commit | 04c0722cf1fc38e6888ef44e413603e0cb461ea4 | [log] [tgz] |
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author | aleloi <aleloi@webrtc.org> | Tue Nov 22 14:42:53 2016 |
committer | Commit bot <commit-bot@chromium.org> | Tue Nov 22 14:42:59 2016 |
tree | 9d9686ee1c72ab5ce7e84dd283640fb6f312c5c9 | |
parent | b42604078864a2bff98d078023122513a837a694 [diff] |
Replace AudioConferenceMixer with AudioMixer. This CL re-routes audio through AudioMixer instead of AudioConferenceMixer. This is done without any modifications to VoiceEngine. Previously, output audio was polled by an AudioDevice through an AudioTransport pointer, which was an instance of VoEBaseImpl. VoiceEngineImpl sent the request for data on to OutputMixer and further to AudioConferenceMixer. This CL changes the audio flow to an AudioDevice. We reconfigure the AudioDevice to have another AudioTransport pointer, which points to an AudioTransportProxy. The AudioTransportProxy is responsible for feeding mixed data to the AudioProcessing component for echo cancellation, and to resample the audio data after AudioProcessing and before it is sent to the AudioDevice. The set up of the audio path was previously done during VoiceEngine initialization. Now it is changed in the AudioState constructor. This list shows where audio-path-related VoiceEngine functionality has been moved: OutputMixer --> AudioTransportProxy VoiceEngineImpl --> AudioState, AudioTransportProxy SharedData --> AudioState Channel --> AudioReceiveStream, ChannelProxy, Channel AudioState owns the new mixer and connects it to AudioTransport and AudioDevice on initialization. The audio input source is AudioReceiveStream, which registers itself with the mixer (which it gets from AudioState) on Start and Stop. # Since the AudioTransport interface contains non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2436033002 Cr-Commit-Position: refs/heads/master@{#15193}
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.