Topic: Does normalize affect sound quality?

[PH]

Boojum and others, these are facts, they are not opinions and it's important that you distinguish betweeen the two here.
You can believe whatever you like, but it's a fact that if you edit the gain of any track in any bit realm that you will alter the makeup of the track.
The suggestions that they are identical is ludicrous and misinformed. Try that test for yourself and show us the results.

Any gain editing is called "destructive" editing if it alters the actual file.
The point is to minimize the amount of errors or artifacts that are introduced when doing destructive editing. The best way to achieve this is to a 16 bit file is to upsample, edit, then downsample.
These are facts people.

Ask yourself, if this were folly or just some random opinion, then why are programs like Nuendo and Pro-Tools designed and created to work/edit in a 24 or 32 bit realm with non-destructive editing?
The information and knowledge is out there, so I don't need to do any AB tests, or look up silly articles on google to find out what I already know from years of personal experience working with digital audio to be fact.
and...yes I can hear the difference. That's what I get paid for.

[datbrad]

I want to say that I am not trying to be argumentative but to find out what is true and what is not.  Audio, as in other professions and hobbies, has some built in "ideas" that are not always true when subjected to reason and analysis. 

Some would say this is true of many of the conventions of audio. For example, the theoretical 144 db dynamic range performance of 24bit versus the actual realized dynamic range as limited by the analog front end performance in real world recorders.

[PH]

I would think that increasing the bit depth to 24, then back down to 16 would be more destructive than simple normailzation of the original 16bit file....

Another misnomer. Upsampling a 16 bit file to any other larger bit depth will only add zeros to the end, making the file have a much larger bandwidth to work with for editing, thus minimizing the aritifacts caused by destructive editing of a narrow 16 bit file. Downsampling applies dither, which effectively widens the scope of the 16 bit file with various techniques to hide or mask any artifacts caused by the downsample. In other words, it moves the artifacts to a non audible realm in most cases. What this means is that any artifacts caused by the gain adjustment or downsample are smoothed away and leave virtually no trace of ever happened, which is not the case with 16 bit editing. There is no such built in noise or artifact cover up, leaving the artifacts in the audible realm.

This is really the entire point of this thread.

[Javier Cinakowski]

I understand that nashphil, my point is that normalizing a 16bit file might be less destructive than upsampling, normaizing, dithering and downsampling....   If you are doing serious editing such as eq, effects or filters than upsampling is certainly going to help the situation.  I just think upsampling a 16bit file for the simple task of normalizing is a bit much....   

[PH]

I understand that nashphil, my point is that normalizing a 16bit file might be less destructive than upsampling, normaizing, dithering and downsampling....   If you are doing serious editing such as eq, effects or filters than upsampling is certainly going to help the situation.  I just think upsampling a 16bit file for the simple task of normalizing is a bit much....   

As I have already stated, it doesn't work that way if you are looking to PRESERVE the audio in a lossless and (minimally) non destructive way. 
I don't understand why that isn't clear to anyone reading this thread. It's a really simple concept based on fact, not myth. 

As has been stated already, it depends on what you want to normalize it for as to what you should do.
If you don't care about artifacts, then go for it. If you do, then try using a professional audio editor to edit your files.
Soundforge, Wavlab, and Nuendo all get plenty of use in my house. They are designed to handle these type of issues for you.
Programs like Nero are not.

Judge for yourself. Your ears will answer these questions for you.
If you can't hear it, then you can't hear it and it doesn't matter, what else can I say.

<<talk amongst yourselves>>

[datbrad]

Phil, What is your opinion when recording at 16 bit regarding 48khz sampling, then resampling to 44.1, versus just mastering at 44.1? I have heard mixed opinions on that. Some say that the analog anti-alaising filters in ADs at 44.1 impact the sound quality more than doing 48khz, since the filters are kicking in at a higher frequency. Then, the thinking goes, resampling to 44.1 delivers a better end result than if mastered at 44.1 originally, supposedly. I have always stayed at 48khz, but now wonder if I am really wasting time with that step.

I ask you this because at one time, resampling was thought to be a very destructive thing to do to audio files. I suppose with today's higher end software, this is not as much an issue, but you are the studio guy and should know.

[PH]

Phil, What is your opinion when recording at 16 bit regarding 48khz sampling, then resampling to 44.1, versus just mastering at 44.1? I have heard mixed opinions on that. Some say that the analog anti-alaising filters in ADs at 44.1 impact the sound quality more than doing 48khz, since the filters are kicking in at a higher frequency. Then, the thinking goes, resampling to 44.1 delivers a better end result than if mastered at 44.1 originally, supposedly. I have always stayed at 48khz, but now wonder if I am really wasting time with that step.

I ask you this because at one time, resampling was thought to be a very destructive thing to do to audio files. I suppose with today's higher end software, this is not as much an issue, but you are the studio guy and should know.

This is a very good question!
From 2000 to up until about a year ago I was recording everything at 24bit/48k, both in the studio and for live stuff too. A well known Nashville engineer and I were talking about this very subject and after listening to his thoughts and then following up with some of my own tests, I changed my mind. I now record at 24bit/44.1k.

Since 99% of what I do is to wind up at 16/44, why resample. I don't ever mix anything down to 16/48.
The differences in quality are just NOT great enough to warrant the extra time, space, and potential artifacts from the resample.
I decided a long time ago that 96k is a ridiculous waste of space and really more about gear hype than actual improvements. The real difference comes at the 16/24 bit level.

Now, if I were still recording at 16bit in the field, I would use 48k. I have always sensed a difference in that at 16bit, even going back to the DAT days in the 90's.
Once you reach the 24 bit realm, that difference is virtually impercievable from 44k to 96k. Some would argue this point I'm sure, but my experience tells me otherwise.
I guess the differences are probably pretty small here, so I don't waste time with the extra step. Makes my life easier.

[Josephine]

I am so happy to read your comments above, Phil.  I've recorded two shows now at 96k (just because I could) and have come home wondering why.  The files are massive and will be a pain in the ass to store. 

So despite my "ability" to record at 16 or 24, at 44.1, 48 or 96, you'd recommend running @ 24/44.1?  It indeed would make life so much easier. 

[boojum]

Boojum and others, these are facts, they are not opinions and it's important that you distinguish betweeen the two here.
You can believe whatever you like, but it's a fact that if you edit the gain of any track in any bit realm that you will alter the makeup of the track.
The suggestions that they are identical is ludicrous and misinformed. Try that test for yourself and show us the results.

Any gain editing is called "destructive" editing if it alters the actual file.
The point is to minimize the amount of errors or artifacts that are introduced when doing destructive editing. The best way to achieve this is to a 16 bit file is to upsample, edit, then downsample.
These are facts people.

Ask yourself, if this were folly or just some random opinion, then why are programs like Nuendo and Pro-Tools designed and created to work/edit in a 24 or 32 bit realm with non-destructive editing?
The information and knowledge is out there, so I don't need to do any AB tests, or look up silly articles on google to find out what I already know from years of personal experience working with digital audio to be fact.
and...yes I can hear the difference. That's what I get paid for.

It's audio basics. You can find plenty of information on the subject with a google search.

Phil, I am sure you are absolutely convinced of what you say being the actual fact.  I am trying to ferret out proof of this as I do not know what you know.  Some folks in the business still swear analog is better.  It may sound nicer but that is a euphonious distortion.  It is not as accurate because it is distorted.  I am trying to get to where I, we, can determine whether this is the case in normalizing or not.  What you say may be true.

But when you first say "google it" and then say the googled stuff is silly you weaken you argument a bit.  That does not make your argument wrong, though.  I am sure you are way better a ME than I will ever be.  But I am just trying to find a definitive proof of this, that normalizing is bad as it "adversely affects sound quality."   I have not tried the experiment suggested with normalizing a file and then restoring it and comparing it to the original.  I am not sure I can do it in SAM, maybe in Cool Edit.  If it can do this the argument is done.  If it cannot it is still up in the air as we are looking for "adverse effects" and maybe we should define our terms.  

On gearlsutz and PSW they are divided on this.  There is another "pro" board out there I have forgotten, but I will try there if I can remember it.  

The search continues.

Cheers

[PH]

I am so happy to read your comments above, Phil.  I've recorded two shows now at 96k (just because I could) and have come home wondering why.  The files are massive and will be a pain in the ass to store. 

So despite my "ability" to record at 16 or 24, at 44.1, 48 or 96, you'd recommend running @ 24/44.1?  It indeed would make life so much easier. 

Yes, running at 24/44 would be the best. There would be virtually no benefit to running at any sample rate above that.

[Gutbucket]

...Now, if I were still recording at 16bit in the field, I would use 48k. I have always sensed a difference in that at 16bit, even going back to the DAT days in the 90's.
Once you reach the 24 bit realm, that difference is virtually impercievable from 44k to 96k..

I totally agree on NOT using an algorithm that is not understood and found in some inexpensive or free software (mostly because it is probably not as well designed or thought out) to adjust the level of one's master recordings.

I also totally agree on 96kHz being overkill and that the true quality jump is adding those 4 or 8 bits over 16 to leave headroom for real world recording (I think its harder to justify that those bits are necessarily the delivery medium).  I'll take your word on the 48 vs 44.1 difference since I've never really tested it.  I doubt I'd hear much of a difference since I don't in very casual comparisons.  However, I don't follow the part above.  Why would choice of bit depth have any influence on bit rate? Specifically in this case, why would a lower bit depth recording benefit from a slightly higher sampling rate?  Conversely why would the increased dynamic range of a 24 bit format effect what should be frequency artifact issues with sample rate changes?


As for the original question- From the theory side, if you analyze a WAV file you can find the highest and lowest sample values. There is no information stored above or below those values.  Assuming you use good tools and understand the implications and real world issues that those tools and their use may have on the file and if you use the correct practice to do so, there is no information lost when shifting the entire range of recorded values in the file up or down within the larger range of the allowable values determined by the bit depth of the file. Doing a simple gain addition to all sample values of the file (which is what we are talking about 'in theory') doesn't loose any information as long as that addition keeps the highest sample value beneath the maximum value allowable.  Experienced users would suggest using good tools to do so and leaving some room at the top.  Doing so shifts the range of values up into the unused portion at the top and adds useless zeros to the bottom.  Same goes for adjusting the other way.

In the real world nothing is simple or straight forward.  If you are doing any other manipulations of the file there are implications and complications.  You need enough 'padding' around the edges of the information to allow for those changes.  Here's a good question, "what do they really mean by normalizing?"

Does normalizing affect sound quality? Like everything the answer is "it depends".  If you are a realist who understands the complications of real world software and equipment because you work with it for a living everyday like Phil, the answer is "Yes usually, unless you are sure of what's going on and use good tools".  If you are a theorist the answer is, "No, with a lot of qualifications".  If you understand the complications, you'll understand those answers approach the question from opposite directions, but are both correct and communicate the same concerns and qualifications. 

[jerryfreak]

for what its worth, soundforge and most other decent wave editors do all the calcs in 32-bit floating point anyway.

normalization done *once* with software like this should essentially sound as good as the original

I would think that increasing the bit depth to 24, then back down to 16 would be more destructive than simple normailzation of the original 16bit file....