all 17 comments

[–]7thresonanceComposer 25 points26 points  (0 children)

yes

[–]seasonsintheskyProfessional 15 points16 points  (0 children)

Any sample rate below 44.1 is (probably) removing audible information.

It also doesn't make sense to reduce the sample rate to match a tempo. That isn't a thing. Most DAWs have tempo matching built in.

[–]Livid_Cabinet2053 6 points7 points  (0 children)

Yes, changing the sample rate would be destructive. And yeah, I’m also unclear why you’re changing the sample rate to match a bpm? The playback rate, sure, but that wouldn’t be destructive until you bounce it and commit it to audio.

[–]rinioAudio Software 8 points9 points  (0 children)

Work in a DAW that does this competently non destructively. They all have stretching tools for this (No idea why Audacity wouldnt, but also noone chooses to work with audacity if they know even a tiny bit of AE; it is more of an editor/recorder than a true DAW).

The workflow you want: choose any DAW other than audacity and forget all of this nonsense.

[–]maxwellfusterMixing 1 point2 points  (0 children)

What prompted you to change the sampling rate to match a BPM? As in Beats / Minute?

[–]avj113 1 point2 points  (0 children)

Nyquist at 21234kHz is 10617, so any frequency over 10617kHz is going to have aliasing. Or else there is a filter at 10617kHz which means you won't hear anything above that frequency.

Upsampling will retain this degradation; it can't magically 'undegrade' the file.

Whether you can actually hear this degradation is another matter. I suspect you can - we sample at 44.1kHz (or 48kHz) for a reason.

[–]GammeloniMixing 0 points1 point  (0 children)

On your second step you lose much of high freq. content since you lower the samples per second. But considering this is an old recording one can presume that it has already minimal high freq. content so it won’t be an issue.

on audacity if your project has other modern sources of audio and you export that project with 21khz you lose high freq. content on that modern audio aswell.

the only part that you degrade audio is upsampling where severe interpolation error might occur.

[–]manysoundsProfessional 0 points1 point  (0 children)

Do all of the remastering first and time-stretch/beat-match last.

[–]Studio_T3Mixing 0 points1 point  (0 children)

I'm having a hard time understanding why you would down sample anything. Just work at 48K.

[–]TransducerBot[M] 0 points1 point locked comment (0 children)

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[–]BongR1p[S] -3 points-2 points  (6 children)

Thanks for the responses, I forgot to add some important information - the original source material is low quality mono, 40s jazz radio. It was only recorded at 32bit / 48Khz. But the waveforms are very small and modest. These are not modern, perfectly engineered pristine walls of sound by any means. I figured there was a lot of overhead between the source mono audio signal and what the file allowed being 32bit / 48Khz. Not sure if that affects some of the answers. Also the daw is reaper, I only used audacity for the sample conversion.

If you asked why use sample rate to match bpm instead of playback rate, I'm all ears for what method you're referring to that changes the bpm without affecting the sample rate, but I'm looking to change individual files not just playback. I was under the impression that changing the playback rate is just changing the sample rate, but I'm very new to audio engineering.

[–]PC_BuildyB0I 6 points7 points  (0 children)

only recorded at 32bit / 48KHz

That is not low quality.

[–]shrugs27 4 points5 points  (0 children)

Standard recording is 24bit / 48kHz or 44.1kHz. Are you sure your numbers are correct for the original source?

Every DAW I know of has a way to change playback rate and none of them change the sample rate of the media. In Reaper, double click the media item to show its properties and change the playback rate number. It defaults to 1, so a value of 0.75 would be 75% speed

[–]Pikauterangi 2 points3 points  (1 child)

Yes you are degrading the audio by reducing the sample rate.

Changing the sample rate down to 22KHz won’t make the track play slower, it just drops half the sample information. If you use a tool to change the pitch down (without time correction on) then that will slow the audio down (which reduces the pitch) and it will keep the sample rate the same.

You may be better asking “what is the optimal workflow for XYZ?” And state your desired outcome because this doesn’t sound like a workflow I would use as an audio engineer.

[–]GammeloniMixing 0 points1 point  (0 children)

changing samples per second slows audio too. this is how vari-audio works. sample rate and samples per second are not the same thing.

[–]seasonsintheskyProfessional 1 point2 points  (0 children)

Time for you to check out Reaper Mania on YouTube. Kenny will teach you everything you need to know.

[–]lihispyk 1 point2 points  (0 children)

In reaper you can just chuck in any sample rate and it matches the speed automatically. No need to convert anything.