commit | d3f3816ad5b588f4363012a8efdbbd044c3129d4 | [log] [tgz] |
---|---|---|
author | Sebastian Jansson <srte@webrtc.org> | Wed Feb 28 15:14:44 2018 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Feb 28 18:40:10 2018 |
tree | ab50f9ad5fdf125d4629dea846d690004d1bc772 | |
parent | 98e0111ea53ba3ac6a4c217e46948e05dd21fb30 [diff] |
Testing multiple retransmission in video nack test. Modifying VideoSendStreamTest.RetransmitsNack to test for multiple lost packets. This covers more failure modes since the RateLimiter class always allows the first packet to get trough. Bug: None Change-Id: I2c408ea10ed4ac130edc55626b5ec03397ac0d9a Reviewed-on: https://webrtc-review.googlesource.com/58743 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22236}
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.
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.