Bjorn Terelius 28db266c9b Add simulation of receive-side bandwidth estimate to event_log_analyzer.
Previously reviewed at https://codereview.webrtc.org/2986683002/

Bug: webrtc:7726
Change-Id: I9568bd8387d79f313d6c7d53ded7c23460df1598
Reviewed-on: https://webrtc-review.googlesource.com/6360
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20141}
2017-10-04 13:11:54 +00:00
2017-10-03 16:02:03 +00:00
2017-10-04 12:49:54 +00:00
.gn
2017-09-25 15:34:41 +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
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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.

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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