14245cc939
Reason for revert:
This change causes excessive logging when running tests, and possibly also broke perf tests, see https://build.chromium.org/p/client.webrtc.perf/builders/Linux%20Trusty/builds/1040/steps/webrtc_perf_tests/logs/stdio
Original issue's description:
> Always call RemoteBitrateEstimator::IncomingPacket from Call.
>
> Delete the calls from RtpStreamReceiver (for video) and
> AudioReceiveStream.
>
> BUG=webrtc:6847
>
> Review-Url: https://codereview.webrtc.org/2659563002
> Cr-Commit-Position: refs/heads/master@{#16393}
> Committed: 6d4dd593a8
TBR=stefan@webrtc.org,brandtr@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2668973003
Cr-Commit-Position: refs/heads/master@{#16400}
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. This page is maintained by the Google Chrome team.
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