Hi, I'm using the ffmpeg decode engine to receive opus encoded audio over IP and push it into my buffer which connects to my audio driver (custom firmware, not a PC). The audio driver expects audio at 48kHz and plays it at 48kHz locked to its system clock rate. However, the audio coming in is from a different system, so is at 48kHz+/-delta relative to my system clock rate.
How do PCs cope with this sample rate difference? Can FFMpeg be trained to a system clock rate, so that it can resample the audio at the 'correct' rate? The final problem I have is that I want latency to be minimal. Any suggestions welcome. Thanks, Simon
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