Steve Anton 741164813a Remove SessionStats.proxy_to_transport
The stats collectors would only ever call this on the signaling
thread, so they might as well just ask the voice/video channel
their transport_name directly.

Bug: None
Change-Id: I8dd36210ff22d222b45b5f5e22c253f601cdc79e
Reviewed-on: https://webrtc-review.googlesource.com/34581
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21339}
2017-12-18 23:37:47 +00:00
2017-12-15 16:48:57 +00:00
2017-11-22 09:12:18 +00:00
2017-12-18 18:08:43 +00:00
2017-12-18 23:37:47 +00:00
2017-12-12 14:32:56 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +00:00
2017-09-15 04:25:06 +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.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

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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