hbos 6769c49418 RTC[In/Out]boundRTPStreamStats: qpSum,framesDecoded,framesEncoded added.
Recently added to the spec:
RTCRTPStreamStats.qpSum - https://w3c.github.io/webrtc-stats/#dom-rtcrtpstreamstats-qpsum
RTCInboundRTPStreamStats.framesDecoded - https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framesdecoded
RTCOutboundRTPStreamStats.framesEncoded - https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-framesencoded

These are added and collected. However, the qpSum is only collected in
the outbound case. It should be collected in the inbound case before
closing crbug.com/657855

BUG=chromium:657854, chromium:657855, chromium:657856

Review-Url: https://codereview.webrtc.org/2588373005
Cr-Commit-Position: refs/heads/master@{#15872}
2017-01-02 16:35:13 +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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