hey, audio question. let's say I have an audio DAC. say my laptop is playing samples, at 48 kHz. let's say I also have an audio ADC. say an USB dongle, recording at 48 kHz. their clock isn't synchronized or the same. one is faster than another. say ADC trails. how does this work?
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yep. do you know of any library that can do resampling without a massive dependency tree, or maybe there's some simple algorithm i could just implement myself
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I'll ask around. I thought ffmpeg had something for it but not sure.
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ffmpeg kind of counts as a really huge dependency
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Huge yes but not a sprawling tree. And you might could just nab the relevant code. But I'll look for lighter stuff.
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If you do it yourself you just need a resampler that can be dynamically reconfigured without discontinuities and a control loop to react to clock skew.
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"Just"...

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hmm, looks like libresample might do what I want? http://www.mega-nerd.com/SRC/api_full.html …
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Yes, I think so.
End of conversation
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If it's the other direction the analog domain handles it, but of course you can't expect same number of digital samples on both sides.
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