Steve Anton 53c7ba632d Add BUNDLE processing to RtpDemuxer.
Extends the RtpDemuxer to do demuxing according to the BUNDLE spec,
using MID and payload types in addition to RSID and SSRC. Also extends
SsrcBindingObserver to receive notification for all types of SSRC
binding that can occur with the new algorithm.

Bug: webrtc:4050
Change-Id: Ie2f347f90d5074ab537fa1162fa7314dd292b68b
Reviewed-on: https://chromium-review.googlesource.com/578628
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19415}
2017-08-18 17:33:08 +00:00
2017-06-30 10:04:59 +00:00
.gn
2017-08-02 08:26:18 +00:00
2017-06-30 10:04:59 +00:00
2017-01-20 20:45:07 +00:00
2017-08-16 06:40:57 +00:00
2017-03-23 10:46:00 +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.

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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