qiangchen 067121ab3f Bug Fix: WebRTC Receiver Timestamp Jump Detection
RTCVideoEncoder does not propagate RTP timestamps properly for encoded video frames, and as such whenever switching between simulcast layers there's a large timestamp gap that causes the incoming stream to freeze (timestamps look like they're either too far ahead or too far behind the previous frame).

Ideally RTCVideoEncoder would propagate these timestamps, but even so, when there's a large timestamp gap it would seem reasonable that the receiver resets quickly and consider this to be a new stream.

This CL detects the large jump for timestamps, if that happens, we reset the time extrapolator, which is the class for convertion from RTP timestamp to clock time.

BUG=chromium:705679

Review-Url: https://codereview.webrtc.org/2776813002
Cr-Commit-Position: refs/heads/master@{#17770}
2017-04-19 16:57:37 +00:00
.gn
2017-04-19 15:37:36 +00:00
2017-01-20 20:45:07 +00:00
2017-03-23 10:46:00 +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.

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%