Bjorn Terelius 07b35bcd55 Remove RtcEventLogEncoder::Encode method.
Use EncodeBatch method in unittest. (Same as in production code.)

Bug: webrtc:8111
Change-Id: Ia194f5138f244da7f348821277f6c712a3ffab0d
Reviewed-on: https://webrtc-review.googlesource.com/34560
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21696}
2018-01-19 16:03:12 +00:00
2018-01-19 10:37:44 +00:00
2018-01-18 08:03:27 +00:00
2018-01-19 10:37:44 +00:00
2018-01-18 14:24:48 +00:00
.gn
2018-01-18 16:55:58 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +00:00
2017-09-15 04:25:06 +00:00
2018-01-12 11:31:52 +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.

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%