Large gaps can cause issues in NetEq simulations, so the simulation is
ended whenever we encounter one. However, the time span of the gap is
still included in the simulation time, leading to incorrect results.
Bug: webrtc:10337
Change-Id: I94a1a0b46259e3718b1b73522a3886a17bedbb7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190287
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32514}
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.
Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
This code used to have a reader-writer lock, and call
std::queue::pop() with only a reader lock, which appears unsafe. Code
changed to use a plain webrtc::Mutex.
Bug: webrtc:12102
Change-Id: Icbea17a824c91975dfebd4d05bbd0c21e1abeadc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190700
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32511}
In the video quality tests the codec-level min bitrate is sometimes set
as if single-stream simulcast will be used. When VP9 spatial layers are
then generated the will get new appropriate min bitrate levels.
The encoder adjuster can however look at the codec level min bitrate
and incorrectly adjust the bitrate up if it is set too high.
This CL removes the codec-level min bitrate if svc is used.
Bug: webrtc:12080
Change-Id: I563a57f3031c90c116448f1d255d3b6711f4ee75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32509}
INCREASING_ID, which is the default mode, triggers HW reset in chromium
decoder wrapper. Set eSpsPpsIdStrategy to SPS_LISTING to prevent that.
Note that WebRTC always resets the encoder on resolution change. This
makes all strategies except INCREASING_ID essentially equivalent to
CONSTANT_ID.
Bug: chromium:1111273
Change-Id: I37405c97b3390f812d1dcaa111694b3b1d638035
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190440
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32505}
"warning: control reaches end of non-void function [-Wreturn-type]"
Reported by gcc (8.3)
In all the reported cases, the end of function is never actually
reached. Add RTC_CHECK(false) to ensure the compiler is aware that
this path is a dead-end.
Bug: webrtc:12008
Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com>
Cr-Commit-Position: refs/heads/master@{#32503}
And add a Create() method to the class.
This makes it possible to experiment with subclassing the
SdpOfferAnswer object without modifying the PeerConnection.
Bug: webrtc:11995
Change-Id: I0a7c91a8999858ddcb1ea59ac4eb9a3b0663b0f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190288
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32501}
These can now be initialized in the constructor and are not touched
explicitly in the destructor.
Bug: none
Change-Id: I3d294b15463a8d02bbe7e37fb14eefd017d5c1e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190284
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32494}
This feature is active if and only if the RTP header extension
playout-delay is used with min playout delay=0 and max playout delay>0.
In this case, a maximum composition delay will be calculated and attached
to the video frame as a signal to use the low-latency renderer algorithm,
which is landed in a separate CL in Chromium.
The maximum composition delay is specified in number of frames and is
calculated based on the max playout delay.
The feature can be completetly disabled by specifying the field trial
WebRTC-LowLatencyRenderer/enabled:false/
Bug: chromium:1138888
Change-Id: I05f461982d0632bd6e09e5d7ec1a8985dccdc61b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190141
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32493}
This CL removes the circular shift to compute the convolution in
`ComputeLpResidual()`, which is now 4x faster (benchmarked with the
`RnnVadTest.DISABLED_ComputeLpResidualBenchmark` unit test).
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: Ia7767d9b57378c12c8ff31f58fea03905be5c5de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189964
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32491}
Use hard-coded constants previously computed at runtime within a loop in
`DenoiseAutoCorrelation`.
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: I02c2fff7dc7153ea2ab8a27cad8a479a0f029996
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189963
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32489}
Now this requires explicitly passing the
WebRTC-LegacyTlsProtocols/Enabled/ field trial flag or an override.
Bug: webrtc:10261
Change-Id: Ib880bcc50cec0a21dcaa4784c228cacb020e5568
NOKEYCHECK: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190282
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32488}
After this change, SdpOfferAnswerHandler implements a read-only
interface called SdpStateProvider, which allows enough access
for WebRtcSessionDescriptionFactory to learn what it needs to know.
Bug: webrtc:12060
Change-Id: Ic888b5027b2df5fee407d32b89da66ff044c40de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190145
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32486}
This allows making more members (including IsUnifiedPlan) const in a future CL.
Also revises the test for ReportUsageHistogram to use a configuration member
variable rather than a hook function in PeerConnectionFactory.
Bug: webrtc:12079
Change-Id: I6f1af7d6164c8a0d8466f76378a925d72d57d685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190280
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32485}
As discussed on a design review, the name RoboCaller is not clear
enough and switching to CallbackList will provide readability benefits.
Bug: webrtc:11943
Change-Id: I010cf0a91b5323e4e9c96b83703be7af1e67439c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190142
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32478}
This changes the default behavior to use pacing factor of 1.1x instead
of 2.5x, it also sets libvpx rate controler as trusted, turns on the
encoder pushback mechanism and sets spatial hysteresis to 1.2.
The unused "dynamic rate" settings in libvpx is removed.
The new settings matches what has been used in chromium since 2019.
If needed, the legacy behavior can be enabled using the field trial
WebRTC-VideoRateControl.
Bug: webrtc:10155
Change-Id: I8186b491aa5bef61e8f568e96c980ca68f0c208f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186661
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32477}
Also move ssrc_generator and audio/video options, as well as some
signal handling that's related.
These variables were not referenced in peer_connection.cc any more.
Bug: webrtc:11995
Change-Id: I29f8661afad488380d256220b35330233e8233e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189967
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32471}
This is a reland of 37d41eea047799ccca2f09c941870c26ee3ffc0a
Original change's description:
> Reland "Remove placeholder Obj-C headers and use angle-bracketed headers."
>
> This is a reland of 6bfad33fd866e682c871c2ef2172b70b609593d1
>
> Original change's description:
> > Remove placeholder Obj-C headers and use angle-bracketed headers.
> >
> > sdk/objc/Framework/Headers are just a placeholder headers
> > for backward compatibility and I don't think it is really need this for now.
> > Instead, we can generate the framework header in
> > ios/mac_framework_bundle_with_umbrella_header.
> > Also clang supports the -Wquoted-include-in-framework-header warning,
> > and in Xcode 12, it's in Xcode's recommended settings. This warnings
> > can be avoided by replacing double-quoted includes with angle-bracketed
> > includes when generate framework headers.
> >
> > No-Presubmit: True
> > Bug: webrtc:9627, webrtc:11984
> > Change-Id: I3f6258dfa77a5acee669614005b2747feee35e39
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185920
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32343}
>
> TBR=tommi@webrtc.org, andersc@webrtc.org
>
> No-Presubmit: True
> Bug: webrtc:9627
> Bug: webrtc:11984
> Change-Id: I8f44232f1a70b8ff2ce6a4b4792f0a18472fcec3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187280
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32429}
TBR=tommi@webrtc.org, andersc@webrtc.org
No-Presubmit: True
Bug: webrtc:9627
Bug: webrtc:11984
Change-Id: Ida92b8864dffaea37d3053d3c00381644988b54e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32470}
When deinitializing usrsctp, we attempt to call usrsctp_finish in a loop
for three seconds (it may fail because another sctp thread is holding a
reference to something). If the three seconds run out, usrsctp is left
in a still initialized state, and bad things happen down the road if
usrsctp_init is called in the state.
Bug: chromium:1138878
Change-Id: I9c24d51d5a274b06bdf4183261694fc2989136c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32467}
