Harald Alvestrand 97716c0132 Implement max-channels for SCTP datachannels.
This involves catching another callback from usrsctp.
It also moves the definition of "connected" a little later
in the sequence: From "ready to send data" to the reception
of the SCTP_COMM_UP event.

Bug: chromium:943976
Change-Id: Ib9e1b17d0cc356f19cdfa675159b29bf1efdcb55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137435
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28004}
2019-05-21 10:24:41 +00:00
2018-10-05 14:40:21 +00:00
2019-05-13 10:41:40 +00:00
2019-04-29 12:55:02 +00:00
2019-05-21 09:41:41 +00:00
.gn
2019-03-07 13:08:17 +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
2019-04-26 12:45:06 +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
Readme BSD-3-Clause 446 MiB
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