The point of this change is to make it possible to create the congestion
controller as part of creating RtpTransportController, later pass it to the
constructor of Call, and then let Call register itself as an observer.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2795643002
Cr-Commit-Position: refs/heads/master@{#17504}
(The two functions are always called together anyway.)
BUG=None
Review-Url: https://codereview.webrtc.org/2784333005
Cr-Commit-Position: refs/heads/master@{#17502}
Reason for revert:
Looks like this has caused multiple Android webrtc perf build bot failures in RampUpTest.UpDownUpTransportSequenceNumberRtx
Original issue's description:
> Enable the bayesian bitrate estimator by default.
>
> BUG=webrtc:6566, webrtc:7415
>
> Review-Url: https://codereview.webrtc.org/2749803002
> Cr-Commit-Position: refs/heads/master@{#17475}
> Committed: c53a17f28eTBR=terelius@webrtc.org,magjed@webrtc.org,stefan@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6566, webrtc:7415
Review-Url: https://codereview.webrtc.org/2786913003
Cr-Commit-Position: refs/heads/master@{#17476}
Implementation owned by call, and passed to VideoSendStream and
AudioSendStream.
BUG=webrtc:6847, webrtc:7135
Review-Url: https://codereview.webrtc.org/2685673003
Cr-Commit-Position: refs/heads/master@{#17389}
This CL is one in a series. To finish the work, the following CLs will be added:
1. CL for connecting RPLR as well
2. CL for RPLR-based FecController
3. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2638083002
Cr-Commit-Position: refs/heads/master@{#17365}
Reason for revert:
Problem was the rename of the include file. Intend to keep the old name for now, and then reland.
Original issue's description:
> Revert of Delete class MockCongestionController. (patchset #4 id:60001 of https://codereview.webrtc.org/2762023004/ )
>
> Reason for revert:
> This is breaking downstream build.
>
> Original issue's description:
> > Delete class MockCongestionController.
> >
> > It became unused in cl https://codereview.webrtc.org/2516983004.
> >
> > BUG=None
> >
> > Review-Url: https://codereview.webrtc.org/2762023004
> > Cr-Commit-Position: refs/heads/master@{#17325}
> > Committed: d19bcb7116
>
> TBR=stefan@webrtc.org,nisse@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2762133003
> Cr-Commit-Position: refs/heads/master@{#17330}
> Committed: e27f1e764eTBR=stefan@webrtc.org,skvlad@webrtc.org
BUG=None
Review-Url: https://codereview.webrtc.org/2766133002
Cr-Commit-Position: refs/heads/master@{#17338}
New class ReceiveSideCongestionController, extracted from CongestionController, and responsible for the
OnReceivedPacket processing.
Rest of the CongestionController moved to a new class
SendSideCongestionController.
To avoid breaking applications, CongestionController is redefined
as a union of these two classes, with no intended change in behavior.
With one exception: CongestionController::SetBweBitrates used to call
remote_bitrate_estimator_.SetMinBitrate, but after remote_bitrate_estimator_ was moved to ReceiveSideCongestionController,
it no longer does this.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2752233002
Cr-Commit-Position: refs/heads/master@{#17321}
1. GetTransportFeedbackVector will now return a vector which also explicitly states lost packets.
2. The returned vector is unsorted (uses default order - by sequence number). It's up to the users to sort otherwise, if they need a different order.
BUG=None
Review-Url: https://codereview.webrtc.org/2707383006
Cr-Commit-Position: refs/heads/master@{#17114}
It's the faster, less strict cousin of checked_cast.
BUG=none
Review-Url: https://codereview.webrtc.org/2714063002
Cr-Commit-Position: refs/heads/master@{#16958}
In order to not make this CL too large I have broken it down into at least two
steps. Previous CL: https://codereview.chromium.org/2628563003/
webrtc::PacedSender::Process <--- previous CL start here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket <--- previous CL end here, this Cl start here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld <--- this CL end here
BUG=webrtc:6822
Review-Url: https://codereview.webrtc.org/2708873003
Cr-Commit-Position: refs/heads/master@{#16796}
Rename loss based and delay based bwe updates in proto (and correspondingly in the C++ code).
BUG=webrtc:6423
Review-Url: https://codereview.webrtc.org/2705613002
Cr-Commit-Position: refs/heads/master@{#16719}
In order to not make this CL too large I have broken it down into at least two steps. In this CL we only propagate the pacing information part of the way:
webrtc::PacedSender::Process <--- propagate from here
webrtc::PacedSender::SendPacket
webrtc::PacketRouter::TimeToSendPacket
webrtc::ModuleRtpRtcpImpl::TimeToSendPacket <--- to here
webrtc::RTPSender::TimeToSendPacket
webrtc::RTPSender::PrepareAndSendPacket
webrtc::RTPSender::AddPacketToTransportFeedback
webrtc::TransportFeedbackAdapter::AddPacket
webrtc::SendTimeHistory::AddAndRemoveOld <--- goal is to propagte it here
BUG=webrtc:6822
Review-Url: https://codereview.webrtc.org/2628563003
Cr-Commit-Position: refs/heads/master@{#16664}
Lateness is determined by the length of the send-side history, currently
set to 60 seconds.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2684353004
Cr-Commit-Position: refs/heads/master@{#16588}
This helps us avoid time-outs on really bad networks with long queues.
Also adding periodic logging of the fake network pipe's queue in milliseconds.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2687013005
Cr-Commit-Position: refs/heads/master@{#16532}
This avoids issues where the bitrate produced by the codec is far lower than the target bitrate in the beginning, which causes the delay-based BWE to be initialized accordingly.
BUG=webrtc:5079
Review-Url: https://codereview.webrtc.org/2653883002
Cr-Commit-Position: refs/heads/master@{#16327}
This means that smaller probe packets will be allowed at lower bitrates.
BUG=webrtc:7043
Review-Url: https://codereview.webrtc.org/2650393002
Cr-Commit-Position: refs/heads/master@{#16317}
It was only assigned at construction, and this improves consistency
with remote_estimator_proxy_.
The declaration of the private WrappingBitrateEstimator had to be
moved to the header file, and it was also converted from
system_wrappers' CriticalSectionWrapper to rtc::CriticalSection.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2642363003
Cr-Commit-Position: refs/heads/master@{#16236}
Reason for revert:
Speculative revert for perf regression related to ramp-up on android. See https://bugs.chromium.org/p/chromium/issues/detail?id=682611
Original issue's description:
> Move congestion controller processing to the pacer thread.
>
> Also rename it from pacer_thread_ to congestion_controller_thread_.
>
> BUG=webrtc:6847
>
> Review-Url: https://codereview.webrtc.org/2637783003
> Cr-Commit-Position: refs/heads/master@{#16134}
> Committed: b3dc2b7b1eTBR=danilchap@webrtc.org,stefan@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2644603003
Cr-Commit-Position: refs/heads/master@{#16163}
