ff61273c01
This reverts commit 9e336ec0b8a77c3461d13677cff3563c11c88daa. Reason for revert: Code can accidentally include the deprecated parser but link with the new one, or vice versa. Reverting to fix naming. Original change's description: > 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} TBR=terelius@webrtc.org,srte@webrtc.org,minyue@webrtc.org Change-Id: Ib4bbcf0563423675a3cc1dce59ebf665e0c5dae9 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8111 Reviewed-on: https://webrtc-review.googlesource.com/72500 Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23026}
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.
Development
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.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct