Environment includes propagated field trials that can be later passed to
RemoteBitrateEstimators member, and would allow not to rely on the global field trial string
Bug: webrtc:42220378
Change-Id: Icf75a433c20352b2c22829c2148c92f69a2517aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349645
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42242}
VideoStreamEncoder creates VideoEncoders. To pass an Environment to VideoEncoder, it should be available in the VideoStreamEncoder.
Bug: webrtc:15860
Change-Id: Id89ac024ce61fdd9673bb66f03f94f243fc0c7f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341840
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41861}
There is to reason to have two separate classes as they both represent the same thing.
Done in order to simplify further refactorings.
Bug: webrtc:14928
Change-Id: I33e5fe032c79396fbae970c8732c90eb2252accb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335040
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41561}
This is a reland of commit b39c2a8464
Original change's description:
> FrameCadenceAdapter: align video encoding to metronome
>
> This CL aligns the video encoding tasks to metronome tick which
> similar with the metronome decoding.
>
> Design doc: https://docs.google.com/document/d/18PvEgS-DehClK6twCSCATOlX-j9acmXd-3vjb0tR9-Y
>
> Bug: b/304158952
> Change-Id: I262bd4a5097fdaeed559b9d7391a059ae86e2d63
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327460
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
> Cr-Commit-Position: refs/heads/main@{#41469}
Bug: b/304158952
Change-Id: Icf4e1ad91f5c98f3c32a88ffe4d6277e907353e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333464
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41479}
To allow custom FecController use propagated rather than global field trials
note that there is one FecControllerFactory per peer connection factory,
but FecController is created per peer connection and may use per peer connection field trials.
Bug: webrtc:10335
Change-Id: Id25bfaf4b49d4f6d551730c8fd55596ddc49ab47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41478}
This CL aligns the video encoding tasks to metronome tick which
similar with the metronome decoding.
Design doc: https://docs.google.com/document/d/18PvEgS-DehClK6twCSCATOlX-j9acmXd-3vjb0tR9-Y
Bug: b/304158952
Change-Id: I262bd4a5097fdaeed559b9d7391a059ae86e2d63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327460
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/main@{#41469}
RtpTransportControllerSend uses all 4 utilities of the environment and
thus cleaner to propagate them as single parameter instead of 4 separate
Bug: None
Change-Id: I38932c21a73ea41d4bdf2fa04bf3961a2adb25a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331821
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41422}
as a step to propagate Environment and thus field trials into Decoders
Bug: webrtc:10335
Change-Id: Ib396421f0fbf34f2c2f90aa4a1b41b461e42253c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330421
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41335}
while cleaning up Call factory function,
- pick rtp_transport_controller_send_factory based on presence in the config instead of based on the call site thus removing one extra factory function.
- when Call is created through test helper TimeControllerBasedFactory use original media factory instead of direct factory, thus allow to configure degraded call through field trials in tests, and ensure difference with production code path stay minimal in the future.
Bug: webrtc:15656
Change-Id: If9c2a9fc871e139502db2bec0a241d8d64c53720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330061
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41329}
In preparation for experimentally supporting different types of
metronomes and metronome use cases we'd like to rename for clarity.
This is the first step, which introduces the new name and prefers it if
it is set, but keeps the old name for backwards compat reasons.
Once Chromium has migrated to the new name, we can delete the old name.
Bug: webrtc:15704
Change-Id: I23077bf2415ebb2b2338320c9a14e3bd17d3abb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330020
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41319}
instead of passing utilities one by one
Bug: webrtc:15656
Change-Id: I1f3bf7ae66dcc62bbf17d81c927aabe748b42163
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328540
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41256}
PacingController per default use a burst interval of 40ms. The behaviour can still be overriden by using the method SetSendBurstInterval.
Bug: chromium:1354491
Change-Id: Ie3513109e88e9832dff47380c482ed6d943a2f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311102
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41254}
MediaType::Any will be used by packets that can not be demuxed by
RtpTransport.
Bug: webrtc:14928
Change-Id: Ib759e65c7eede29defdad8073fd1ed6be814ab81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299280
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39710}
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.
Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
Today, behaviour is decided based on if transport sequence number v2 is
in the SDP answer. But it might be better to decide based on received
packets since it is valid to negotiate both extensions.
Another bonus With this solution is that Call does not need to know
about receive header exensions.
This is an alternative to https://webrtc-review.googlesource.com/c/src/+/291337
Bug: webrtc:7135
Change-Id: Ib75474127d6e2e2029557b8bb2528eaac66979f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291525
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39226}
This reverts commit 897ea04db5.
Reason for revert: Speculative revert as it could be the reason why perf tests started failing: https://ci.chromium.org/p/webrtc/g/perf/console?limit=200
Original change's description:
> Delete PacketReceiver::DeliverPacket from all implementations
>
> And fix tests that still depend on extensions to be known by the receiver.
>
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
>
> Bug: webrtc:7135,webrtc:14795
> Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39184}
Bug: webrtc:7135,webrtc:14795,b/266658815
Change-Id: I9d03f4952938d176ffee110a707acadc1846457c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291400
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39189}
And fix tests that still depend on extensions to be known by the receiver.
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Bug: webrtc:7135,webrtc:14795
Change-Id: I62227829af81af07769189e547f1cdb8ed4d06b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290996
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39184}
DeliverRtpPacket use a parsed RTP packet as argument where the RTP extensions are supposed to be known.
This method is implemented in webrt::Call and temporary used by the exising method Call::DeliverRtp, but the idea is to instead avoid extra packet parsing by forwarding a RtpPacketReceived from RtpTransport::DemuxRtpPacket via WebrtcVideoChannel::OnPacketReceived and WebrtcVoiceChannel.
DeliverRtcpPacket is also implemented in Call and is directly used in PeerConnection::InitializeRtcpCallback.
Bug: webrtc:14795, webrtc:7135
Change-Id: Ib6ffe8e1229ac07fa459ee2fc9a0af8455a23bac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290401
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39015}
In order for a packet to be parseable and include a transport sequence number, it has to be negotiated. Thus, there is no need to check again.
Bug: webrtc:14795
Change-Id: I1fa76abdbad11d15ecae80fbaa227bd12a8035bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290565
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39011}
This is a reland of commit 97ba853295
This cl did not cause the regression in Chrome rolls https://chromium-review.googlesource.com/c/chromium/src/+/4132644?tab=checks. Real culprit reverted in https://webrtc-review.googlesource.com/c/src/+/290502.
Original change's description:
> Remove use of ReceiveStreamRtpConfig:transport_cc
>
> With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
> http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
> I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
>
>
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
>
> Bug: webrtc:14802
> Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38980}
Bug: webrtc:14802
Change-Id: Ib98e61413161d462da60144942cdb0140e12bc42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290503
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38997}
With this change, webrtc will send RTCP transport feedback for all received packets that have a transport sequence number, if the header extension
http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions is negotiated.
I.e the SDP attribute a=rtcp-fb:96 transport-cc is ignored.
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Bug: webrtc:14802
Change-Id: I95d8d4405dc86a2f872f7883b7bafd623d5f7841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290403
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38980}
Instead of getting header extension mapping from a receiver object, get the mapping from the received packet.
The purpose is to be able to remove extension information from webrtc/call/receive_stream.h.
Header extensions are negotiated per mid, not per receive stream.
The goal is to reduce the number of places where packets are parsed and demuxed.
Bug: webrtc:7135, webrtc:14795
Change-Id: I8944bc06a11dc572d9e14e7d7ee446a841096295
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288968
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38944}
This CL finalizes the Metronome refactor undertaken in
crbug.com/1381982 and enables it again in call.cc.
Fixed: chromium:1381982
Change-Id: I1642103e9c8a3f2a1f12d7635a1b27310802c1c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282920
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38605}
The Chromium implementation unfortunately has a rare deadlock.
Rather than patching that up, we're changing the metronome
implementation to be able to use a single-threaded environment
instead.
The metronome functionality is disabled in VideoReceiveStream2
construction inside call.cc.
The new design does not have listener registration or
deresigstration and instead accepts and invokes callbacks, on
the same sequence that requested the callback. This allows
the clients to use features such as WeakPtrFactories or
ScopedThreadSafety for cancellation.
The CL will be followed up with cleanup CLs that removes
registration APIs once downstream consumers have adapted.
Bug: chromium:1381982
Change-Id: I43732d1971e2276c39b431a04365cd2fc3c55c25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282280
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38582}
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}
Resolves an issue where, in Chrome, WebRTC event logs do not capture outgoing packets for video receivers because no reference to the event log was passed to the video receiver.
Bug: webrtc:14338
Change-Id: Ia33ce6f2d69a0e341530648b10a08516dc53abf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271080
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37746}
instead of using Lock/Unlock attributes, use Assert attribute to annotate code is running on certain task queue or thread.
Such check better matches what is checked, in particular allows to
recheck (and thus better document) currently used task queue
Bug: None
Change-Id: I5bc1c397efbc8342cf7915093b578bb015c85651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269381
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37619}
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}
