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Capture microphone input and stream it out to a peer with a processing effect applied to the audio. The audio stream is: o Recorded using live-audio input. o Filtered using an HP filter with fc=1500 Hz. o Encoded using Opus. o Transmitted (in loopback) to remote peer using RTCPeerConnection where it is decoded. o Finally, the received remote stream is used as source to an <audio> tag and played out locally. Press any key to add an effect to the transmitted audio while talking. Please note that: o Linux is currently not supported. o Sample rate and channel configuration must be the same for input and output sides on Windows. o Only the Default microphone device can be used for capturing. R=phoglund@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1256004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4006 4adac7df-926f-26a2-2b94-8c16560cd09d |
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