| commit | 9e336ec0b8a77c3461d13677cff3563c11c88daa | [log] [tgz] |
|---|---|---|
| author | Bjorn Terelius <terelius@webrtc.org> | Tue Apr 24 16:28:35 2018 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Wed Apr 25 09:37:03 2018 +0000 |
| tree | a2fecac8c2a7cbd748295ef2e82052d4c476b99d | |
| parent | a945aee72e4e9ab3c753a60c3f4fe31b55db005f [diff] |
Create new API for RtcEventLogParser. The new API stores events gathered by event type. For example, it is possible to ask fo a list of all incoming RTCP messages or all audio playout events. The new API is experimental and may change over next few weeks. Once it has stabilized and all unit tests and existing tools have been ported to the new API, the old one will be removed. This CL also updates the event_log_visualizer tool to use the new parser API. This is not a funcional change except for: - Incoming and outgoing audio level are now drawn in two separate plots. - Incoming and outgoing timstamps are now drawn in two separate plots. - RTCP count is no longer split into Video and Audio. It also counts all RTCP packets rather than only specific message types. - Slight timing difference in sendside BWE simulation due to only iterating over transport feedbacks and not over all RTCP packets. This timing changes are not visible in the plots. Media type for RTCP messages might not be identified correctly by rtc_event_log2text anymore. On the other hand, assigning a specific media type to an RTCP packet was a bit hacky to begin with. Bug: webrtc:8111 Change-Id: I8e7168302beb69b2e163a097a2a142b86dd4a26b Reviewed-on: https://webrtc-review.googlesource.com/60865 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23015}
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.
Authoritative list of directories that contain the native API header files.