Bjorn Terelius 18f65dc20a Don't attempt to unwrap RTP timestamps for RTX stream.
This fixes a bug where the event_log_visualizer hits a DCHECK when the RTP timestamp jumps.

TBR = kwiberg

Bug: webrtc:10170
Change-Id: I127a8e6165265d0726892a912f5bcdc33d98ced5
Reviewed-on: https://webrtc-review.googlesource.com/c/119664
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26410}
2019-01-25 15:59:22 +00:00
2019-01-25 08:31:12 +00:00
2018-10-05 14:40:21 +00:00
2018-12-05 12:20:56 +00:00
2019-01-25 13:58:57 +00:00
2019-01-17 22:38:57 +00:00
2018-08-13 13:54:05 +00:00
2019-01-18 10:55:41 +00:00
2017-09-15 04:25:06 +00:00
2018-12-18 12:30:58 +00:00
2018-11-09 14:23:59 +00:00
2017-09-15 04:25:06 +00:00
2018-07-23 15:28:48 +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
No description provided
Readme 255 MiB
Languages
C++ 88.6%
C 3.3%
Java 3%
Objective-C++ 1.9%
Python 1.9%
Other 1%