zhihuang 0426222f4c Modified the rtp_receiver_unittests.
Implemented operator == in RtpSource and use the gmock EXPECT_THAT to make the test cleaner.

Related CL: https://codereview.webrtc.org/2770233003/

BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2813753002
Cr-Commit-Position: refs/heads/master@{#17659}
2017-04-11 18:28:10 +00:00
.gn
2017-04-11 09:52:49 +00:00
2017-03-09 10:04:57 +00:00
2017-01-20 20:45:07 +00:00
2017-04-11 14:40:26 +00:00
2017-03-23 10:46:00 +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.

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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