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Signal Processing

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  • $\begingroup$ What conversion tool did you have in mind? A good result requires a good conversion tool, but if it's done right you'll get all the fidelity you need. If an audiophile can tell the difference, then there's a problem with the audio system, not the data. $\endgroup$ Commented Apr 30, 2014 at 15:57
  • $\begingroup$ I guess the imperfections they refer to come from the filtering involved in the upsampling procedure. However, these artifacts can be made very small if complexity is no issue. $\endgroup$ Commented Apr 30, 2014 at 16:30
  • $\begingroup$ So, the specific application is streaming audio via an Apple TV. [network source]-->Apple TV-->Optical audio out-->DAC. Apparently, everything going in to or out of the Apple TV converts to 48Khz/16-bit. I could understand the DAC coloring the sound, but assuming nothing on the digital path is lossy, I don't understand why an ideally-implemented upsampling stage would degrade the signal. $\endgroup$ Commented Apr 30, 2014 at 17:01
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    $\begingroup$ There are no ideal digital filters (that are stable and causal and that can be implemented with finite complexity). They will introduce amplitude and phase distortions. These distortions can be made small by increasing the complexity of the filters. Check this link: en.wikipedia.org/wiki/Sample_rate_conversion $\endgroup$ Commented Apr 30, 2014 at 20:45
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    $\begingroup$ If you specify the degree of "degradation" to as small of an epsilon as you want, the interpolation that computes the new samples in between can accomplish that as long as you're willing to pay for it with computational effort. if your interpolation looks at 64 samples (32 before and 32 after your interpolated sample), no one, including dogs, can hear any degradation. you can implement a pretty damn good brick-wall polyphase LPF with 64 samples. $\endgroup$ Commented Apr 30, 2014 at 21:57