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Upsampling from 44.1 khz to 88.1 khz conversion problem


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Hi guys,

 

My Dubspot professor told me that in the final mixdown stage it's a good idea to bounce out your audio into a new project and work in 88.2 khz because although the audio will not actually get better, the plugins will have the opportunity to work at a much more complex and more effective state. He said if your computer can handle working in 88.2 you should def do it.

 

Only problem is, when I import my 44.1 khz stems into the 88.2 khz session it converts the audio into double the speed and it sounds like chipmunks. What exactly am I doing wrong here? Couldn't find that answer anywhere on the web!

 

Thanks!!

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You would have to also convert all your audio files to the new sample rate - which IMO is not worth it (converting the sample rates of audio files will introduce distortion), but the best way to tell is to try so you can make up your own mind.

 

In Logic, you can select the audio files in the Bin and choose Audio File > Copy/Convert File(s) to convert their sample rate.

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44.1khz or 44100 hertz is 44100 units being processed per second. If you put this file into a system expecting 88100 units per second, it's going to playback twice as fast, resulting in the 'chipmunk' sound. The task now would be to make it double every sample to fill it in and return it to the original tempo. Try opening your sound in the Sample Editor, go to Factory > Time and Pitch Machine, and then make it half-tempo or 50%; this should fix it, I believe.

 

However... I do question the validity of your professor saying you should work in 88.1 khz for two reasons; one, he works for Dubspot, and two, sample rates over a certain point become null and void to the human ear and I doubt having that many more samples processed in your reverb will improve the sound- at all.

 

Take all of this with a grain of salt; sample rates, bit depth and computer audio confuse me a lot of the time.

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44.1khz or 44100 hertz is 44100 units being processed per second. If you put this file into a system expecting 88100 units per second, it's going to playback twice as fast, resulting in the 'chipmunk' sound. The task now would be to make it double every sample to fill it in and return it to the original tempo. Try opening your sound in the Sample Editor, go to Factory > Time and Pitch Machine, and then make it half-tempo or 50%; this should fix it, I believe.

NOOOOO!!! Do not use the Time and Pitch machine - simply convert the sample rate of the file to match the sample rate of the project.

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NOOOOO!!! Do not use the Time and Pitch machine - simply convert the sample rate of the file to match the sample rate of the project.

 

/facepalm

 

My mind just completely failed there. Time&Pitch is going to attempt to keep it at the same pitch while changing the tempo... which is definitely not what we want Dx

 

Not to derail the thread, but on the subject of sample rate conversion- in this scenario, would it be duplicating each sample? Or would it be averaging the values in-between each sample and adding that in? Or something else entirely?

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My Dubspot professor told me that in the final mixdown stage it's a good idea to bounce out your audio into a new project and work in 88.2 khz because although the audio will not actually get better, the plugins will have the opportunity to work at a much more complex and more effective state. He said if your computer can handle working in 88.2 you should def do it.

I'm afraid your "professor" is talking through his hat.

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I'm afraid your "professor" is talking through his hat.

 

A hat that he's holding in front of his @$$ as he speaks.

 

I've had mix engineers upsample my tracks and process them at 88.2K, but their reason for doing this goes far beyond your professor's half-hat advice (it has to do with processing audio through outboard gear, where the engineers' high-end converters are going to do D-to-A and A-to-D conversion with half the latency you'd get at 44.1K and better overall fidelity running at higher sample rates).

 

because although the audio will not actually get better, the plugins will have the opportunity to work at a much more complex and more effective state.

 

The idea that "the audio won't actually get better" is the first red flag (there's an old saying regarding digital data: "garbage in, garbage out". Holds true for digital audio, photography, videography, etc.) The second red flag is this notion that plugins will work better at higher sampling rates. Well, maybe they will in certain cases, but if the only thing you're processing through them is upsampled 44.1K in-the-box, what's the point? (Again, garbage in, garbage out). Besides, after you get done with this process, you'll just have to down-sample your final mix. Waste of time and effort.

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  • 2 years later...

Well, i had an occasion where upsampling fixed an issue. Flextime. I had a file that when I slightly sped up the tempo there were audible undesirable artifacts. I upsampled to 88k from 44k, mastered the file and shortened the file time length/increased tempo and all was good. There were no audible audio artifacts, just a good finished product.

Jesse Klapholz Elkins Park, PA

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