| commit | ea8b62a3e74fe91cd6bf66304839cd5677880a4e | [log] [tgz] |
|---|---|---|
| author | Zhi Huang <zhihuang@webrtc.org> | Mon Mar 26 21:31:17 2018 |
| committer | Commit Bot <commit-bot@chromium.org> | Mon Mar 26 22:40:05 2018 |
| tree | d4a6b7d1f4c086301be6745927b2d32743ef8758 | |
| parent | cf76d1c845b93d404692e9f545d1df88499ac305 [diff] |
Replace BundleFilter with RtpDemuxer in RtpTransport. BundleFilter is replaced by RtpDemuxer in RtpTransport for payload type-based demuxing. RtpTransport will support MID-based demuxing later. Each BaseChannel has its own RTP demuxing criteria and when connecting to the RtpTransport, BaseChannel will register itself as a demuxer sink. The inheritance model is changed. New inheritance chain: DtlsSrtpTransport->SrtpTransport->RtpTranpsort NOTE: When RTCP packets are received, Call::DeliverRtcp will be called for multiple times (webrtc:9035) which is an existing issue. With this CL, it will become more of a problem and should be fixed. Bug: webrtc:8587 Change-Id: I1d8a00443bd4bcbacc56e5e19b7294205cdc38f0 Reviewed-on: https://webrtc-review.googlesource.com/61360 Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22613}
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.