hbos 84abeb1d37 RTC[In/Out]boundRTPStreamStats.mediaTrackId collected.
Based on the mapping between [Audio/Video]TrackInterface and
[Voice/Video][Sender/Receiver]Info.

The IDs of RTCMediaStreamTrackStats are updated to distinguish between
local and remote cases. Previously, if local and remote cases had the
same label only one of them would be included in the report (bug).

BUG=webrtc:6758, chromium:657854, chromium:657855, chromium:657856, chromium:627816

Review-Url: https://codereview.webrtc.org/2610843003
Cr-Commit-Position: refs/heads/master@{#16095}
2017-01-16 14:16:44 +00:00
2016-06-14 09:39:40 +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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