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is WAV normalization lossless?, such as MP3 with MP3Gain
kjoonlee
post Mar 22 2006, 11:39
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If you can't accept actual use by professionals as valid proof of the meaning of "lossy" or "lossless", then I suggest you say "artificial" to only mean "artistic," "treacle" to only mean "salve for snake venom," "culture" to only mean "agriculture," "starve" to only mean "to die by freezing," and "lossy" and "lossless" to only mean what you want it to be, regardless of actual use.

This post has been edited by kjoonlee: Mar 22 2006, 11:41


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john33
post Mar 22 2006, 13:00
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QUOTE (cliveb @ Mar 22 2006, 09:43 AM)
I don't have any practical experience using WavGain, but my understanding is that it uses the ReplayGain algorithm to determine a linear gain that is applied to the WAV file. I don't know whether it backs off the required gain in order to avoid introducing clipping - perhaps someone else here can verify this.
*

Yes, it does, unless you tell it not to, in which case brute force peak limiting will be applied.

I'm sure the OP lost interest in this thread long ago, but he had his answer early on. wink.gif

Any process applied to the original wave data that is not exactly reversible is, by definition, a 'lossy process'. This also includes the dithering of studio masters down to 16 bit for CD masters. Whether, or not, the changes made are audible is completely irrelevant.

We also have 'lossless compression' and 'lossy compression', but these need no further explanation.


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SMT [AQP]
post Mar 22 2006, 13:49
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QUOTE (AndyH-ha @ Mar 20 2006, 03:42 PM)
...There is no argument on my part that there isn't an irreversible change, but I suspect you can't find your definition of lossy anywhere used by the rest of the world. Normalization is definitely not compression, of any sort. Lossy just isn't the correct word in that context. Show me that I'm wrong and I'll acknowledge it. Ok, I'll even acknowledge it here: it is the slang usage this group has adopted as its own in-group speak; I should recognize the sloppy terminology in the future without comment.
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QUOTE (AndyH-ha @ Mar 22 2006, 02:48 PM)
This isn't about what I want to call a destructive edit, it about what the 'professional' audio world calls it. Being professional 'they' are a little more exacting with their use of technical terms than is this neighborhood.
*

Wow, only now do I realise that HA is full of "non-professionals" stuck in some freaky alternate universe. yeahright.gif
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smok3
post Mar 22 2006, 13:53
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from wikipedia:
QUOTE
Technically, reducing text size by removing all vowels can be considered a lossy data compression as well.


so the usage of the term is of Technical nature and as such appropriate for forums like this one.

This post has been edited by smok3: Mar 22 2006, 13:57


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cliveb
post Mar 22 2006, 14:38
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QUOTE (AndyH-ha @ Mar 22 2006, 11:22 AM)
You introduce a new descriptor "formal sense" into the debate. Have you any evidence, that being the sticking pont or the whole issue? I maintain that is an incorrect use of the term lossy. Its not losing anything, its changing it, but more importantly, it isn't the accepted technical use of the term.
*

I don't think anyone around here denies that even simple linear gain adjustment is a non-reversible operation, so there's no need for "evidence" as such. I guess the sticking point is what types of non-reversible operations qualify for the description "lossy". I'm a practical guy - I don't really care much about the theory: results are what interests me. And in that respect, I strongly agree with what you said in the earlier parts of this thread: for all practical purposes, linear gain adjustment is not lossy (in the sense that we usually use the term around here - for things like MP3 encoding, etc).

But then it all got a bit deep and adversarial over the strict meaning of the term "lossy", and I don't feel qualified to contribute to that part of the thread. (I personally feel that all non-reversible operations are "lossy" in a strictly hair-splitting sense, but I don't have any letters after my name that make my views authoritative).
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AndyH-ha
post Mar 23 2006, 02:18
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You people who want to debate really should make a better effort. I've acknowledged that I might be wrong. I've shown you the evidence that supports my position -- or at least pointed you at it, if you care to consider it. What do I get in return. In many cases (to distinguish from all cases), I see replies that come across as someone screwing shut their eyes, stamping their foot, and demanding their word be accepted. Evidence? Who needs stinking evidence when they have an emotional need to be right? Forget about facts, just experience belonging.

There has been some effort to present evidence. stephanV pointed out another legitimate use of the terms lossy and lossless. It has nothing to do with audio or any operation on audio data, however.

smok3 just provided an actual reference, but it has a couple of difficulties. First, it is from wikipedia. This may be a generally useful reference source but is it not the case that anyone can make entries? If I made a detailed entry strictly expressing the position I've taken in this thread, are you likely to now accept that as authoritative?

More importantly, that entry just happens to be an example of what I've been saying is the one and only correct usage. It supports my position, not my opposition.

There may indeed be professions participating here. I say nothing against them generally, especially as I know nothing. I'll take it on your word that they are very competent at doing whatever they do professionally. Let's just consider this hypothetical, however: A individual might get along swimmingly in a local pond, and be very influential, especially when he can demonstrate ability and insight into some topics, yet not realize that out in the big world his professional colleagues roll their eyes behind his back whenever he talks about some certain subject.

In other words, as I've repeatedly said, what happens here in HA isn't necessarily indicative of views or understandings in the world at large. If it is, in regard to the current topic, where is the evidence? john33 says, as have a few others, "by definition," but what definition? The HA definition! The terms certainly do not seem to be defined that way elsewhere. I wouldn't discount the possibility that some of you have infected a few other forums, where immunity to error is low, but that still doesn't qualify as ‘the professional view.'
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kl33per
post Mar 23 2006, 02:32
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You want proof:
1. Take a audio track
2. Make a copy
3. Normalise the copy
4. Hash the original file
5. Hash the copy
6. Compare the hash codes

If you use a decent algorith (i.e. MD5) the hash codes will not match, garunteed. Furthermore, there is no way to restore the original audio. Normaliseing back to orignal levels will not create the same audio data. Therefore, any form of normalisation is a lossy process.... Period... Full Stop... Close The Thread.

Now for some definitions:
QUOTE (From dictionary.com)
Main Entry: loss·less
Pronunciation: 'los-l&s
Function: adjective
: occurring or functioning without loss

<algorithm, compression> A term describing a data
compression algorithm which retains all the information in
the data, allowing it to be recovered perfectly by
decompression


It would seem to me the definiton of lossless (meaning no loss whatsoever) is pretty standard.

This post has been edited by kl33per: Mar 23 2006, 02:46


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Farpenoodle
post Mar 23 2006, 03:21
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So what DO they call it then? I'm willing to bet they just call it 'volume normalisation.' But it's still a lossy process no matter how you look at it. A professional might go, 'hmmm I need to normalise these tracks.' And he will do it knowing that the lossy effect is negligeable. Still lossy sir.


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audio2u
post Mar 23 2006, 04:58
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QUOTE (AndyH-ha @ Mar 22 2006, 08:22 PM)
Bourne, the original inquirer, asked about CoolEdit's Group Waveform Normalize which I believe is an RMS based process used to balance perceived loudness. I'm not positive about that, as I've never had interest in doing that task, but I seem to recall reading that is its purpose.
*


That's correct... Audition's (the app formerly knows as...) Group Waveform Normalise function does works on RMS values, not peak values.


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AndyH-ha
post Mar 23 2006, 05:12
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Come-on guys, at least try to follow along so you might have a chance at adding something new to the debate.

As for closing the thread, it has been fun. I may check occasionally to see if something non-repetitive comes along, but there are plenty of interesting things to engage the attention, including things right here at good ol' HA.


Thanks for the confirmation on Group Waveform Normalise.
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Bourne
post Mar 26 2006, 07:25
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[JAZ]
post Mar 26 2006, 10:56
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@ Bourne:

Not sure if you care any longer about this debate, but the points were:

1) When decoding (a), the effect of changing volume (b) will change the values to adapt to the desired volume.

2) This change of volume is generally small enough (+-10dB) that any problem derived from it is not going to be heard.

3) MP3Gain can be applied to an mp3, (the file!), and changed at a later time without losing any quality (the audio data is not modified, only a multiplier)

4) WavGain (the one applied to .wav files, not lossless files), cannot be reverted back if desired so, because the audio data is modified.

5) Some Lossless codecs have "replaygain", which is mp3gain/wavgain but stored as a multiplier too (similar to mp3gain, but more precise).

6)In these lossless files, replaygain can be modified losslessly.


Closing)
If you make an *audio* cd, use a lossless codec with replaygain or wavgain the .wav's. This will give you the best you can get.

If you make a *data* CD, not an audio CD, the best thing you can do is use one of these lossless codecs, like WavPack : http://www.wavpack.com/ . You can put the mp3's there too if you want, to play them on any mp3-compliant player.



(a) burning an *audio* cd, playable on normal cd/dvd players is a case of decoding.

(b) mp3gain, wavgain, replaygain, rms normalization..
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Kees de Visser
post Mar 26 2006, 14:28
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QUOTE (Bourne @ Mar 26 2006, 08:25 AM)
From what I read, then there's no way of making WAVs the same level losslessly


The "loss" by changing level in a WAV audio file is so small that you shouldn't worry about it, assuming it's done correctly.
I think much of the confusion is caused by not distinguishing between "audio" and "data". Audio can't be copied or transferred losslessly, but data can. AD and DA conversion (of audio in this case) will never be lossless, although the loss can be made as small as possible.
Modification of the data doesn't necessarily mean that the resulting (analog) signal will degrade in quality.

What you are basically trying to do with your different sounding music tracks is called "mastering". In the context of this thread that could be described as the "intentional modification of (digital or analog) audio in order to optimize it for its intended purpose". True (professional) mastering is an impossible task when you want to apply it to your complete song collection. Level matching (not the same as normalizing!) can be a good approach and imho WAV is the more robust format (compared to mp3), especially if you want to apply EQ, dynamics- or any other signal processing afterwards.

Just my 2 (euro)cents smile.gif
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Bourne
post Mar 26 2006, 18:31
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Shade[ST]
post Mar 26 2006, 19:07
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My suggestion to you, would be to abx several solutions :
lame -V4 --vbr-new with mp3gain;
lame -V4 --vbr-new without mp3gain, but with source file wavegained;
lame -b 320 with mp3gain;
lame -b 320 with wavegained original

you can use abc/hr or foobar's abx methods : then, if you can't tell them apart from the original, you'll be better off choosing the smaller file size (which should be lame -V4 --vbr-new with wavegained original.
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singaiya
post Mar 26 2006, 19:40
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bourne: if I understand you correctly, you have a bunch of WAV files of differnet artists with different volumes, and you want to burn a mix CD so they sound the same volume, but without the files losing the original volume information. In that case you can do it like this:

1. convert the wav files to lossless format. My choice is Wavpack, but you can use whatever lossless format you like.

2. Load all the lossless files (not the wavs) into Foobar 2000 and do track replaygain on the files. This will not alter their volume, but just add a tag that says how to interpret the volume changes.

3. Burn the CD with Burrrn, enabling the program's replaygain option.

Then the CD will have the same volume on every song, and your lossless files will have the original volume information also. Good luck.
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Bourne
post Mar 26 2006, 19:43
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The final burned audio in CD is going to be modified, isn't it? Anyways, that's a good option.

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Shade[ST]
post Mar 26 2006, 20:00
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it will. Another option would be to use foobar2000 to convert the files to mp3 using replaygain/track gain. (edit:after scanning with replaygain)

You probably can't distinguish ~130 kbps mp3 anyways.

Try it wink.gif

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Bourne
post Mar 26 2006, 20:04
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Shade[ST]
post Mar 26 2006, 20:14
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Right.

1) no.

2) I'm not too sure what you mean about this, but the replaygain scanning is automated. You just need to update your tags afterwards.

A thing to note, though, is you'll need to change the replaygain mode used on mp3 conversion. It should be defaulted to album, you want it on track.
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AndyH-ha
post Mar 26 2006, 20:26
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Level matching (when applied to audio or any electronics) is more commonly used to mean bringing differing signals in a system to a common voltage level so they can function together properly. There are various ways to do this to electrical signals. One such is with a transformer to increase or decrease the voltage output of a particular device.

Not all equipment conforms to the same ‘line level’ voltage. If devices made to differing standards are used together, their outputs frequently must be level matched or you can’t get the proper contribution from some of them. The term is often used when both balanced signal and unbalanced signal equipment is being used together. One must be brought to the level of the other or the results won’t be very good.

You interest is loudness matching. Group Waveform Normalization attempts to do that by working with the RMS measurements rather than the peak levels that regular normalization uses. It is possible that some other programs achieve a more uniform loudness than does CoolEdit, but no program is going to be more precise or ‘correct’ than CoolEdit. Some of the other programs might do the job with more crude calculations or by making more radical changes to the data.

You want a certain ultimate result. You want the loudness of at least some of the songs you write to CD to be greater or lesser than their original source. You want this in order that all songs on the CD sound about the same loudness when you play the CD.

There is absolutely no way to do this by any means in this physical universe without changing those tracks that are going to be louder or softer. The only question is ‘will you change the source file or will you make the change only as the signal is on it way to becoming sound?’ The ultimate result, what you hear, will be identical in either case.

MP3s and some other formats allow the possibility of the second option, dynamic change to the signal rather than static change to the source. An audio CD absolutely does not, no matter what route you take to get there. The change must be made to the file before it is to be written to the CD. If someone tells you anything which seems different, they simply don’t fully understand what happens or they are deliberately confusing you for some purpose of their own.

There are some hardware device that attempt to maintain a constant loudness regardless of the signal fed to them. The Automatic Loudness Control (I think that’s the label), at one time widely used on car radios, is one such hardware approach. These methods have never been considered HiFi because of how much they mangle the audio in the process. However, if you play your audio CD in a system with such a control, then, and only then, can you get the loudness results you want without having changed the tracks before you put them on the CD.

If you hide this fact from yourself by utilizing Mp3Gain, or any other means that does not change your hard disk source, it is still being done for you. The program you use creates a temporary file with the changes, writes that temporary file to the CD, then deletes that temporary file so you never see it.

If you apply Group Waveform Normalize in CoolEdit, then do a Save As and write the changed output to a different directory, so that your original source remains unchanged, then write the CD from the files in that different directory, you do the same thing without lying to yourself about it.

If you have no use for the original source on hard disk after writing the CD, it is simply silly to do the different directory thing since you don’t have any reason to care that the original source gets changed. You are only going to delete it all once the CD is finished. But regardless of how you do it, and regardless of whether you insist on calling a change a loss, what you hear will always be created from samples with different numerical values than those in the diverse sources you started with. To think otherwise is to confuse yourself.
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Bourne
post Mar 26 2006, 20:43
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Shade[ST]
post Mar 26 2006, 20:46
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Right. It's automatically 89 db.
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Bourne
post Mar 26 2006, 22:05
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Shade[ST]
post Mar 26 2006, 22:32
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QUOTE (Bourne @ Mar 26 2006, 04:05 PM)
oh my... that's really curious isn't it... never thought foobar would do the same 89 db... !
*

You _can_ set it if you like. There's a slider, when you change it from album mode to track mode (or at least in .9, which is faster, there is)
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