Danil Chapovalov eb0ed28b90 Add rtcp::CommonHeader::packet_size function
as an alternative to NextPacket function
to allow cleaner iterating over stacked rtcp packets.

Bug: webrtc:5565
Change-Id: I261afe2684e5fcb5fa3e7bcce272fbefeebd0b66
Reviewed-on: https://webrtc-review.googlesource.com/21360
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20615}
2017-11-09 10:50:15 +00:00
2017-10-31 21:08:43 +00:00
2017-11-07 18:09:45 +00:00
.gn
2017-09-25 15:34:41 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-10-31 17:46:42 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +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.

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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