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<p>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?<br>
</p>
<div class="moz-cite-prefix">On 2021-10-12 10:35 a.m., Simon Brown
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CAHW09jd1pcVOZKg6wwCkXt+sjx1Y2KCXJVjU_zHZrOCeUCN09Q@mail.gmail.com">
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<div dir="ltr">Hi,
<div>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. </div>
<div><br>
</div>
<div>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. </div>
<div><br>
</div>
<div>Any suggestions welcome.</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Simon</div>
</div>
<br>
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