| commit | 0255acb0855b0b1c5f14238b809052f85cc8c220 | [log] [tgz] |
|---|---|---|
| author | asapersson <asapersson@webrtc.org> | Tue Mar 28 09:44:58 2017 |
| committer | Commit bot <commit-bot@chromium.org> | Tue Mar 28 09:44:58 2017 |
| tree | b986d1d790e19dbebce93fc33c82361bfefd15f9 | |
| parent | b1a897680d5bb2b0a6fa8f350083bc22b38162df [diff] |
Change VideoReceiveStream::Stats total_bitrate_bps to include all received packets. The total_bitrate_bps is based on complete frames (from jitter buffer callback: ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe, size_t size_bytes)) (since M58). Receive total_bitrate_bps can be incorrectly reported. Example log of stats_.total_bitrate_bps in video_loopback call (target bitrate 600kbps): SEND: stats_.total_bitrate_bps 682832 RECV: stats_.total_bitrate_bps 1078104 (new: 677384) SEND: stats_.total_bitrate_bps 694280 RECV: stats_.total_bitrate_bps 1091768 (new: 663152) SEND: stats_.total_bitrate_bps 626248 RECV: stats_.total_bitrate_bps 7683776 (new: 636080) Changed total_bitrate_bps to be based on incoming packets (reported via callback from ReceiveStatisticsImpl: ReceiveStatisticsProxy::DataCountersUpdated(const webrtc::StreamDataCounters& counters, uint32_t ssrc)). BUG=webrtc:7400 Review-Url: https://codereview.webrtc.org/2775813002 Cr-Commit-Position: refs/heads/master@{#17411}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.