04c0722cf1
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}
18 lines
528 B
Python
18 lines
528 B
Python
include_rules = [
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"+webrtc/base",
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"+webrtc/call",
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"+webrtc/common_audio/resampler",
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"+webrtc/logging/rtc_event_log",
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"+webrtc/modules/audio_coding/codecs/mock",
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"+webrtc/modules/audio_device",
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"+webrtc/modules/audio_mixer",
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"+webrtc/modules/audio_processing/include",
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"+webrtc/modules/bitrate_controller",
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"+webrtc/modules/congestion_controller",
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"+webrtc/modules/pacing",
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"+webrtc/modules/remote_bitrate_estimator",
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"+webrtc/modules/rtp_rtcp",
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"+webrtc/system_wrappers",
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"+webrtc/voice_engine",
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]
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