This reverts commit 4c85828ab272d9bd58789bad7b135b6287395f97. Reason for revert: Speculatively reverting this because it makes several web platform tests relating to RTCDataChannel flaky, see first failing roll: https://chromium-review.googlesource.com/c/chromium/src/+/1776711 Original change's description: > Refactor SCTP data channels to use DataChannelTransportInterface. > > This change moves SctpTransport to be owned by JsepTransport, which now > holds a DataChannelTransport implementation for SCTP when it is used for > data channels. > > This simplifies negotiation and fallback to SCTP. Negotiation can now > use a composite DataChannelTransport, just as negotiation for RTP uses a > composite RTP transport. > > PeerConnection also has one fewer way it needs to manage data channels. > It now handles SCTP and datagram- or media-transport-based data channels > the same way. > > There are a few leaky abstractions left. For example, PeerConnection > calls Start() on the SctpTransport at a particular point in negotiation, > but does not need to call this for other transports. Similarly, PC > exposes an interface to the SCTP transport directly to the user; there > is no equivalent for other transports. > > Bug: webrtc:9719 > Change-Id: I0d3151c48c1a511368277981fc4cf818a9f8ebb4 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150341 > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Reviewed-by: Benjamin Wright <benwright@webrtc.org> > Commit-Queue: Bjorn Mellem <mellem@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#29012} TBR=steveanton@webrtc.org,mellem@webrtc.org,benwright@webrtc.org Change-Id: I074b9e68f298d20d0cabb4239084b4843e76e910 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9719 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150944 Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29025}
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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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