Bjorn A Mellem 487f9a17e4 Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
Also clears SctpTransport before deleting JsepTransport.

SctpTransport is ref-counted, but the underlying transport is deleted when
JsepTransport clears the rtp_dtls_transport.  This results in crashes when
usrsctp attempts to send outgoing packets through a dangling pointer to the
underlying transport.

Clearing SctpTransport before DtlsTransport removes the pointer to the
underlying transport before it becomes invalid.

This fixes a crash in chromium's web platform tests (see
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: I64e94b88afb119fdbf5f22750f88c8a084d53937
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/151981
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29120}
2019-09-09 21:58:36 +00:00
2019-08-20 14:00:49 +00:00
2018-10-05 14:40:21 +00:00
2019-08-28 14:57:28 +00:00
2019-07-08 13:45:15 +00:00
2018-12-18 12:30:58 +00:00
2019-05-17 18:11:58 +00:00
2018-07-23 15:28:48 +00:00
2018-07-23 15:28:48 +00:00
2019-09-03 14:55:43 +00:00
2019-08-20 14:00:49 +00:00

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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