[Libav-user] Synchronising audio to system clock

[Simon Brown]

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://ffmpeg.org/pipermail/libav-user/attachments/20211012/2286bf37/attachment.htm>