e9eca8f5ae60116a2351634575faf4cd5e338e61

We have decided not to do a switch from old (AudioCodingModule) to new (AudioCoding) API. Instead, we will gradually evolve the old API to meet the new design goals. As a consequence of this decision, the AudioCoding and AudioCodingImpl classes are deleted. Also removing associated unit test sources. No test coverage is lost with this operation, since the tests for the "old" API are testing more than the deleted tests did. BUG=webrtc:3520 Review URL: https://codereview.webrtc.org/1415163002 Cr-Commit-Position: refs/heads/master@{#10406}
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
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
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