Ownership of the retransmission rate limiter for video is moved
from send side congestion controller to Call. This is to reduce the
interface on the rtp transport controller send.
Bug: webrtc:8415
Change-Id: Ie9c7317400a9eb61a3c8325b9e527844ffc13769
Reviewed-on: https://webrtc-review.googlesource.com/58745
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22254}
DeRegisterNetworkObserver is not used, since
RtpTransportControllerSend owns the thread on which
SendSideCongestionController runs it would not be safe to allow it to be
called from outside. Deregistration should be done by destroying
RtpTransportControllerSend.
In the future, the RegisterObserver functions should be removed as well,
in favor of providing the observer in the constructor. This requires
breaking a circular dependency between RtpTransportControllerSend and
Call.
Bug: webrtc:8415
Change-Id: Ifeb4c5d4a41e4d8419994b3146980bdaaf9cd6a9
Reviewed-on: https://webrtc-review.googlesource.com/58098
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22249}
SetTransportOverhead was used by send streams to signal the packet
overhead that they received from Call. However, call receives the value
from OnNetworkRouteChanged in WebRtcVideoChannel and
WebRtcVoiceMediaChannel which is already propagated to
RtpTransportControllerSend. By skipping the round trip, the interface on
the rtp transport controller can be reduced.
Bug: None
Change-Id: I759b1207aab214bbc2b993106f6ff7cc24e177f7
Reviewed-on: https://webrtc-review.googlesource.com/57182
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22226}
Moving the module process thread responsible for running the pacer
and the send side congestion controller to RtpTransportControllerSend
since it already owns the pacer and the congestion controller. They
are also moved to a common thread rather than using two separate
threads.
As part of the move, the remote bitrate estimator has been moved to the
common process thread in the Call class. Previously it was run on the
removed pacer thread.
Bug: webrtc:8415
Change-Id: I4322eef30d8b97b9611f33af7e560703b710d232
Reviewed-on: https://webrtc-review.googlesource.com/55700
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22166}
Since rtp transport controller send owns the congestion controller it
also should own the bitrate configuration logic, this way it can
initialize the send side congestion controller with the bitrate
configuration.
Bug: webrtc:8415
Change-Id: Ifaa16139ca477cb1c80bf4aa24f17652af997553
Reviewed-on: https://webrtc-review.googlesource.com/54303
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22127}
This CL removes direct access to SendSideCongestionController (SSCC) via
the RtpTransportControllerSend interface and replaces all usages with
calls on RtpTransportControllerSend which will in turn calls SSCC. This
prepares for later refactor of RtpTransportControllerSend.
Bug: webrtc:8415
Change-Id: I68363a3ab0203b95579f747402a1e7f58a5eeeb5
Reviewed-on: https://webrtc-review.googlesource.com/53860
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22044}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
