Reason for revert: Intending to fix issues and reland. Original issue's description: > Revert of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #9 id:160001 of https://codereview.webrtc.org/2659563002/ ) > > 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} > Committed:14245cc939TBR=stefan@webrtc.org,brandtr@webrtc.org BUG=webrtc:6847 Review-Url: https://codereview.webrtc.org/2673523003 Cr-Commit-Position: refs/heads/master@{#16440}
Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ )
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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