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We are unable to use HeyGen Realtime API as it is because it expects audio in PCM codec 16Bit @ 24KHz with bytes encoded as Base64. The WebRTC audio is mostly Opus codec @ 6-128kb.

How do we transcode the webrtc/opus codec raw audio to your requested PCM 16bit 24Hz base64 bytes? Alternatively Is there a way you guys can accept raw Opus audio directly from an ongoing WebRTC call?

How do we transcode the webrtc/opus codec raw audio to your requested PCM 16bit 24Hz base64 bytes?

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