Sebastian Jansson e6c0964572 Ensures that arrival is past send time in SimulatedNetwork.
Bug: webrtc:8415
Change-Id: I2797c7dfb3e7b9622a12c2d1e35462e0c686fa8e
Reviewed-on: https://webrtc-review.googlesource.com/76101
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23228}
2018-05-15 07:18:00 +00:00
2018-05-14 06:57:38 +00:00
2018-05-14 06:57:38 +00:00
2018-04-11 06:45:07 +00:00
2018-05-15 07:01:20 +00:00
2018-05-14 06:57:38 +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
2018-05-15 07:01:20 +00:00
2018-04-19 06:52:18 +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%