[Libav-user] Synchronising audio to system clock

drwho drwho at infidigm.net
Tue Oct 12 17:41:15 EEST 2021

One of the audio hifi streaming devices used to drop or insert samples.  
In newer versions they fine tune the audio pll to match the stream 
rate.  Beware Sonos has a patent on the behavior you just described.  
What is your application?

On 2021-10-12 10:35 a.m., Simon Brown wrote:
> 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
> _______________________________________________
> Libav-user mailing list
> Libav-user at ffmpeg.org
> https://ffmpeg.org/mailman/listinfo/libav-user
> To unsubscribe, visit link above, or email
> libav-user-request at ffmpeg.org with subject "unsubscribe".
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://ffmpeg.org/pipermail/libav-user/attachments/20211012/ea51cf0a/attachment.htm>

More information about the Libav-user mailing list