hbos a51d4f34d9 Re-land of RTCInboundRTPStreamStats.qpSum collected.
This was previously only collected for local tracks
(RTCOutboundRTPStreamStats.qpSum).

Spec: https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum

This CL also improves some testing in rtcstatscollector_unittest.cc.
Default and non-default values are tested in the same unittests,
removing the test that was specific to default-values, which was
otherwise code duplication.

This is a re-land of https://codereview.webrtc.org/2675943002 after
dependent CL that was re-landed.

BUG=webrtc:7065
TBR=hta@webrtc.org, sakal@webrtc.org

Review-Url: https://codereview.webrtc.org/2703503003
Cr-Commit-Position: refs/heads/master@{#16642}
2017-02-16 13:34:48 +00:00
2017-02-03 14:37:05 +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

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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