If the previous value of the histogram is unknown, no scaling should be performed. Without this check a crash would occur. This issue was introduced in https://webrtc-review.googlesource.com/c/src/+/8101, and can only be triggered if the corresponding field trial is set.
Bug: webrtc:8381
Change-Id: I6e7cd8e14f6f4cc972fc094f010ecdf5091b2017
Reviewed-on: https://webrtc-review.googlesource.com/12380
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20336}
We added a new bot to client.webrtc.fyi (https://build.chromium.org/p/client.webrtc.fyi/builders/Win%20%28more%20configs%29).
It seems it is spotting some unsafe conversions and this CL is a test to see if we can use rtc::dchecked_cast to fix them:
../../modules/audio_coding/neteq/neteq_unittest.cc(547): error C2220: warning treated as error - no 'object' file generated
../../modules/audio_coding/neteq/neteq_unittest.cc(547): warning C4267: '=': conversion from 'size_t' to 'uint16_t', possible loss of data
../../modules/audio_coding/neteq/neteq_unittest.cc(548): warning C4267: '=': conversion from 'size_t' to 'uint32_t', possible loss of data
../../modules/audio_coding/neteq/neteq_unittest.cc(977): warning C4267: '+=': conversion from 'size_t' to 'uint32_t', possible loss of data
../../modules/audio_coding/neteq/neteq_unittest.cc(979): warning C4267: '+=': conversion from 'size_t' to 'uint32_t', possible loss
Bug: chromium:759980
Change-Id: Icd0f32ccf620c7c6642fadff797dc2482918648d
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/12921
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20335}
This reverts commit 54d1da13a584680ae80a1f229291e5bb7e76e6e1.
Reason for revert: Breaking tests
Original change's description:
> BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic.
>
> This CL implements the main logic and IOS appRTC integration.
>
> Unit tests and Android appRTC will be in separate CL.
>
> Bug: webrtc:8243
> Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
> Reviewed-on: https://webrtc-review.googlesource.com/4860
> Commit-Queue: Alex Narest <alexnarest@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20329}
TBR=deadbeef@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,alexnarest@webrtc.org
Change-Id: I5be1da78f360f72be66f9d56dd6b88c1cc13e963
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8243
Reviewed-on: https://webrtc-review.googlesource.com/12560
Reviewed-by: Lu Liu <lliuu@webrtc.org>
Commit-Queue: Lu Liu <lliuu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20330}
This CL implements the main logic and IOS appRTC integration.
Unit tests and Android appRTC will be in separate CL.
Bug: webrtc:8243
Change-Id: If8e5195294046a47316e9fade1b0dfec211155e1
Reviewed-on: https://webrtc-review.googlesource.com/4860
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20329}
Extract and save some simple annotations for the clean speech input.
The annotations are estimated level, VAD (assuming clean speech) and speech level.
TBR=
Bug: webrtc:7494
Change-Id: Id73358e228fac721a77fc8a61a3474a5d52bdc84
Reviewed-on: https://webrtc-review.googlesource.com/12321
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20327}
We are using <math.h>, not <cmath>. While the latter defines additional
overloads for abs(), including abs(float), they are not guaranteed to be
available in <math.h>.
libc++ ships its own math.h with the additional overloads, and libstdc++ (v6
or later) has a math.h that includes <cmath>, but this is not always
expected to work: for example, GCC 5.x's libstdc++ does not have these
additional overloads and causes the build to fail.
Just use fabsf() from the C standard library directly, as it achieves the
same thing in a more portable fashion.
Bug: None
Change-Id: I805728269b35051edb54126e204eccd2706e3a92
Reviewed-on: https://webrtc-review.googlesource.com/11460
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Raphael Kubo da Costa (rakuco) <raphael.kubo.da.costa@intel.com>
Cr-Commit-Position: refs/heads/master@{#20325}
PacedSender::probing_send_failure_ and PacedSender::packet_counter_ should probably also be protected by the critical section.
(This isn't the cause of webrtc:8331.)
TBR=stefan@webrtc.org
Bug: webrtc:8331
Change-Id: I94ebe77341137aa511c736d18a63e3e8ec0d1bac
Reviewed-on: https://webrtc-review.googlesource.com/12220
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20324}
This CL changes the filter delay detection to rely on the largest peak
while the correctness of the filter is changed to be based on the
performance achieved by the filter.
Bug: webrtc:8397,chromium:774867
Change-Id: I70c953815192478f9a8e0da9f2b8fd9edac3f481
Reviewed-on: https://webrtc-review.googlesource.com/10803
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20321}
This CL changes the AEC3 behavior to be more transparent when there
is uncertainty about the amount of echo in the microphone signal.
Bug: webrtc:8398, chromium:774868
Change-Id: I88e681f8decd892f44397b753df371a1c4b90af0
Reviewed-on: https://webrtc-review.googlesource.com/10801
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20319}
Echo Control is enabled in capture_nonlocked_ when injected.
Renamed echo_canceller3_enabled to echo_controller_enabled.
Bug: webrtc:8346
Change-Id: Icf441f07ce64719358841544da7579feeb7cfdbb
Reviewed-on: https://webrtc-review.googlesource.com/10808
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20311}
...and at least one of our compilers (Visual Studio 64-bit) complains
about it.
BUG=none
Change-Id: I271334f4da564690ff2a16a8322e7ed4a00ae173
Reviewed-on: https://webrtc-review.googlesource.com/10809
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20309}
TBR=sprang@webrtc.org
This is a reland of af721b72cc1bdc5d945629ad78fbea701b6f82b9
Original change's description:
> Remove sent framerate and bitrate calculations from MediaOptimization.
>
> Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
>
> Store sent frame info in map to solve potential issue where sent framerate statistics could be
> incorrect.
>
> Bug: webrtc:8375
> Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
> Reviewed-on: https://webrtc-review.googlesource.com/7880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20225}
Bug: webrtc:8375
Change-Id: I06ea90ae8646ba11ddd8ddceb82ea82d75ae2109
Reviewed-on: https://webrtc-review.googlesource.com/11320
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20308}
Unit test now checks that ADM:Init() works before any test runs.
It means that all tests will be skipped on bots that lack Pulse
support which is as how it worked before this CL as well. But then,
we detected the lack of support by checking that the audio layer had
changed from Pulse to Alsa.
As a consequence, I also decided to inject fake/mock ADMs in more
unit tests. One was actually already injected for other reasons
(see https://codereview.webrtc.org/2997383002/) but it had accidentally
been "reverted" later in combination with other changes.
To summarize: before this change we had a set of unit tests where real
audio was tested but it was not the intention of the test or required.
In addition, some Linux bots (VM:s) did not support PulseAudio and on
them the tests relied on a fallback mechanism to ALSA to work, i.e.,
we had a rather complex dependency on hardware. This dependency has now
been removed and it should result in more stable tests.
Bug: webrtc:7306, webrtc:7806
Change-Id: Ia0485658c04a4ef3b3f2dc0d557d73738067304b
Reviewed-on: https://webrtc-review.googlesource.com/8640
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20307}
Since WEBRTC_ANDROID is defined by WebRTC while ANDROID is defined by
Chromium we should stop using ANDROID in WebRTC source code.
Bug: webrtc:8400
Change-Id: I1d59caaabd8af2423e86476b72e0e9185e6c7a3a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/10805
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20306}
Added EchoCanceller3Factory that implements EchoControlFactory and can
be used for injecting EchoCanceller3 into Audio Processing Module.
Renamed InitializeEchoCanceller3 to InitializeEchoController.
Bug: webrtc:8346
Change-Id: I47078da6a49aca1ee41f6a9d5b7b8e91bb5c11a3
Reviewed-on: https://webrtc-review.googlesource.com/9220
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20299}
Specifically:
ScreenCapturerTest.StartCapture
ScreenCapturerTest.Capture
WindowCapturerTest.Capture
Also adding a DCHECK for the capturer actually being created, since it
seems like that's the problem.
TBR=zijiehe@chromium.org
Bug: webrtc:7830
Change-Id: I200dc0c15f5039b95f591597bc00d3f1084ae876
Reviewed-on: https://webrtc-review.googlesource.com/9562
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20297}
The field trial effects two things: after a frame length change the IAT
histogram is scaled to prevent an immediate change in target buffer
level. Also, the peak history in the delay peak detector is cleared,
because the size of the peaks is stored in number of packets (which
will be incorrect after a frame length change).
Bug: webrtc:8381
Change-Id: I214b990f6e5959b655b6542884a7f75da181a0d8
Reviewed-on: https://webrtc-review.googlesource.com/8101
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20284}
Nothing is different from the original change.
The original change was reverted because it was thought it might
have caused an outage for the windows bots. (It didn't.)
This is a reland of 28addd03c1040fc5a634ed92a7edde2aea811e79
Original change's description:
> Make it possible to isolate bwe_simulations_tests
>
> Add missing data dependencies and add it to gn_isolate_map.pyl
>
> Bug: chromium:749648
> Change-Id: I6b6c1bb2e4d647471a2747042788a691ce2e1e5d
> Reviewed-on: https://webrtc-review.googlesource.com/8721
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20258}
Bug: chromium:749648
No-Try: true
Change-Id: I5531c5f8b58e0d5b8b6ceeea90acb4bd8e6e6e4f
Reviewed-on: https://webrtc-review.googlesource.com/9320
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20280}
In preparation of coming CLs that will add an AGC interface to make the
gain controller injectable.
This CL simplifies AGC2 (dummy sub-module of audioproc_f) since it only
implements the fixed digital mode with hard-clipping - i.e., no limiter
is used.
The AGC2 config now includes the fixed gain to apply and audioproc_f
has been adapted accordingly.
Finally, this CL slightly simplifies the AGC2 integration into APM.
This CL is a continuation of https://codereview.webrtc.org/2995043002/
Bug: webrtc:7494
Change-Id: I3d554ea4dc6208928352059feb14987edabf14c7
Reviewed-on: https://webrtc-review.googlesource.com/4661
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20278}
The factory for EchoControl is changed from an rtc::Callback1 to an
interface. This avoids using rtc::Callback1 outside of WebRTC.
This also makes the EchoControl factory more similar to other
factories in the code base.
Bug: webrtc:8345
Change-Id: Ie61b9416ed771f8c756326736d17e339eb768469
Reviewed-on: https://webrtc-review.googlesource.com/8900
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20272}
For each single experiment, a URL is defined by adding a specific anchor.
A URL can be copied using the button beneath the score of the experiment
one would like to share.
This CL also includes a few optimizations and fixes:
- JS and CSS are minified
- Dialog close event listener added, this fixes a small bug preventing
the play out audio to stop when pressing ESC instead of using the close
button
- Snackbar notifications added
- Simple unit test for the export module
BUG=webrtc:7218
Change-Id: Iad00ce69094a5968ee0520d105d59656cfafa4e2
TBR=
Change-Id: Iad00ce69094a5968ee0520d105d59656cfafa4e2
Reviewed-on: https://webrtc-review.googlesource.com/7960
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20266}
When the test was created, it was disabled for mobile platforms from
the beginning. This is likely a copy-paste from the related
NetEqDecodingTest.TestBitExactness which includes testing codecs not
supported on mobile platforms (e.g., iLBC). This restriction is not
needed for the Opus-only test.
The test remains disabled for iOS, since none of the bots actually run
the relevant test binary on actual iOS devices.
Bug: none
Change-Id: I9071e0e32c83b62c8c7af59ac1cb3e46227f8e8e
Reviewed-on: https://webrtc-review.googlesource.com/8561
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20264}
Add missing data dependencies and add it to gn_isolate_map.pyl
Bug: chromium:749648
Change-Id: I6b6c1bb2e4d647471a2747042788a691ce2e1e5d
Reviewed-on: https://webrtc-review.googlesource.com/8721
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20258}
This CL enables a factory method creating acoustic echo cancellers
that inherit EchoControl to be injected into the audio processing
module. AudioProcessing will call the factory method to create an
instance of the EchoControl subclass when needed. In the event of
sample rate changes, AudioProcessing will recreate the object using
the factory method.
Bug: webrtc:8346
Change-Id: I0c508b4d4cdb35569864cefaa0e3aea2555cc9b9
Reviewed-on: https://webrtc-review.googlesource.com/7742
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20251}
This reverts commit af721b72cc1bdc5d945629ad78fbea701b6f82b9.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Remove sent framerate and bitrate calculations from MediaOptimization.
>
> Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
>
> Store sent frame info in map to solve potential issue where sent framerate statistics could be
> incorrect.
>
> Bug: webrtc:8375
> Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
> Reviewed-on: https://webrtc-review.googlesource.com/7880
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20225}
TBR=asapersson@webrtc.org,sprang@webrtc.org
Change-Id: Ic914f03ff7065ac410ae06b6f82b56a935399b66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8375
Reviewed-on: https://webrtc-review.googlesource.com/8480
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20248}
Temporary files created by AudioFormat tests in modules_unittest are
removed after each test case rather than after the whole suite is
finished. This saves disk space on the running device.
Bug: webrtc:8344
Change-Id: Iace3a7a62bb06e15fa596caf32da873944654c9a
Reviewed-on: https://webrtc-review.googlesource.com/8100
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20244}
Remove REMB accessor as used for debug checks only.
Merge SetRembData and SetRembStatus(true) eliminating
state 'remb can be send, but no data available yet'
Bug: None
Change-Id: I4c1c19435657e5cde02a17de90ec6de9f00b7daf
Reviewed-on: https://webrtc-review.googlesource.com/7983
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20240}
This CL changes the aggregation of the matched filter delay
estimates in AEC3 to using a histogram approach.
Bug: chromium:773541,webrtc:8379
Change-Id: I5322c65858188599397ef5716fecdebc34852e6a
Reviewed-on: https://webrtc-review.googlesource.com/8261
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20234}
This CL changes the tuning of AEC3 to increase the transparency.
In particular:
-The present parameters are re-tuned.
-An oversuppression factor is added in the newly added soft-knee in
the NLP gain. The purpose of this is to avoid fluctuations in the
residual echo.
-The dynamics of the computed gain are bounded to ensure that the
specified gain characteristics are realizable without echo leakage.
This also adds robustness against echo leakage in frequency regions
that are poorly estimated.
This change was needed to avoid echo leakage from the above
tunings.
Bug: chromium:773543,webrtc:8378
Change-Id: If8acc41c1423a6a2fa6f8c4daf2735c86f0b529a
Reviewed-on: https://webrtc-review.googlesource.com/8262
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20231}
Add RateTracker for sent framerate and bitrate in SendStatisticsProxy.
Store sent frame info in map to solve potential issue where sent framerate statistics could be
incorrect.
Bug: webrtc:8375
Change-Id: I4a6e3956013438a711b8c2e73a8cd90c52dd1210
Reviewed-on: https://webrtc-review.googlesource.com/7880
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20225}
This CL adds a smooth rampup of the NLP gain in AEC3.
Bug: webrtc:8361
Change-Id: I49aa75904751ffe9150db1572271fe7a26232449
Reviewed-on: https://webrtc-review.googlesource.com/7740
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20213}
This CL separates the NLP gain computation for the different variants
of echo estimation. This simplifies the setting of tuning
parameters, with resulting transparency improvements and increased
echo removal performance.
Bug: webrtc:8359
Change-Id: I9b97064396fb6f6e2f418ce534573f68694390a1
Reviewed-on: https://webrtc-review.googlesource.com/7613
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20209}
This CL adds the ability to set a default echo path delay to use
in AEC3 when there is prior knowledge about the delay in the echo
path.
Bug: webrtc:8358
Change-Id: Ie368f9a6dec9f412e09bf0e095f89d84305045f9
Reviewed-on: https://webrtc-review.googlesource.com/7604
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20204}
