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Higher Sample Rate When Capturing From Slower Analog Playback Source
JasonCA
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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saratoga
post Jan 6 2013, 22:33
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QUOTE (JasonCA @ Jan 6 2013, 16:16) *
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?


On modern A/D converter's, the clock rate of the actual conversion is more or less constant. If you double the sampling rate, the converter halves the oversampling ratio to maintain the clockspeed its designed for. So if you take twice as long, you effectively do get twice as many samples, and thus sqrt(2) better SNR.

However, as no analog capture device is limited by the SNR of a moderately good A/D, this gain is of negligible value in practical applications. Instead, I would be much more concerned about distortion resulting from playing back an analog device at other then it's design speed. Few mechanical systems are truly time invariant in the way that digital systems are.

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JasonCA
post Jan 6 2013, 23:07
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I appreciate your feedback! smile.gif

QUOTE (saratoga @ Jan 6 2013, 23:33) *
On modern A/D converter's, the clock rate of the actual conversion is more or less constant. If you double the sampling rate, the converter halves the oversampling ratio to maintain the clockspeed its designed for. So if you take twice as long, you effectively do get twice as many samples, and thus sqrt(2) better SNR.


Ok, so the idea is good. Your saying that if you playback the analog source at half the normal playback speed of the audio source, then you are effectively getting twice as many samples. And, you are saying the SNR is therefore improved. But...you are also more concerned about the caveat...

QUOTE (saratoga @ Jan 6 2013, 23:33) *
However, as no analog capture device is limited by the SNR of a moderately good A/D, this gain is of negligible value in practical applications. Instead, I would be much more concerned about distortion resulting from playing back an analog device at other then it's design speed. Few mechanical systems are truly time invariant in the way that digital systems are.


So you are saying here that halving the playback speed of the normal playback speed for a given along audio source, you are worried that the mechanical system or device is not capable of correctly reproducing an analog audio signal that it would normally be able to produce at a faster playback rate?

What about for devices that have a speed selector? There are tape decks (of various kinds) where you can play back the tapes at different speeds (user selectable way to change the IPS of playback)? So sometimes the device is capable of controlling playback. So what if the analog source was recorded originally at a normal playback speed, but then there is a half speed tape IPS switch? If the device supports halving the speed by design, wouldn't this imply that the device is capable of playing back the analog source mechanically at a slower rate?

In terms of vinyl record, I've read arguments that a vinyl record played back on a 45 RPM machine doesn't equate to the same audio output if the vinyl record was originally recorded at 78 RPM. The argument was the needle works better/different on a 78 RPM turntable then it does on a 45 RPM. How true or untrue this is, I'm not sure.

But, in regards to cassette tapes, I don't see why there would be mechanically anything different with a tape head reading analog tape back at slower playback rate especially if the device is capable of adjusting the playback rate of the tape?

So maybe there would be no audio degradation with audio cassettes, but there would be with vinyl records? It's just to say that for maybe some analog source devices, capturing at a slower playback rate would increase the sampling rate on capture and would result in a final digital capture that is no worse (but may be better) than if it were captured at the normal playback rate? Or, maybe there is no real way to know.
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