| commit | 77cd58e140fb8046a7f918c88530ccb6f380bf11 | [log] [tgz] |
|---|---|---|
| author | perkj <perkj@webrtc.org> | Tue May 30 10:52:10 2017 |
| committer | Commit Bot <commit-bot@chromium.org> | Tue May 30 10:52:10 2017 |
| tree | 93642f05ba3b5fda387c69c372289d2916009c37 | |
| parent | 367aba92bf14bd1101fa61cb5766c766fde86152 [diff] |
This cl removes RtcEventLog deps to call:call_interfaces. The purpose is to be able to use the event log from the upcoming RtpTransport. Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader. Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead. BUG=webrtc:7538 TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc Review-Url: https://codereview.webrtc.org/2855143002 Cr-Commit-Position: refs/heads/master@{#18324}
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.