This reverts commit 4ea50c2b421ae3e40d1d02b8eb8c5802288b181e.
Reason for revert: This change is causing crashes in video calls.
RTCCVPixelBuffer.mm - line 120
Compare is asserting as 420f is not 420v
Original change's description:
> Add unit tests for RTCCVPixelBuffer and ObjCVideoTrackSource.
>
> This CL also fixes a couple of bugs found in the toI420 method for
> RTCCVPixelBuffers backed by RGB CVPixelBuffers.
>
> Bug: webrtc:9007
> Change-Id: I19ab8177f4b124a503cfda9f0166bd960f668982
> Reviewed-on: https://webrtc-review.googlesource.com/64940
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22656}
TBR=andersc@webrtc.org,kthelgason@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:9007
Change-Id: I500514ce05dd0555f8c4a05010ad52bd67c2fed3
Reviewed-on: https://webrtc-review.googlesource.com/65561
Commit-Queue: JT Teh <jtteh@webrtc.org>
Reviewed-by: JT Teh <jtteh@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22686}
No functional change. This makes the code more generic
and this reduces the size of screen_capturer_mac.mm
Bug: webrtc:8652
Change-Id: I37743ba385fea5e1b40df3b094bfc321b8d796ae
Reviewed-on: https://webrtc-review.googlesource.com/65082
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Commit-Queue: Zijie He <zijiehe@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22684}
The inclusion of <d3d9.h> is probably a leftover from the past. As of
today system_wrappers/cpu_info.cc doesn't need access to Direct 3D.
TBR=tommi@webrtc.org
Bug: webrtc:9073
Change-Id: I679d161f4b1d098a7864d82e4a52fa70d57aae84
Reviewed-on: https://webrtc-review.googlesource.com/65440
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22683}
The MID header extension is handled by the RtpTransport
which lives above Call and takes care of demuxing to SSRC.
Bug: webrtc:4050
Change-Id: I27135e296ae9c7b15e926ba17547c26c75684ad6
Reviewed-on: https://webrtc-review.googlesource.com/65025
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22682}
After deploying the new DesktopAndCursorComposer ctor in chromium in cl
https://chromium-review.googlesource.com/c/chromium/src/+/980668
The old ctor and relative stuffs can be removed now.
Bug: webrtc:9072
Change-Id: Ibbf23a374883c096b13169bd5289a2d4ece539fa
Reviewed-on: https://webrtc-review.googlesource.com/65341
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22679}
This method is only used for logging and is blocking further refactoring
work. Once the refactoring and cleanup of the external AudioDeviceModule
is complete, we can revisit what logging we want and need and add it in
a cleaner way.
Bug: webrtc:7452
Change-Id: If08bcfb37860e9e7b9b5105cb75f748b53775f69
Reviewed-on: https://webrtc-review.googlesource.com/65460
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22678}
This reverts commit 04dd1768625eb2241d1fb97fd0137897e703e266.
Reason for revert: Regression in ramp up perf tests.
Original change's description:
> Reland "Move rtp-specific config out of EncoderSettings."
>
> This is a reland of bc900cb1d1810fcf678fe41cf1e3966daa39c88c
>
> Original change's description:
> > Move rtp-specific config out of EncoderSettings.
> >
> > In VideoSendStream::Config, move payload_name and payload_type from
> > EncoderSettings to Rtp.
> >
> > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > and should perhaps be renamed in a follow up cl. It's no longer
> > passed as an argument to VideoCodecInitializer::SetupCodec.
> >
> > The latter then needs a different way to know the codec type,
> > which is provided by a new codec_type member in VideoEncoderConfig.
> >
> > Bug: webrtc:8830
> > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22532}
>
> Bug: webrtc:8830
> Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> Reviewed-on: https://webrtc-review.googlesource.com/63721
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22595}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Bug: webrtc:8830,chromium:827080
Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
Reviewed-on: https://webrtc-review.googlesource.com/65520
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22677}
The VolumeLogger class contains enough logic to deserve its own file.
Also, I want to potentially remove WebRtcAudioManager.java but keep
volume logging. One problem I see with the VolumeLogger is that it
spawns a new thread, and we should try to keep the number of threads
in WebRTC to a minimum. Right now we use excessively many threads.
Bug: webrtc:7452
Change-Id: I4dd8ffb4265903926f0b372715fc6b876fe5d393
Reviewed-on: https://webrtc-review.googlesource.com/65401
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22676}
Added ToString functions in a separate header and move the ostream
operators to a test only header.
Bug: webrtc:8982
Change-Id: If547173aa39bbae2244531e2d3091886f14eae31
Reviewed-on: https://webrtc-review.googlesource.com/65280
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22674}
The class called AudioDeviceModule today is an implementation of a
future interface. We want to reserve the name AudioDeviceModule for
the actual interface. The implementation class has been renamed to
JavaAudioDeviceModule. 'Java' here refers to the fact that the
implementation is using android.media.AudioRecord as input and
android.media.AudioTrack as output, and this is opposed to native
AudioDeviceModule implementations such as OpenSLES and AAudio.
Bug: webrtc:7452
Change-Id: Ifc243c2e169b12a50128ee3252f06d574aa7b358
Reviewed-on: https://webrtc-review.googlesource.com/65400
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22673}
The configurator decides number of spatial layers, their resolution
and bitrate thresholds based on given input resolution and maximum
number of spatial layers.
The allocator distributes available bitrate across spatial and
temporal layers. If there is not enough bitrate to provide acceptable
quality for all spatial layers allocator disables enhancement layers
one by one until the condition is met or number of layers is reduced
to one.
VP9 SVC related unit tests have been updated. Input resolution and
bitrate in these tests have been increased to the level enough to
provide desirable number of spatial layers.
Bug: webrtc:8518
Change-Id: I9df790920227c7f7dd4d42a50a856c22f0f4389b
Reviewed-on: https://webrtc-review.googlesource.com/60340
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22672}
jsr305 is necessary dependency for Nullable annotations.
Also adds a flag to release_aar.py to specify the build directory
manually. This makes it easier to test the script without full
recompilation.
Bug: webrtc:8881
Change-Id: Ib4b8cd4592ced9c92ca2810928bcbb6173d2164e
Reviewed-on: https://webrtc-review.googlesource.com/65081
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22671}
Static const objects can cause what's called a "static initialization
order fiasco". This CL removes the statically initialized network units
in favor of constexpr defined versions available via static functions.
Bug: webrtc:8415
Change-Id: Ib1b316ae007481c52a53b2d1bb0352a630a220e2
Reviewed-on: https://webrtc-review.googlesource.com/65164
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22670}
We update the configuration settings for AGC2. We also update their
effects. Now, 'gain_controller2.enable=true' means 'first run Adaptive
AGC2; then run AGC2 limiter'.
Previously, only the AGC2 limiter was implemented. To run that, one
had to set both 'gain_controller2.enable=true' and
'gain_controller2.enable_limiter=true'.
This setting also enables adaptive AGC2 in the test tool 'audioproc_f'.
Bug: webrtc:7494
Change-Id: I0d5dfe443f2cdc0ecf3aa4054442dab6276d284d
Reviewed-on: https://webrtc-review.googlesource.com/64990
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22669}
This patch adds TurnPort::Release that release a TURN allocation
by sending a REFRESH with lifetime 0 without destroying the object.
This allows for graceful shutdown of a TurnPort that can e.g be used
for mobility.
Bug: webtrc:9067
Change-Id: I1e4d9232ae08d6fe14f5612f776a541c03c3beec
Reviewed-on: https://webrtc-review.googlesource.com/64722
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22666}
IceConfig contains a set of parameters that affect the behavior of ICE.
Inconsistent or conflicting parameters lead to erroneous or
unpredicatble behavior in the network stack. Sanity checks are now added
to validate IceConfig.
TBR=magjed@webrtc.org
Bug: webrtc:8993
Change-Id: I708bc3f1ef970872754a82a47a509bda15061ca6
Reviewed-on: https://webrtc-review.googlesource.com/60847
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22664}
We prefer the Google style guide over the chromium guide in this case:
avoid forward declarations whenever possible. We don't have the same
compilation time issue that chromium does, so we can afford to do this.
The majority of our code already follows this guideline, as far as I'm
aware, though some forward declarations are still used to resolve
circular dependencies.
Bug: None
Notry: true
Change-Id: I712e0361acf76217067b13b8b3e4bc931d2a0238
Reviewed-on: https://webrtc-review.googlesource.com/8800
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22662}
The OpenSLES engine is currently managed by the AudioManager which is
a generic class shared between different kinds of audio input/output.
This CL moves the responsibility of the OpenSLES engine to the actual
OpenSLES implementations.
Bug: webrtc:7452
Change-Id: Iecccb03ec5cd12ce2f3fdc44daaedae27aecf88b
Reviewed-on: https://webrtc-review.googlesource.com/64520
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22661}
This CL further decreases the look window size, as well
as the effect of the look window used by AEC3 when is is
in the nonlinear mode.
Bug: chromium:826720,webrtc:9083
Change-Id: I193539c0af74eea18d2821a3b7e1fae2f783d38a
Reviewed-on: https://webrtc-review.googlesource.com/65161
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22659}
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}
This CL also fixes a couple of bugs found in the toI420 method for
RTCCVPixelBuffers backed by RGB CVPixelBuffers.
Bug: webrtc:9007
Change-Id: I19ab8177f4b124a503cfda9f0166bd960f668982
Reviewed-on: https://webrtc-review.googlesource.com/64940
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22656}
This CL refactors the way RecordedAudioToFileController is connected to
AudioRecord. Instead of allowing to dynamically set and update the
AudioSamplesCallback, it's set once at start time and then stopping is
implemented in RecordedAudioToFileController by simply ignoring calls to
onWebRtcAudioRecordSamplesReady.
The reason for this CL is to reduce the amount of methods we need to
add to the future AudioDeviceModule interface. The more functionality
we can move to creation time in the ctor, the less methods we need to
have in the interface.
Bug: webrtc:7452
Change-Id: I462df275d8579c848e1d2c86cbd8e881da89cbf3
Reviewed-on: https://webrtc-review.googlesource.com/64988
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22653}
This reverts commit 8ac9bb4d52a687b34158dc52c8c25830b23b8333.
Reason for revert: <INSERT REASONING HERE>
Original change's description:
> Added BBR network controller.
>
> BBR is a congestion control method that is initially developed for TCP.
> This CL adds an implementation of BBR ported from QUIC for use with
> WebRTC. An upcoming CL enables it via a field trial.
>
> Bug: webrtc:8415
> Change-Id: Ie4261d2e43bafa15aa928a7cadcfec256107cdbc
> Reviewed-on: https://webrtc-review.googlesource.com/39788
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22647}
TBR=philipel@webrtc.org,srte@webrtc.org
Change-Id: Ife354d40bfc755f899cf485f3308575516206997
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8415
Reviewed-on: https://webrtc-review.googlesource.com/65180
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22651}
Reduce synchronization in the class significantly and not hold a lock
while calling out to external implementations.
* Rewrite tests to use a real ProcessThread.
* Update some code to use C++ 11 constructs & library features.
Bug: webrtc:9064
Change-Id: I240a819efb6ef8197da3f2edf7acf068d2a27e8b
Reviewed-on: https://webrtc-review.googlesource.com/64521
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22649}
There is no need to use real video as input for encoder in unit tests.
Using generator simplifies testing on mobile devices (no need to upload
files to device).
Bug: none
Change-Id: Ic48609cc6f8eecf90d9956edfdd33135be949398
Reviewed-on: https://webrtc-review.googlesource.com/64526
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22648}
BBR is a congestion control method that is initially developed for TCP.
This CL adds an implementation of BBR ported from QUIC for use with
WebRTC. An upcoming CL enables it via a field trial.
Bug: webrtc:8415
Change-Id: Ie4261d2e43bafa15aa928a7cadcfec256107cdbc
Reviewed-on: https://webrtc-review.googlesource.com/39788
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22647}
This function is not present in std::optional
The only use of MoveValue doesn't need move since
copying underneath struct is as correct and as fast as moving
Bug: webrtc:9078
Change-Id: Ic0c87e50ffd8f6c024759b14ceeb8922b5d3a6fd
Reviewed-on: https://webrtc-review.googlesource.com/64986
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22646}
This is used by BBR which is introduced in a future CL.
Bug: webrtc:8415
Change-Id: Ie5b3e6e58b7c9c7a35fc21acb636103d7f5daec3
Reviewed-on: https://webrtc-review.googlesource.com/64920
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22643}
Needed to be able to add an SdpVideoFormat member to
VideoEncoderConfig or other move-only classes.
Bug: webrtc:8830
Change-Id: Ie15dbfec616b059e1492d2c17ebd210f0edbe00f
Reviewed-on: https://webrtc-review.googlesource.com/64983
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22642}
The level estimator (AdaptiveModeLevelEstimator) produces a biased
estimate of the speech level. In our model, we use another module
(the SaturationProtector) to compute the bias. This CL contains the
estimator and a stub of the saturation protector.
Bug: webrtc:7494
Change-Id: I0df736d0346063f544fa680b4cc84177ea548545
Reviewed-on: https://webrtc-review.googlesource.com/64820
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22641}
