zhihuang cf5b37cc46 Accept all the media profiles required by JSEP.
JSEP section 5.1.3 states that:
  Any profile matching the following patterns MUST be accepted:
  "RTP/[S]AVP[F]" and "(UDP/TCP)/TLS/RTP/SAVP[F]"

NOTRY=True
BUG=webrtc:5638

Committed: https://crrev.com/b7f425ab68ec58e2a5beaaf5ef79f50f1982c6f9
Cr-Commit-Position: refs/heads/master@{#12338}

Review-Url: https://codereview.webrtc.org/1880913002
Cr-Commit-Position: refs/heads/master@{#12637}
2016-05-05 18:44:44 +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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