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Normalization methods for LP digitization

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Lttlwing16

Lttlwing16

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1.4.2? According to https://wiki.audacityteam.org/wiki/Audacity_Versions they went from 1.3.14 to 2.0.0.
Yep, error on my part. Audacity is on version 2.4.1 on Manjaro. Thanks for bringing that to my attention. Messing around in Ardour at the moment.
Normalization is an extremely simple and straight forward process that cannot change the sound. Only the amplitude in a digital way.
Thanks for taking the time to lay that out further.
 
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Lttlwing16

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All good stuff folks. Thanks for engaging and sharing what you know and experiences with the process.
 
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Lttlwing16

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Do you have a way the measure the LUFS loudness of the peak normalized file?
FWIW, Ardour can evaluate the loudness of a track. Pretty cool, with lots of useful info. Below is just an example:

1632936224529.png
 

JP

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They might as well worry about a piece of dust floating in the room landing on top of the tonearm while it was playing.
They do.
 

snaimpally

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Nothing wrong with recording in 24 bit.This is some benefit, as you can set the level conservatively lower without losing resolution. Then after you shift peak levels to -1 dB, you can convert to 16-bit without losing information.

Some (very few, but some) very high quality LPs have frequency response beyond 20 kHz, if you have really high quality properly adjusted playback equipment; recording at 48 kHz sampling can preserve that information. It also gives a wider transition band for the anti-aliasing filter of the AD converter, which can make smoother passband response.

Yes, under some circumstances recording at 48khz might be appropriate. However, if you want to burn the file to CD, then the 48khz file will have to be re-sampled. However, if you use 24 bits for the original recording, it should be fine.

I am a part-time musician and record my concerts (of acoustic instruments) using an Edirol/Roland digital recorder with Sound Professional's Audio Technica cardoid mics. The difference between recording 24 bits rather than 24 bits was a bit of an eye opener for me. I hadn't thought the difference would be so noticeable. It also makes level setting less critical and you can set levels a bit low to allow for dynamics.
 

snaimpally

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I don't see why not.
That's also a sampling rate that one find often on audio recordings.
If you want to burn a CD, the file has to be resampled if the original recording is 48khz. If you record at 44.1khz (24 bit), you can avoid that.
 

dc655321

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The difference between recording 24 bits rather than 24 bits was a bit of an eye opener for me. I hadn't thought the difference would be so noticeable.

Assuming you mean differences between 16 and 24 bits were "an eye opener", um, how?
All else being equal, any differences should be 90+dB down.
 

MRC01

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... Yes, under some circumstances recording at 48khz might be appropriate. However, if you want to burn the file to CD, then the 48khz file will have to be re-sampled. However, if you use 24 bits for the original recording, it should be fine. ...
The few LPs that I recorded at 48 kHz, I burned to DVD in DVD-A format, so I didn't have to resample them. And saved them as FLAC, of course, not just to optical media.
 

rdenney

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Assuming you mean differences between 16 and 24 bits were "an eye opener", um, how?
All else being equal, any differences should be 90+dB down.
But if you are recording source material, having 24-bit depth means you can do things with it without running out of tonal gradation.

So, for me, the eye-opener would be when I applied a +12 dB EQ filter after the fact for some special reason, or needed to boost gain by 18 dB because I was accidentally too low to begin with but still need to salvage the work. That 18 dB is three bits that I would be glad I had.

Rick "noting the person you responded to was talking about making field recordings" Denney
 

dc655321

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But if you are recording source material, having 24-bit depth means you can do things with it without running out of tonal gradation.

So, for me, the eye-opener would be when I applied a +12 dB EQ filter after the fact for some special reason, or needed to boost gain by 18 dB because I was accidentally too low to begin with but still need to salvage the work. That 18 dB is three bits that I would be glad I had.

Rick "noting the person you responded to was talking about making field recordings" Denney

I understand the advantages to having extra bits for recording and processing. The comment I was responding to seemed geared towards playback. Or that's how I understood it, at least.
 

rdenney

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We agree within those contexts, but the sentence you responded to started out as: "I am a part-time musician and record my concerts..." and that's how I saw the use case he was addressing.

Rick "who works in 24-bit color but publishes images online as 8-bit JPEGs" Denney
 
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Lttlwing16

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For those using Audacity who'd like to find the actual peaks within the waveforms quickly, you can Amplify>allow clipping>0.7 dB. You also have to ensure you have "show clipping" enabled under the View menu. Obviously set the level back when you're done. Also useful to import the same file to a second track directly below to actually work with, using the over amplified track as a reference.

1632962152804.png


This can be useful for identifying loud pops/clicks quickly, as well as finding the track to set the level with if going peak capture method.
 
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Lttlwing16

Lttlwing16

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For those using Audacity who'd like to find the actual peaks within the waveforms quickly, you can Amplify>allow clipping>0.7 dB. You also have to ensure you have "show clipping" enabled under the View menu. Obviously set the level back when you're done. Also useful to import the same file to a second track directly below to actually work with, using the over amplified track as a reference.

View attachment 156305

This can be useful for identifying loud pops/clicks quickly, as well as finding the track to set the level with if going peak capture method.
I've gone to using +0.1 Amplify to find the single peak throughout an album. I can then set the capture volume according to that peak.
 

MKreroo

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Interestingly AES recommends -16 LUFS for track normalization, which method 2 was slightly below. Obviously album normalization would be preferred here to preserve the album dynamics from song to song. Album Dynamic is recommended to -14 LUFS. Read more about AES recommendations here.
Came across this one, and I'm now wondering how one should apply this to purchased/owned music. It mentioned that the doc is intended for distributor/streaming service provider, so how should we go about making sure that the source of the file applied normalization? or do we apply the recommended normalization ourselves? (w/t RG etc)
 

Rock Rabbit

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The simple way is peak level normalization, RMS don't take account the crest factor of music genre

IMG_20220307_122740.jpg

And the ear loudness perception curve (i.e.: Fletcher Munson @ 70 dBA). On the other extreme we have BS-1770 that can measure loudness more appropriately. But don't forget it uses a gated method, sliding overlapping window so any peak below 0.4 seconds is discarded!. So depending on crest factor and peak transient time RMS could sound louder. The real problem with any BS-1770 method is compromising original dynamic range by compression!
 

Rock Rabbit

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But even with proper normalization the loudness can be very different!. Using "to each his own" track from America in vinyl and HD we have this statistics
IMG_20220307_113457.jpg
IMG_20220307_113731.jpg
last is digital (lower min RMS level!). They are similar in loudness (-1 dB peak normalized) but they sound very different. Vinyl loudness comes more from the bright treble and HD track loudness comes mainly from high level in bass frequency. In this example is better to use EQ for fair loudness comparison but then destroying the "vinyl" sound.
IMG_20220307_112716.jpg
Then comes the problem with forcing high loudness to a rip track, can it change or alter original dynamic range?
 

Rock Rabbit

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Take a -17 LUFS track IMG_20220307_115119.jpg
And apply -15 LUFS target with -1 dB peakIMG_20220307_115209.jpg
Is easy to observe the "brickwalled" sound effect (many extra peaks) destroying the original good dynamic range.
Probably this could be a good option to listen music on portable gear or in the car, but no high fidelity in anyway.
Use only peak dB to assure original dynamic range (and noise compromise), loudness normalization is only good for streaming
 
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