3c39c0137a
Reason for revert:
A few perf tests broken, including
RampUpTest.UpDownUpAbsSendTimeSimulcastRedRtx
RampUpTest.UpDownUpTransportSequenceNumberRtx
RampUpTest.UpDownUpTransportSequenceNumberPacketLoss
Original issue's description:
> Use RtxReceiveStream.
>
> This also has the beneficial side-effect that when a media stream
> which is protected by FlexFEC receives an RTX retransmission, the
> retransmitted media packet is passed into the FlexFEC machinery,
> which should improve its ability to recover packets via FEC.
>
> BUG=webrtc:7135
>
> Review-Url: https://codereview.webrtc.org/3008773002
> Cr-Commit-Position: refs/heads/master@{#19649}
> Committed: 5c0f6c62ea
TBR=brandtr@webrtc.org,danilchap@webrtc.org,stefan@webrtc.org,magjed@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/3010983002
Cr-Commit-Position: refs/heads/master@{#19653}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct