commit | fc5e81c979183df3ae5e451838a19e2228205c0d | [log] [tgz] |
---|---|---|
author | asapersson <asapersson@webrtc.org> | Thu Apr 20 06:28:53 2017 |
committer | Commit bot <commit-bot@chromium.org> | Thu Apr 20 06:28:53 2017 |
tree | 59fa62c14cc6d3b7fd1095bc8680350752226023 | |
parent | c2a18c2aae3e199e9ca1e19fe0a8999f6d207f0e [diff] |
Replace first_packet_sent_ms_ in Call. Instead of using the time on the first callback to Call::OnSentPacket, use the time when the first packet is sent from the pacer (to make sure this packet corresponds to an audio/video RTP packet). BUG=webrtc:6244 Review-Url: https://codereview.webrtc.org/2825333002 Cr-Commit-Position: refs/heads/master@{#17777}
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.