terelius 4c9b4af53a Compute packet loss for event log visualization similar to how it is defined in RFC 3550.
The main difference to the old computation is that the expected number of packets during an interval is now computed as the change in highest sequence number encountered, rather than the sequence number difference between the first and last packet in the interval.

BUG=webrtc:7046

Review-Url: https://codereview.webrtc.org/2656333002
Cr-Commit-Position: refs/heads/master@{#16361}
2017-01-30 16:44:51 +00:00
2017-01-20 04:20:45 +00:00
2016-06-14 09:39:40 +00:00
2017-01-20 20:45:07 +00:00
2015-09-11 09:04:09 +00:00
2016-11-23 16:42:57 +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. This page is maintained by the Google Chrome team.

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
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