Taylor Brandstetter a1c303535f Relanding: Implement RTCConfiguration.iceCandidatePoolSize.
Depends on this CL in order to work in Chromium:
https://codereview.chromium.org/1976673002/

It works by creating pooled PortAllocatorSessions which can be picked up
by a P2PTransportChannel when needed (after a local description is set).

This can optimize candidate gathering time when there is some time between
creating a PeerConnection and setting a local description.

R=pthatcher@webrtc.org

Committed: 48e9d05f51

Review URL: https://codereview.webrtc.org/1956453003 .

Cr-Commit-Position: refs/heads/master@{#12729}
2016-05-13 15:15:20 +00:00
2014-06-17 08:54:03 +00:00
2015-09-11 09:04:09 +00:00
2016-04-16 17:24:53 +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. This page is maintained by the Google Chrome team.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

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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