This CL fixes a bug in the feedback based GoogCC where packets lost
was swapped with expected packets received. Since this version of
GoogCC isn't yet used this wasn't discovered. There was also a lack
of unit test coverage. To ensure reasonable behavior, unit tests was
added.
Unit tests was also converted from relevant unit tests on send side
congestion controller for the regular GoogCC controller.
Bug: webrtc:9586
Change-Id: I83c40ff4766104820cb72ec1e8b95c5782def19a
Reviewed-on: https://webrtc-review.googlesource.com/59401
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24498}
This reverts commit 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a.
Reason for revert: Makes the perf test RampUpTest.AudioTransportSequenceNumber fail on windows, almost every time.
Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
>
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
>
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}
TBR=henrika@webrtc.org,nisse@webrtc.org,titovartem@webrtc.org
Change-Id: I8867a22d695494bd5abfda6a97f0719cb3ff3d66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:3380
Reviewed-on: https://webrtc-review.googlesource.com/96840
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24496}
Return value always passed as the |retransmitted| argument to
ReceiveStatistics::IncomingPacket. The implementation of this method,
StreamStatisticianImpl::IncomingPacket, can call its own
IsRetransmitOfOldPacket, which is demoted to a private method.
Bug: webrtc:7135
Change-Id: I904db676738689c7a1db4caa588f70e64e3c357d
Reviewed-on: https://webrtc-review.googlesource.com/95649
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24494}
This CL separates the files under sdk/objc into logical directories, replacing
the previous file layout under Framework/.
A long term goal is to have some system set up to generate the files under
sdk/objc/api (the PeerConnection API wrappers) from the C++ code. In the shorter
term the goal is to abstract out shared concepts from these classes in order to
make them as uniform as possible.
The separation into base/, components/, and helpers/ are to differentiate between
the base layer's common protocols, various utilities and the actual platform
specific components.
The old directory layout that resembled a framework's internal layout is not
necessary, since it is generated by the framework target when building it.
Bug: webrtc:9627
Change-Id: Ib084fd83f050ae980649ca99e841f4fb0580bd8f
Reviewed-on: https://webrtc-review.googlesource.com/94142
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24493}
In addition, let the processing thread loop explicitly, and not use
the deprecated builtin looping in PlatformThread.
Bug: webrtc:3380
Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
Reviewed-on: https://webrtc-review.googlesource.com/96544
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24492}
Merge SetPayloadData into constructor,
Remove payload size member because now used only during construction.
Remove member that should be constant
Bug: None
Change-Id: Ib2083439f466ad9151ce8e54fceede6cef51d955
Reviewed-on: https://webrtc-review.googlesource.com/96740
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24491}
Replaces the VP8 simulcast index and VP9 spatial index formely part of
CodecSpecificInfo.
Bug: webrtc:9378
Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
Reviewed-on: https://webrtc-review.googlesource.com/83161
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24485}
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, webrtc:163
Change-Id: I74cb86c29cebb69dd22083718f1446f18f705cd4
Reviewed-on: https://webrtc-review.googlesource.com/95883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24483}
CustomAudioAnalyzer is an interface of a component into APM that
reads AudioBuffer without changing it.
The APM sub-module is optional. It operates in full band.
As described in the comments, it is an experimental interface which
may be changed in the nearest future.
Change-Id: I21edf729d97947529256407b10fa4b5219bb2bf5
Bug: webrtc:9678
Reviewed-on: https://webrtc-review.googlesource.com/96560
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Valeriia Nemychnikova <valeriian@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24481}
It's reasonable to allow clients implementing their own VideoCodecTests
to decide wether they should run in real-time.
Removes the IsAsyncCodec helper, as the assumptions it made are outdated,
and it is no longer useful.
Bug: None
Change-Id: If766935d4947555af54f499a30cfe553bde4c1ab
Reviewed-on: https://webrtc-review.googlesource.com/95722
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24478}
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}
The old test supported audio but only in combination with a fake ADM.
The new version allows the user to run real video and audio.
Now possible to do:
./out/Debug/video_loopback.exe --audio --use_real_adm
To run the test in loopback using real default audio devices.
By default:
./out/Debug/video_loopback.exe --audio
runs with fake audio devices as before.
Bug: webrtc:9265
Change-Id: Id89924ec0276f929487c71fc6321dcd9cb92693d
Reviewed-on: https://webrtc-review.googlesource.com/96161
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24463}
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}
When FixedGainController::SetGain() is called first on a large value (e.g., 40 dB)
and afterwards on a smaller one (e.g., 0 dB), the limiter used by FixedGainController
takes time (about 10-20 seconds) to converge. During that period, the audio is not
audible and the volume slowly increases.
Even if switching from 40 dB to 0 dB is unlikely, this behavior can be corrected by
resetting the limiter every time that FixedGainController::SetGain() is called.
This eliminates the undesired effect described above even when the transient is short.
Bug: webrtc:7494
Change-Id: I419b8986d2181448b4671cdbbd1c256dfb460216
Reviewed-on: https://webrtc-review.googlesource.com/94902
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24451}
Also adds a running picture id for the old generic format when
kVideoCodecGeneric is used (behind "WebRTC-GenericPictureId" field trial).
Bug: webrtc:9361
Change-Id: I6f232a2663bb60257c97ed3473eb07044d325b90
Reviewed-on: https://webrtc-review.googlesource.com/94842
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24449}
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}
Also consolidate lock operations to public methods only, moving one
CritScope out of UpdateCounters (private) up to IncomingPacket
(public).
Bug: webrtc:7135
Change-Id: I458857d3cfa49961fa34196ffe02cdefd83cec10
Reviewed-on: https://webrtc-review.googlesource.com/96122
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24443}
This will allow us to configure VP9 encoder to produce spatial layers
with different frame rates.
Bug: webrtc:9650
Change-Id: I3a9c58072003b8a8da681d5291d8f7ede7f52fa4
Reviewed-on: https://webrtc-review.googlesource.com/95427
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24435}
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}
The AGC2 is enabled by flipping
AudioProcessing::Config::GainController2::enabled. The flag enables
both AdaptiveAgc and FixedGainController. Before this CL, there was no
way(*) to only enable the FixedGainController. After this CL, it's
also possible to flip the setting
|AudioProcessing::Config::GainController2::adaptive_digital_mode|. The
default is |true|, which is the previous behavior.
* Except for instantiating and setting it up outside of the APM like
it's done in the AudioMixer.
Bug: webrtc:7494
Change-Id: I506e93b6687221ac467f083fa8db3d45c98c1b83
Reviewed-on: https://webrtc-review.googlesource.com/95426
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24432}
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 adds a quiet mode for audioproc_f and hooks up the verbose
output of the AEC3 settings read from the JSON input file to that.
Bug: webrtc:8671
Change-Id: I93bbd1efc6502649da7b2b3e9f7557e9c184b0ed
Reviewed-on: https://webrtc-review.googlesource.com/95700
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24416}
The introduction of the new AEC proxies caused audioproc_f to fail.
This CL corrects audioproc_f so that the AEC2 and AECM echo cancellers
are properly activated using the new AEC proxies.
Bug: webrtc:9535
Change-Id: I48b9deaad873aee597f56ebd33814420024e0d58
Reviewed-on: https://webrtc-review.googlesource.com/95645
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24405}
Videosendstream can be created before capturer starts, so initially the frame resolution may be zero. Add a check to prevent test failure and undesired behavior.
Bug: webrtc:7737
Change-Id: I8f4402e866f45ea1eb112437f866170691a111f6
Reviewed-on: https://webrtc-review.googlesource.com/95102
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24404}
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}
