This reverts commit ea8b62a3e74fe91cd6bf66304839cd5677880a4e. Reason for revert: Broke chromium tests. Original change's description: > 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} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,zhihuang@webrtc.org Change-Id: If245da9d1ce970ac8dab7f45015e9b268a5dbcbd No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8587 Reviewed-on: https://webrtc-review.googlesource.com/64860 Reviewed-by: Zhi Huang <zhihuang@webrtc.org> Commit-Queue: Zhi Huang <zhihuang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22614}
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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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