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This article mainly records the remote and local audio streams of WebRTC into mp3; it also involves the use of WebRTC's resampling algorithm and the use of audio mixing; about the audio class, you can add a class to the WebRTC source
code file (such as LYMMdiasRecord.cc), and then modify the files in the corresponding directory
gn
, and add the newly added files to participate in the compilation of WebRTC. The newly added class should be a global class (because there may be multiple peers in WebRTC), so this class The initialization suggestion is placed in the
PeerConnectionFactor