Danil Chapovalov 6026f05ef1 Calculate max payload size for an rtp packet to fit full video frame
instead of sometimes incorrectly guessing it

Bug: webrtc:9868
Change-Id: I8b15ecca4c660d83ea129dc9df6ec174ad83b4c6
Reviewed-on: https://webrtc-review.googlesource.com/c/106281
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25213}
2018-10-16 15:32:37 +00:00
2018-10-05 14:40:21 +00:00
2018-10-15 06:59:19 +00:00
.gn
2018-08-13 13:54:05 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2018-07-23 15:28:48 +00:00
2018-07-23 15:28:48 +00:00
2017-09-15 04:25:06 +00:00
2018-10-11 11:52:28 +00:00
2018-02-23 10:34:16 +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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