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Higher Sample Rate When Capturing From Slower Analog Playback Source
post Jan 6 2013, 22:16
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Hi Everyone,

I'm wondering if there is a loss of audio quality by capturing from a recorded analog audio source at a slower playback rate? This applies to those who may not have a device that plays back at a higher playback rate, but instead can only playback at a slower playback rate.

I've seen some people who have had no choice but to playback vinyls on say at 45 RPM since those machines can't playback at 78 RPM. They can then pitch shift the audio digitally to get the playback speed correct. But, the question is...is there audio loss? At a slower playback speed, my understand is you get a higher sample rate on your audio source. With a higher sample rate, wouldn't the quality even better than having had captured it at the normal 78RPM's for a 78 vinyl record?

In this case, I'm not asking specifically for vinyl...but more or less in general. For example, let's look at an audio cassette tape deck as another example that plays back audio at half the normal playback speed. My understanding is that the if you were to record the audio when the cassette tape plays back at half the normal playback speed, you are essentially increasing the sample rate of the audio? True? Or? So when going back to the normal playback speed, you would essentially be digitally down-sampling your captured audio to create a final audio file in which so far to me wouldn't be any worse than to have captured it at the normal playback speed to begin with?

People have said that even though you save an audio source digitally at a down-sampled rate (ex: saving final audio at 44 kHz by down-sampling the captured 192 kHZ audio file), that internally the sound card is already capturing the data at the highest sample rate and then down-sampling it when it passes it off to the software when the software application is recording at a lower sample rate than the sound card is capable. So some would argue that there is no difference between a 192 kHz final audio source file over a 44 kHZ because the 44kHz file was internally created by the sound card having down-sampled the 192 kHZ audio to create it in the first place.

Going back to my original question, what I'm trying to understand is if there is a loss of audio quality by down-sampling or essentially pitch shifting audio that was captured at a slower playback rate? To me, it seems as if the audio quality would instead be better because by playing your analog source back a slower playback rate, you allow the sound card (internally or otherwise) to sample the audio at a higher sample rate than it would have had the chance to do if you played it back at a faster rate. So, to me it would seem the audio quality may even be 'better' by capturing the audio at a slower playback rate?

So whether the audio source is vinyl records or audio cossets or otherwise, does capturing audio data at a slower analog playback speed effect the final audio quality? And, if so why? Or, does it result in essentially the same? Or, is the audio quality even better?
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