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#658523 06/02/21 10:49 AM
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When rendering RTs or RDs from the mixer by right clicking on track actions and saving as a .wav file, what is the file's bit depth, and are the mixer settings (Reverb & Tone) rendered as well?


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Hi,

The audio will render as 44.1 kHz, 16 bit. And yes... pan, volumes reverb, tone, etc., will all be rendered.

If you don't want these settings in the rendering, right-click on "Master"—on the far left of the upper menu—and then choose "Set mix to flat, dry, center". This is how I export because I prefer to add these effects in my DAW.

Regards,
Noel


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Thanks Noel... I'm still not sure of the best way to export the highest quality renderings to my DAW.


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I take it the only way to render to a higher bit depth is to do a batch render to individual files, and not from the track actions command..?


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There's nothing to be gained by rendering the file to a higher bit depth or sampling frequency. It won't change the quality of the file. If fact, it's possible that it might even degrade the quality a little because the computer needs to calculate the missing information. Try and it see if you can hear any difference.

BIAB's audio files are 16bit/44.1kHz and I always leave them as that. This is what I use in my DAW. I record vocals at 16/48 in my DAW (Reaper) and it automatically adjusts the files of different characteristics so that everything plays and renders fine.

Here's one of my songs to show you what I mean...

https://soundcloud.com/noel-adams/i-like-walking-beside-you

Hope this helps,
Noel


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Thanks again Noel. Sure sounds like you know what you're doing. great work...


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I agree with Noel except possibly one point. I was taught that up-sampling a file cannot degrade it in any way. It just adds leading zeroes to the appropriate bytes.


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Originally Posted By: Matt Finley
I agree with Noel except possibly one point. I was taught that up-sampling a file cannot degrade it in any way. It just adds leading zeroes to the appropriate bytes.


That is correct regarding bit depth - going from 16-bit to 24 or 32 will not degrade the audio in any way - as you said, it just adds zeros for anything below -96dbfs (16 bit dynamic range limit). Going from 44.1khz to 88.2khz or 176.4khz similarly should not reduce quality in any way, as the resampler is just calculating samples to go between each recorded sample - though that depends on the resampler algorithm.

Going to a sample rate that is not an exact multiple of the original could theoretically reduce sound quality, as the resampler has to then interpolate each sample from ones around it. Generally speaking, you're unlikely to hear the difference, as most upsamplers have some level of oversampling (multiplying sample rate by 128x for example). Some resamplers do sound bad though, but again upsampling usually sounds fine for most.

Last edited by Simon - PG Music; 06/04/21 12:18 PM.

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Hi Simon,

Following on from what you said, I had a look through the mathematics at Wikipedia (below) and it parallels what you're saying. As Sweetwater point out, apparently with Upsampling the frequency, it's possible to add spectral images that require lowpass filtering.

https://www.sweetwater.com/insync/upsampling/

https://en.wikipedia.org/wiki/Upsampling

Regards,
Noel

Last edited by Noel96; 06/04/21 03:50 PM.

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That is correct, however most if not all audio resamplers have some type of interpolation, where rather than stuffing zeros between samples it figures out an average value of the two surrounding samples and inserts that instead.

Say you have two adjacent samples, one is at -12db and the next is at -6db. If for example you're going from 48khz to 192khz, a resampler would be quadrupling the amount of samples, but rather than inserting zeros (which in the case of 24-bit audio would be -144db) it would average the values in between -12db and -6db, and your samples would be at -12, -10.5, -9, -7.5, and finally -6. This obviously would sound much better than -12, -144, -144, -144, -6.

Of course, resampling with interpolation does have some drawbacks, in that the interpolated samples do not actually represent the original audio. This can sometimes add some sort of distortion, though this is usually at frequencies we can't hear and is usually at a very low level anyway.

The upshot of upsampling audio is that there's usually less filtering required on playback, since the Nyquist frequency is much higher than before.


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Simon,

Thanks for the extra insight!

Much appreciated,
Noel


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No worries, glad to help!


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