commit | 11d583f41484913fd1e7b3e283966eb7b7e11ed2 | [log] [tgz] |
---|---|---|
author | henrik.lundin <henrik.lundin@webrtc.org> | Fri Sep 18 08:28:05 2015 |
committer | Commit bot <commit-bot@chromium.org> | Fri Sep 18 08:28:14 2015 |
tree | f623528687f14838eac72e497aaf0386176df1d0 | |
parent | 35624c2c3686a2ad40daffe073aa78507b0ef88e [diff] |
Fix a bug in RtpFileSource related to RTCP packets in rtpdump files According to http://www.cs.columbia.edu/irt/software/rtptools/#rtpdump, RTCP packets are marked with plen==0. In this class, plen is mapped to original_length, not length. Review URL: https://codereview.webrtc.org/1356543002 Cr-Commit-Position: refs/heads/master@{#9981}
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.