Alex Narest bbbe4e1a15 Better handle target audio bitrate allocation.
Do not update target audio bitrate if WebRTC-Audio-SendSideBwe-For-Video is enabled but other side does not support TWCC

Bug: webrtc:8243
Change-Id: I6c3c4f223dc5168d726996324717d7ba9ec96e6c
Reviewed-on: https://webrtc-review.googlesource.com/88440
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23963}
2018-07-13 09:06:06 +00:00
2018-07-13 08:39:41 +00:00
2018-07-13 08:42:11 +00:00
.gn
2018-06-29 09:36:17 +00:00
2018-06-20 12:39:11 +00:00
2018-06-26 13:57:35 +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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