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
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