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}
The cross-correlation function in the anonimous namespace of
lp_residual.cc is used once as to compute auto-correlation coefficients.
Hence, it can be simplified (and renamed).
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: Ie769e42907c15572172a8ad1a34e82ce60c4a386
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189962
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32464}
Use hard-coded constants previously computed at runtime within a loop in
`ComputeAndPostProcessLpcCoefficients`.
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: Iaec178c56937449c3fe9af881354d19cc8c47c7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189961
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32462}
This change makes the transparent mode classifier consider filter
convergence also for microphone signals with very low volume. This is
in order to prevent entering transparent mode when there is low, but
audible, echo.
Furthermore, the suppression gain during transparent mode is restored
to avoid leaks when the echo is too low to be reliably detected by
filter convergence.
Bug: webrtc:10232, chromium:1140452
Change-Id: Idd4f40c4aee7c20baa444afaa5ec604eb65bcfd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189786
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32461}
This factory is only used by SdpOfferAnswerHandler, so it should not
need to depend on PeerConnection.
Bug: webrtc:11995
Change-Id: Ib27d9d9fdf440be7db8890bf0e7520d0c67bde22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189780
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32460}
Add multiple accessors to PeerConnection, and make multiple
formerly private functions public for access from SdpOfferAnswerHandler.
Reducing the surface of PeerConnection is a job to be done iteratively.
Bug: webrtc:11995
Change-Id: Iab176824ae557af84ac934e40ff674a1008a29d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189540
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32459}
