hbos 82ebe02491 Correct stats for RTCPeerConnectionStats.dataChannels[Opened/Closed].
DataChannel.SignalOpened and unittests added.
PeerConnection.SignalDataChannelCreated added and wired up to
RTCStatsCollector.OnDataChannelCreated on RTCStatsCollector
construction.
RTCStatsCollector.OnSignalOpened/Closed added and wired up on
OnDataChannelCreated.
rtcstatscollector_unittest.cc updated, faking that channels are opened
and closed.

I did not want to use DataChannelObserver because it is used for more
than state changes and there can only be one observer (unless code is
updated). Since DataChannel already had a SignalClosed it made sense to
add a SignalOpened.

Having OnSignalBlah in RTCStatsCollector is new in this CL but will
likely be needed to correctly handle RTPMediaStreamTracks being added
and detached independently of getStats. This CL establishes this
pattern.

(An integration test will be needed for this and all the other stats to
make sure everything is wired up correctly and test outside of a
mock/fake environment, but this is not news.)

BUG=chromium:636818, chromium:627816

Review-Url: https://codereview.webrtc.org/2472113002
Cr-Commit-Position: refs/heads/master@{#15059}
2016-11-14 09:41:56 +00:00
2016-06-14 09:39:40 +00:00
2015-09-11 09:04:09 +00:00
2016-10-19 11:52:19 +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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