests make sense to run both APIs in parallel, and
b) we want test coverage of the old VoiceDetection until it is removed.
Bug: webrtc:9947
Change-Id: Ifb21a1e6e931d7ad3c3a4e38f5cc4f146da3c9a3
Reviewed-on: https://webrtc-review.googlesource.com/c/116160
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26134}
This adds a second (!) VoiceDetection instance in APM, activated via webrtc::AudioProcessing::Config and which reports its values in the webrtc::AudioProcessingStats struct.
The alternative is to reuse the existing instance, but that would require adding a proxy interface returned by AudioProcessing::voice_detection() to update the internal config of AudioProcessingImpl when calling voice_detection()->Enable().
Complexity-wise, no reasonable client will enable both interfaces simultaneously, so the footprint is negligible.
Bug: webrtc:9947
Change-Id: I7d8e28b9bf06abab8f9c6822424bdb9d803b987d
Reviewed-on: https://webrtc-review.googlesource.com/c/115243
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26101}
This CL guards against null pointer dereference, as caught by
clang static analyzer [1].
It also removes a useless field initialization, which happened
to trigger a false positive from said analyser.
[1] https://chromium.googlesource.com/chromium/src/+/HEAD/docs/clang_static_analyzer.md
Bug: webrtc:8793
Bug: webrtc:9855
Change-Id: Ia0fee24395eb2df16b526bbdffa5da6275b0909a
Reviewed-on: https://webrtc-review.googlesource.com/c/115044
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#26091}
This replaces the current usage of AudioProcessing::level_estimator()
in that test.
The unit tests that specifically test the level_estimator API are left
in place, until the level_estimator API itself is removed.
Bug: webrtc:9947
Change-Id: I73301c1478d2c9763bb49598a692142229102876
Reviewed-on: https://webrtc-review.googlesource.com/c/114550
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26049}
Changing VAD (voice activity detector) confidence threshold from 40%
to 90%. The proportion of samples classified as speech drops to ca 80%
of what it was when the threshold was 40%. Therefore,
kFullBufferSizeMs has to be increased by 1.0/0.8. We increase it from
1600ms to 2000ms.
TESTED = Did run the new and old configs on AEC dumps. With one minute
of kitchen noise, the new tuning boosted the noise by 3-4 db less.
Bug: chromium:913430
Change-Id: I4a2ebb6d1d309c6c20dd23c3685818b1b5ad4a66
Reviewed-on: https://webrtc-review.googlesource.com/c/113806
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25950}
It's currently used only by the VCMJitterBuffer and VCMReceiver
classes. Injection is needed by the VCMReceiverTimingTest test, which
defines a subclass(!) of EventWrapper.
Bug: webrtc:3380
Change-Id: I765be0ceac58e941928319cc426ba49f1cbdc5fa
Reviewed-on: https://webrtc-review.googlesource.com/c/113002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25893}
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 adds an interface for accessing stats on the capture stream, and
adds a level estimator to report one of the stats.
Bug: webrtc:9947
Change-Id: Id472534fa2e04d46c9ab700671f620584a246afb
Reviewed-on: https://webrtc-review.googlesource.com/c/109587
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25786}
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.
It also removes some unused dependencies in the WebRTC build graph.
Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
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}
Avoid that the client code relies on the adaptive digital mode being
enabled by default (error prone).
Bug: webrtc:7494
Change-Id: I765fecf535cf31a2163e10595a42520473c233b6
Reviewed-on: https://webrtc-review.googlesource.com/c/111586
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25728}
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}
This CL makes possible to choose the level estimation for the adaptive
digital GC of AGC2. The options are RMS (default and currently used
estimator) and peak-based (already computed, but not used).
Besides adding the new AGC2 config param for the level estimator, this CL
also refactors the config class by making it more structured.
Bug: webrtc:7494
Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc
Reviewed-on: https://webrtc-review.googlesource.com/c/110144
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25620}
Gain specified by fuzzer in APM config was too high.
Bug: chromium:901661
Change-Id: Id3ea8d23a4284a35c827bb16125902d84e37ca1e
Reviewed-on: https://webrtc-review.googlesource.com/c/110604
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25594}
This reverts commit 61c6e5643e.
Reason for revert: downstream projects prepared for this change
Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
>
> This reverts commit a7f77a7c05.
>
> Reason for revert: breaking downstream
>
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> >
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> >
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> >
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
>
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
>
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
This reverts commit a7f77a7c05.
Reason for revert: breaking downstream
Original change's description:
> Isolating APM API build target: making :api an actual target.
>
> This CL is part of a refactoring work to unblock other CLs
> that would generate a circular dependency when including
> modules/audio_processing. It will also allow to easily move
> the APM interface part under //api.
>
> More in detail, this change moves the APM interface files from
> the build target modules/audio_processing to
> modules/audio_processing:api. It also adds :api as dependency
> where needed.
>
> Bug: webrtc:9535
> Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25539}
TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109820
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25540}
This CL is part of a refactoring work to unblock other CLs
that would generate a circular dependency when including
modules/audio_processing. It will also allow to easily move
the APM interface part under //api.
More in detail, this change moves the APM interface files from
the build target modules/audio_processing to
modules/audio_processing:api. It also adds :api as dependency
where needed.
Bug: webrtc:9535
Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
Reviewed-on: https://webrtc-review.googlesource.com/c/109501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25539}
