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Normalize Vs Volume increase
tinpanalley
post Jun 29 2011, 23:08
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I'm having trouble figuring out which of the two is better for me to create a vinyl recording at the highest volume without clipping.

I capture a record via my preamp into Audacity, and I get what seem to be relatively low levels even though I make sure the levels are right when I record (Peak: -6.4db, RMS -24.6db). Am I doing something wrong? Or are those levels right? These recordings don't seem as loud as mastered CDs.

Anyway, to get the most accurate sound possible, should I use normalize or just increase the volume? If normalizing makes everything sound the same at a peak level, doesn't that completely screw with the dynamic range of the music?

Thanks!
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DVDdoug
post Jun 30 2011, 00:43
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Audacity's Amplify effect has an option of setting the peaks to 0dB, which is the same as normalizing.*

IIRC, Audacity's Normalize, adjusts both channels separately which can upset the left-right balance. On the Audacity forum, they seem to recommend Amplify .

Normalizing or adjusting the volume doesn't affect the dynamics, unless you push it into clipping. With clipping, you are getting a nasty kind of dynamic compression where the average volume is increased without increasing the peaks (plus you're getting distortion).

If you've got some loud "clicks" & "pops", these defects may represent the loudest peaks. So, you'll want to normalize after de-clicking.

QUOTE
(Peak: -6.4db, RMS -24.6db). Am I doing something wrong? Or are those levels right?
That's about right (before normalizing). You can go a little higher, but if the peaks "try" to go over 0dB, the analog-to-digital converter will clip (distort).


The ratio** of RMS to peak is determined by the dynamics of the music & recording. I usually check the peak level after recording, and if it's at 0dB, I assume it's clipped, and I re-record at a lower level. (I use a different program, but I believe Audacity shows clipping as red in the waveform.)

These recordings don't seem as loud as mastered CDs. Most modern popular CDs are highly (dynamically) compressed to make them constantly loud (see Loudness War). Compression is a way of increasing the average/RMS level without increasing or clipping the peaks. This is why some people prefer older vinyl over the remastered & over-compressed CD version.

Audacity has a compressor, so you can get a "louder" or "more modern" sound if you wish. Or if you want it listen louder, you can crank-up the playback-volume a bit and enjoy the dynamics! wink.gif




* You can actually "normalize" to other peak values, such as -1dB, but setting the peak(s) to the "digital maximum" of 0dB is the most common.)


** Decibels are logarithms. So the difference in dB is a ratio, and it stays constant when you amplify (multiply) or attenuate (divide). i.e. If you boost the peaks from -6dB to 0dB, the RMS level will also be boosted by 6dB from -24 to -18dB.

This post has been edited by DVDdoug: Jun 30 2011, 01:01
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tinpanalley
post Jun 30 2011, 01:06
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Thanks so much for the thorough reply. So, 2 questions:
1. What do you recommend I do with my settings when I capture vinyl, for someone like me who is all about the purity of the sound as it is reproduced by the stylus and cartridge?
2. I have Sound Forge and have decent knowledge of sound tech. Should I be using that instead of Audacity or is there some other program which would be even better? I'd rather learn a more complex program and capture my vinyl correctly than use a novice program that isn't going to reproduce sound as faithfully.

THANK YOU!!
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DVDdoug
post Jun 30 2011, 01:52
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QUOTE
1. What do you recommend I do with my settings when I capture vinyl, for someone like me who is all about the purity of the sound as it is reproduced by the stylus and cartridge?
CDs are 44.1kHz, 16-bit. These settings are more than adequate for vinyl. There's no harm in using 48kHz (or higher), and if you've got a 24-bit soundcard there's no harm in recording at 24-bits.

But, don't buy-into the "audiophile nonsense" about analog vinyl having "infinite resolution". That's like saying a tape measure has infinite resolution, but a digital caliper is limited. The noise on a record (and in the preamp) limits the resolution... You can (usually) hear noise between tracks on a record, and you can tell if the record is spinning or stopped, or if the stylus is in the groove or not (especially with headphones). You can't (usually) tell if a CD is paused/stopped, because it's very quiet, and has greater dynamic range & resolution than a record.


QUOTE
2. I have Sound Forge and have decent knowledge of sound tech. Should I be using that instead of Audacity or is there some other program which would be even better? I'd rather learn a more complex program and capture my vinyl correctly than use a novice program that isn't going to reproduce sound as faithfully.
The recording software doesn't affect sound quality. It just sets-up the soundcard & driver for the proper format and routes the digital data to your hard drive.

Sound Forge may have better tools for noise reduction or other enhancements in "post production" (after you have a digital audio file on your hard drive).

The hardware is a different story. Some cheap soundcards/soundchips can generate audible noise. (Distortion and frequency response are usually not issues.) If you suspect that your soundcard is contributing noise,* you might look for a better "audio interface". You can get interfaces with line-level inputs, and a few have built-in phono preamps, which you don't need. (But, watch out for USB soundcards... Most "regular" USB soundcards only have a microphone input and line outputs, and are not suitable for high quality recording.)

* I'm talking about recording noise. If you are using your computer for playback, that could be an issue too, but you really want to avoid adding noise to the digitized recording.

This post has been edited by DVDdoug: Jun 30 2011, 02:05
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tinpanalley
post Jun 30 2011, 02:05
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QUOTE (DVDdoug @ Jun 29 2011, 20:52) *
CDs are 44.1kHz, 16-bit. These settings are more than adequate for vinyl. There's no harm in using 48kHz (or higher), and if you've got a 24-bit soundcard there's no harm in recording at 24-bits.


So there's no need (getting the dbs I showed you earlier) to do any normalizing and volume increasing if what I want is the most faithful capture of sound from the record?...within reason
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AndyH-ha
post Jun 30 2011, 02:55
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There is no "need" to amplify or normalize, but there is no downside either. Neither the S/N ratio nor the distortion will change in any noticeable way. The quantization distortion of one, or a few, digital transforms will not be even close to audible, but if you work in floating point format and convert to 16 bit only as the very last step, you will minimize even that.

If any part of your playback system has significant noise, it is more likely to be before the analogue volume control. In that case, amplifying your recording digitally will supply a higher S/N ratio at the speakers.
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tinpanalley
post Jun 30 2011, 03:09
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But doesn't normalizing in making all loudness equal, in essence, affect the intended range of volume from track to track? Because it makes the entire album "loud" by the same amount even if one song is intentionally meant to be quieter?
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tropicalfish
post Jun 30 2011, 05:15
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QUOTE (tinpanalley @ Jun 29 2011, 21:09) *
But doesn't normalizing in making all loudness equal, in essence, affect the intended range of volume from track to track? Because it makes the entire album "loud" by the same amount even if one song is intentionally meant to be quieter?

If you amplify your entire recording (either all in one track, or separate tracks) by the same amount, then you will still get the intentional volume differences between each of the different songs.
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tinpanalley
post Jun 30 2011, 08:31
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QUOTE (tropicalfish @ Jun 30 2011, 00:15) *
If you amplify your entire recording (either all in one track, or separate tracks) by the same amount, then you will still get the intentional volume differences between each of the different songs.


Perhaps this is a naive question, but why then after normalizing, does the waveform of the entire album side have consistently similar peaks and valleys through the whole side such that all the songs look like they are the same loudness and why, on the un-normalized version, can you clearly see which song has more loudness and which is softer?
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db1989
post Jun 30 2011, 12:31
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tinpanalley, your observation matches tracks that have been independently normalised by separate factors, rather than tropicalfish’s scenario of the album being normalised as a single entity or its tracks all being amplified by the same degree.

QUOTE (DVDdoug @ Jun 30 2011, 00:43) *
Audacity's Amplify effect has an option of setting the peaks to 0dB, which is the same as normalizing.*

IIRC, Audacity's Normalize, adjusts both channels separately which can upset the left-right balance. On the Audacity forum, they seem to recommend Amplify .

It still does that? Wow, that is dumb.

So if you wanted to remove DC offset, you’d have to use the Normalize dialog but with normalisation off and only DC offset correction on, and then go into Amplify and adjust it to a peak of 0 dB? That’s some great workflow right thar, even ignoring the potentially damaging effects and simply illogical nature of per-channel normalisation.
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tinpanalley
post Jun 30 2011, 17:51
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Well here's what I have... I think this will help you guys advise me.

I start with my turntable and go into a phono preamp. This one, in fact, the USBPhonoPlus: http://www.artproaudio.com/pro.....amp;id=128

Then I go, via USB, into my laptop where I have Sound Forge 10 and Audacity, but Audacity seemed to be easiest to get my head around (even though I do sound editing with Sound Forge..go figure). Then I tend to use Brian Davies' Click Repair to try to clean things up.

That's the extent of my setup. Am I missing important gear? All I want to do is faithfully capture my records.
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tropicalfish
post Jun 30 2011, 19:25
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Record with Audacity.
Go to Normalize, and check ONLY remove DC offset (or DC offset correction). Uncheck normalize track to ____dB

Then after that, Amplify. It'll automatically find the right amount to amplify to get your highest peak to 0dB.

If you use normalize, then all the tracks (if you record each track separately) will lose the intentional volume differences and they will appear/sound as if they're the same loudness.
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db1989
post Jun 30 2011, 19:43
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QUOTE (tropicalfish @ Jun 30 2011, 19:25) *
Then after that, Amplify. It'll automatically find the right amount to amplify to get your highest peak to 0dB.

If you use normalize, then all the tracks (if you record each track separately) will lose the intentional volume differences and they will appear/sound as if they're the same loudness.
As will also occur as a result of your suggestion to amplify to 0 dB, the only difference being that this option is free of the other’s idiosyncrasy whereby it treats both channels separately (as I noted in the past and as is still the case according to DVDDoug).
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tinpanalley
post Jun 30 2011, 20:31
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Perhaps I should have also stated that the way I record is, I play and record one whole side, and then I play and record the other. I end up with 2 WAVs (SideA.wav and SideB.wav) and then when I take them into Sound Forge I get this (I shrunk it so it wouldn't be so big)


Do those dbs look fine then?

How do I know if my sound card can handle 32-bit or not? Audacity clearly shows it as an option, does that mean it can do it?

This post has been edited by tinpanalley: Jun 30 2011, 20:32
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AndyH-ha
post Jun 30 2011, 20:51
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A recording made with proper equipment should not have any DC offset to fix. The DC offset is your evidence of a hardware problem in the recording chain, or a bad soundcard. I have seen a DC offset in a few LP recordings that was on the LP itself, part of what I got off the disk, not introduced by my recording process.

I have no idea if Audacity or Sound Forge contain tools that can easily tell you if there is anything to concern yourself with, but that isn't a strange or uncommon software function.

Removing the very low frequencies (a rumble filter) will eliminate small amounts of DC offset. Any recording from a TT has a major amount of very low frequency noise you will be better off without anyway.

Normalizing to 0dB, or any other particular value, will not effect the dynamic range of the recording. Doing each channel independently could alter total dynamics but will not change the dynamics within the channel, no matter what the factor (of course that doesn't apply to something really stupid like working in integer format and amplifying enough to drive many peaks into clipping).

Some people like to break a recording into individual tracks and work on each separately. I find it more convenient to record both sides of the LP into one file and keep it that way until I am done with all processing. Normalizing, at the proper time, is then a single operation against the entire album. There is no possibility of altering the album's dynamics.

My LP recording chain is of fixed gain, which is to say I cannot adjust amplitude in the recording step. I normalize almost all recordings, usually to -1.0dB. I suppose there aren't many DACs these days that will clip the analogue output for peaks at or very near 0dBfs, but there also isn't anything to be gained by pushing to 0dB.

Normalizing should be done after any and all filters are applied, especially if you intend to use 0dB. Even when the filter is removing something (e.g. a rumble filter), it may push some peaks past 0dB, which means clipping. Of course if you do the more reasonable thing and work in floating point format there will be no clipping and you have the opportunity to apply a negative amplification to bring the peaks back down below 0dB before finally converting to integer (i.e. to the 16 bit final product), but it seems silly to apply a positive amplification early on just to have to reverse it later.



32 bit, floating point, is a data format, not a hardware format. Any soundcard can record into floating point format, the recorded data is the same as from recording into integer format, but is more convenient to work with. Regardless of all the discussion about 16 bit being enough, the convenience, the lower noise level, and the much smaller quantization errors from any post recording processing all are markedly in favor of using floating point unless you have really puny computer resources.



Nothing looks unusual in your recording. If the other side is much the same, if looks like you can apply about 6dB of amplification, but using actual program measurements to choose the factor is the proper way.

This post has been edited by db1989: Jul 1 2011, 13:22
Reason for edit: Is there a reason that you triple-posted?
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tinpanalley
post Jun 30 2011, 20:55
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Ok, how about if I just control that from the phono preamp? What should my music be levels be peaking at? I always thought it was 3/4 of the way up the meter which ends up being about -12db. Is that wrong?
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mixminus1
post Jun 30 2011, 21:01
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I'll second AndyH's advice on recording both sides into one file - and keeping them as one file - until you're done with all processing. It makes things such as normalization much easier. Once you've got everything cleaned up to your satisfaction, and the levels where you want them, then you can break it up into individual tracks.

I use Cool Edit 2000's cue function to create regions, and then do a batch export to FLAC files - works great. It's been awhile since I've used Audacity's label function, but I think you can get it to do something similar, and I'm pretty sure Sound Forge can do that, too.

I'll also point out that in the screenshot you posted, you have the fairly common situation of the needle drop producing a pop or tick louder than any of the actual music. As such, you want to make sure you edit those out *before* you normalize, or the level of the music will still be several dB quieter than it should be.


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tinpanalley
post Jun 30 2011, 21:06
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I basically already do what you're suggesting mixminus. I record each side and end up with 2 WAVs, 1 for each side. Then I trim off the start and end. Then I do click removal and that's where I am now with trying to understand normalizing vs increasing volume. Then I turn the final edit into several tracks and into flacs. Then I do my metadata, just the way I work.

But at what point am I supposed to be concerned with making a 24-bit recording? Or am I just not able to do that?
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AndyH-ha
post Jun 30 2011, 21:24
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The gain is normally set to leave plenty of headroom when doing live recordings. Setting to reach about -20dB on expected peaks is common. This leaves that 20db to capture the unexpectedly higher peaks without clipping. If you knew for certain what the highest peak would be, you would not need to leave any headroom.

The easiest, most convenient, and most reasonable thing is to record into a floating point format. You will thus capture the maximum resolution of which the soundcard is capable, and with the lowest quantization error, and not have to fiddle with or worry about it thereafter. Stay in floating point until everything is done. As a last step, or next to the last if you intend to pull the recording apart into separate tracks, convert to integer. Here, 16 bit (preferably with dithering and good noise shaping) is perfectly adequate.
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tinpanalley
post Jun 30 2011, 21:30
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QUOTE (AndyH-ha @ Jun 30 2011, 16:24) *
Stay in floating point until everything is done.


So you're saying I shouldn't export to Sound Forge but rather do all my normalizing, etc in Audacity? Seems that when I export both sides to Sound Forge I end up with a 16bit file. Or are these different kinds of bits? Sorry for all my confusion. sad.gif

Format : Wave
File size : 216 MiB
Duration : 21mn 23s
Overall bit rate : 1 411.2 Kbps

Audio
ID : 0
Format : PCM
Codec ID : 1
Codec ID/Hint : Microsoft
Duration : 21mn 23s
Bit rate : 1 411.2 Kbps
Channel(s) : 2 channels
Sampling rate : 44.1 KHz
Bit depth : 16 bits
Stream size : 216 MiB (100%)
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DVDdoug
post Jun 30 2011, 21:36
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QUOTE
How do I know if my sound card can handle 32-bit or not? Audacity clearly shows it as an option, does that mean it can do it?
There are no 32-bit analog-to-digital converters. And the "rumor" is, that most 24-bit soundcards are only accurate to about 20 bits. Your driver may be converting to 32-bit as you record, but of course that does not increase the true resolution... It just makes a bigger file.

You'll have to check your device specs to see if it's 16 or 24-bits. Most soundcards (and interfaces) are 16-bits, and I didn't see anything on the ART website "bragging" about 24 bits.

Audacity (as far as I know all audio editors) work internally "behind the scenes" at 32-bit floating-point (or 64-bit), no matter what format you load-in. As I understand it, DSP (digital signal processing) is "easier" in floating point. From the user point of view, floating point allows you to do things that might boosts the peaks over 0dB (like boost the bass, or mixing) and you don't have to worry about clipping, as long as you normalize before saving in an integer format. Floating-point also allows you to reduce the volume (temporarily) without loosing any resolution. (If you reduce the volume in integer format, the least significant bits are truncated or rounded-off and you loose resolution.)
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mixminus1
post Jun 30 2011, 21:40
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Audacity has the option to export to bit depths greater than 16 - you have to select "Other uncompressed files" from the drop-down in the Save dialog and then click Options...

From there you can select a 32-bit float WAV, which Sound Forge *should* be able to read.

"Worst-case" scenario, you may have to export as "only" wink.gif a 24-bit PCM WAV, but that would still give you more resolution than you need for any editing/processing you might do.

Of course, just recording in Sound Forge in the first place would simplify things quite a bit - what is it about Sound Forge's recording process that you don't like?


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tinpanalley
post Jun 30 2011, 21:43
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QUOTE (mixminus1 @ Jun 30 2011, 16:40) *
Of course, just recording in Sound Forge in the first place would simplify things quite a bit - what is it about Sound Forge's recording process that you don't like?


Oh, I LOVE Sound Forge, I use it for audio editing in filmmaking. I just saw so many people using Audacity for vinyl captures that I thought, "geez, maybe I'm an idiot sitting here trying to use Sound Forge!".
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AndyH-ha
post Jun 30 2011, 22:15
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Audacity is FREEware.

If may also be adequate for some things but it is hardly best at much, perhaps nothing.

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tinpanalley
post Jun 30 2011, 22:16
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Ok, I'm trying Sound Forge now. Aiming for -6db as a peak. The preamp's little light is telling me it's clipping, but Sound Forge says -7.0 has been the max.
Also, it won't let me do 32-bit float, only 24-bit.
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