commit | 35151f35ecb52a3a06825a946d8ee9f21bf7bc3e | [log] [tgz] |
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author | asapersson <asapersson@webrtc.org> | Mon May 02 23:44:01 2016 -0700 |
committer | Commit bot <commit-bot@chromium.org> | Tue May 03 06:44:11 2016 +0000 |
tree | 382334bf436921b28d15b8539dbc456ad4c8da5d | |
parent | 5a2463796e3af75eec36b953878f4ce536240af3 [diff] |
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket. - "WebRTC.Video.SendDelayInMs" Change so that PacketOption packet id is always set in RtpSender (if having a TransportSequenceNumberAllocator). Add SendDelayStats class for computing delays. Add SendPacketObserver to RtpRtcp config and register SendDelayStats as observer. Wire up OnSentPacket to SendDelayStats. BUG=webrtc:5215 Review-Url: https://codereview.webrtc.org/1478253002 Cr-Commit-Position: refs/heads/master@{#12600}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.