Sergey Silkin 2f864fb4ab Handle empty GOF.
Assume that stream has single temporal layer if number of frames in GOF
is set to zero (valid case).

Bug: chromium:879584
Change-Id: I7ced082190e40c1bf4cc1468babfd98b0a61f0dd
Reviewed-on: https://webrtc-review.googlesource.com/98800
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24622}
2018-09-07 10:19:06 +00:00
2018-09-01 09:20:03 +00:00
2018-09-06 23:29:15 +00:00
2018-09-07 10:19:06 +00:00
2018-09-06 12:53:19 +00:00
2018-09-06 12:53:19 +00:00
2018-09-06 12:53:19 +00:00
2018-09-05 09:30:08 +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
2018-09-04 11:14:07 +00:00
2018-07-23 15:28:48 +00:00
2017-09-15 04:25:06 +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
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%