and in DegradedCall. In DegradedCall - ThreadPacketReceiver is no longer needed.
Implementation of DeliverRtpPacket is done in preparation of https://webrtc-review.googlesource.com/c/src/+/290540, where the parsed packet will be propagated to Call without extra parsing.
Bug: webrtc:7135, webrtc:14795
Change-Id: Ic068105d6d1f337afc6b4539b0e7184e736e7ee0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290704
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39048}
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
This is in the webrtc-stats spec at
https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode.
This adds the scalability mode to CodecSpecificInfo which is used to
plumb the modes for each simulcast layer.
TBR=orphis@webrtc.org
Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals.
Bug: webrtc:14730
Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38847}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
This is a reland of commit c1d5fda22c8ae456950c5549d22d099b478c67e2
Original change's description:
> Add documentation, tests and simplify webrtc::SimulatedNetwork.
>
> This CL increases the test coverage for webrtc::SimualtedNetwork, adds
> some more comments to the class and the interface it implements and
> simplify the logic around capacity and delay management in the
> simulated network.
>
> More CLs will follow to continue the refactoring but this is the
> ground work to make this more modular in the future.
>
> Bug: webrtc:14525, b/243202138
> Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38388}
Bug: webrtc:14525, b/243202138, b/256595485
Change-Id: Iaf8160eb8f8e29034b8f98e81ce07eb608663d30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280963
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38557}
This reverts commit c1d5fda22c8ae456950c5549d22d099b478c67e2.
Reason for revert: This CL created thousands of metric alerts in the perf tests. It's possible that these are all expected, but since mbonadei@ is OOO right now, I think it's better to revert, and have him re-land when he is back.
Most alerts are here: https://bugs.chromium.org/p/webrtc/issues/detail?id=14549
Original change's description:
> Add documentation, tests and simplify webrtc::SimulatedNetwork.
>
> This CL increases the test coverage for webrtc::SimualtedNetwork, adds
> some more comments to the class and the interface it implements and
> simplify the logic around capacity and delay management in the
> simulated network.
>
> More CLs will follow to continue the refactoring but this is the
> ground work to make this more modular in the future.
>
> Bug: webrtc:14525, b/243202138
> Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38388}
Bug: webrtc:14525, b/243202138
Change-Id: I5bc56c954bb12e7c27cb859e838f0b7a89e006f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279522
Owners-Override: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38415}
This CL increases the test coverage for webrtc::SimualtedNetwork, adds
some more comments to the class and the interface it implements and
simplify the logic around capacity and delay management in the
simulated network.
More CLs will follow to continue the refactoring but this is the
ground work to make this more modular in the future.
Bug: webrtc:14525, b/243202138
Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38388}
This spills to a few more clasess....
Change-Id: Iea79e3b4ac86b30db6f13da89a47ab7000c5440a
Bug: webrtc:14502
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277803
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38334}
Instead of re-using the sender task queue, a new task queue will
suffice.
Bug: webrtc:14445
Change-Id: Ia7395ace2f0bb66bf9e76e3783b208f2cd0385dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275771
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38332}
This cl move VideoEncoderConfig from api/ to video/config.
VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.
brandt@ think that the reason these were in api/ in the
first place had to downstream project.
Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).
Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
Ability to emulate degraded networks using DegradedCall has not
been covered by tests and it is crashing due to DCHECKs.
Fix threading issues so this no longer crash.
Bug: None
Change-Id: I9276dfb1f71762faa02146af0bfaab713bebb7f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276060
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38216}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
This is unused at the moment and webrtc::SimulatedNetwork is going
through a refactoring, to keep things simple and well tested this CL
removes CoDel but nothing blocks us from re-implementing it when needed.
No-Try: True
Bug: webrtc:14426
Change-Id: Ie7d40d20a66d3939fc7d3251c47e4f13f3869a27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274407
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38032}
This way call no longer needs dedicated process thread
Bug: webrtc:7219
Change-Id: I8ab677b1e6b909eeb726aefed5e6d10ce4bc43b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265921
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37279}
To make it usable in tests without depending on all of CallTest.
Bug: None
Change-Id: Ie3102ab71bcfe3862dd6c35d3285098e961e54df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262807
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36932}
GetRtpExtensions() is still used in one corner case for audio receive
streams, so GetRtpExtensions has migrated to AudioReceiveStream.
Updated FlexfecReceiveStream config management (incl. pass by value) and
now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl.
Call GetRtpExtensionMap() from call.cc instead of constructing one on
the fly for each rtp packet (for video packets at least).
Bug: webrtc:11993
Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36839}
While the target has a restricted visibility, since it was in rtc_base_approved
public deps, a lot of targets were able to bypass the visibility check.
So we remove the visibility restrictions and use the dependency explicitely
everywhere instead.
Bug: webrtc:8603
Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36566}
This is part of a large scale effort to increase adoption of
absl::string_view across the WebRTC code base.
This CL adds absl::string_view versions of the OnLogMessage functions in
rtc::LogSink. The const std::string& versions are kept for now since
downstream clients use subclasses of LogSink and need to be migrated
before these are removed.
Bug: webrtc:13579
Change-Id: I57bb72ad503805ff0ca16f1d7aece2d44c65cb73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Owners-Override: Xavier Lepaul <xalep@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36189}
Wires up DecodeSynchronizer in Call if there is a Metronome injected
into the PeerConnectionFactoryDependencies.
Change-Id: I362cd12648bfa0c32e73111fcd0f3296fca2b275
Bug: webrtc:13658
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251341
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35996}
Since another CL [1] grew a bit in scope, this CL needs to land first,
coupled with downstream updates.
[1]: https://webrtc-review.googlesource.com/c/src/+/244083
Bug: webrtc:12517, webrtc:11993
Change-Id: I80d826a0d3f3240a2cecf8cbe09842cba0902b16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/244085
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35611}
Task posted by OnSendRtp might be scheduled after `send_stream_` is
destroyed. Fix by using a PendingTaskSafetyFlag, killed from the
OnStreamsStopped callback.
Bug: webrtc:12726
Change-Id: I935917a3d80e82c3536261d72059448fb7aac00d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228643
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34754}
Except for a use of std::multimap (for which there currently is no
drop-in replacement), use webrtc::flat_map and flat_set for increased
performance.
The number of ssrcs/mids/payload types/etc is likely to be small and is
generally updated very rarely, but looked up a lot.
RtpDemuxer::ResolveSink's use of std::map represents about 0.6% CPU in
heavily loaded media servers. It does quite a few map lookups, and by
taking advantage of the increased cache locality of the flat_map and
flat_set containers, performance should be increased.
A previous attempt to use std::unordered_map failed in change 216243,
which was reverted. This is the second attempt.
Bug: webrtc:12689
Change-Id: Ifdbec63b2fd8f2f52e45ebaf12834b11dd7a41c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/224662
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34429}
