e680c83a41
This reverts commit 8848229234aae01ec19582ece7b748d557119d66. Reason for revert: break chromium compilation on iOS https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8915214519549611184/+/steps/compile/0/stdout Original change's description: > Add Video Bwe stats collection to DefaultVideoQualityAnalyzer. > > This CL adds the possibility to collect the following Video BWE stats: > - available_send_bandwidth > - transmission_bitrate > - retransmission_bitrate > - actual_encode_bitrate > - target_encode_bitrate > > Example of the output: > > RESULT available_send_bandwidth: smoke_test/alice= {487754.33,87583.093} bytesPerSecond > RESULT transmission_bitrate: smoke_test/alice= {465779.17,212075.5} bytesPerSecond > RESULT retransmission_bitrate: smoke_test/alice= {20036,26326.751} bytesPerSecond > RESULT actual_encode_bitrate: smoke_test/alice= {418779.33,200486.03} bytesPerSecond > RESULT target_encode_bitrate: smoke_test/alice= {469491.17,77866.909} bytesPerSecond > RESULT available_send_bandwidth: smoke_test/bob= {642924.83,168842.34} bytesPerSecond > RESULT transmission_bitrate: smoke_test/bob= {626115.5,294783.56} bytesPerSecond > RESULT retransmission_bitrate: smoke_test/bob= {0,0} bytesPerSecond > RESULT actual_encode_bitrate: smoke_test/bob= {594235.33,297289.54} bytesPerSecond > RESULT target_encode_bitrate: smoke_test/bob= {640463.5,167676.66} bytesPerSecond > > Bug: webrtc:10138 > Change-Id: I0414055af0010b8fb4d909297e6da86d398157c2 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132703 > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Tommi <tommi@webrtc.org> > Reviewed-by: Artem Titov <titovartem@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Mirko Bonadei <mbonadei@google.com> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#27760} TBR=mbonadei@webrtc.org,mbonadei@google.com,ilnik@webrtc.org,tommi@webrtc.org,titovartem@webrtc.org Change-Id: Ib0ef94331410d9b22b6425e4da412b75360fa2d9 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10138 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134210 Reviewed-by: Artem Titov <titovartem@webrtc.org> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27771}
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.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- 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