Moving/renaming webrtc/common.h. This file defines webrtc::Config which was mostly used by modules/audio_processing. The files webrtc/common.h, webrtc/common.cc and webrtc/test/common_unittests.cc are moved to modules/audio_processing and the few remaining uses of webrtc::Config are replaced with simpler code. - For NetEq and pacing configuration, a VoEBase::ChannelConfig is passed to VoEBase::CreateChannel(). - Removes the need for VoiceEngine::Create(const Config& config). No need to store the webrtc::Config in VoE shared state. BUG=webrtc:5879 Review-Url: https://codereview.webrtc.org/2307533004 Cr-Commit-Position: refs/heads/master@{#14109}
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.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.