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}
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}
It combines and simplify use of GetStatusVector and GetReceiveDeltas accesors.
Replace use of all GetStatusVector/GetReceiveDeltasUs
BUG=None
Review-Url: https://codereview.webrtc.org/2633923003
Cr-Commit-Position: refs/heads/master@{#16139}
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}
