Henrik Boström 8b14b0dea6 Revert "Refactor SCTP data channels to use DataChannelTransportInterface."
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}
2019-08-30 12:31:21 +00:00
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2018-07-23 15:28:48 +00:00
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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

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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