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.

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.

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.

Wow, only now do I realise that HA is full of "non-professionals" stuck in some freaky alternate universe.

from wikipedia:


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

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).

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.'

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:


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

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.

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

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:

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..

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

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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.

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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The final burned audio in CD is going to be modified, isn't it? Anyways, that's a good option.

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

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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.

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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Right. It's automatically 89 db.

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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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This link gets posted fairly often in response to the loudness race:
http://www.prorec.com/prorec/articles.nsf/...6256C2E005DAF1C

have i use EAC to extract the tracks in Flac or wave? thanks!

about normalizations with audition:
'The Normalize effect lets you set a peak level for a file or selection. When you normalize audio to 100%, you achieve the maximum amplitude that digital audio allows--0 dBFS.

The Normalize effect amplifies the entire file or selection equally. For example, if the original audio reaches a loud peak of 80% and a quiet low of 20%, normalizing to 100% amplifies the loud peak to 100% and the quiet low to 40%.

To apply RMS normalization, you must use the Group Waveform Normalize command. If desired, you can apply that command to only one file. '

is RMS normalization lossless?

Read the thread and find out The answer is the same regardless of whether you use peak or RMS method.

thanks!
i read the whole thread yesterday and did some tests:
source Beatles-Martha my dear extracted with EAC.
in audition:
to see how much the track need to amplify to 0dB was used amplify/fade
with peak level in 0dB that show me 4.14dB(62,9%) of amplification with lock left/right!
than i:
normalize the track to 100%- saved as norm100.wav
using 'calculate now' to amplify the track to 0dB- saved as ampl0dB.wav

using the "saved norm100.wav" and normalize again to -4.14dB(62.9%)- saved as finalnorm.wav
using the "saved ampl0dB.wav" and amplify/fade to -4.14dB(62.9%)- saved as finalampl.wav

comparing source ripped with the "finalnorm.wav" and "finalampl.wav",
nothing is change all tracks seems(waveform view) and sounds equals,
for more than i listen or use the zoom tool to compare!

please,give me more explanations and tell me if i forgot something!
thanks.

The entire point of this thread is that the changes are always much too small to notice any change with human senses, regardless of how you may wish to label the results. You can try till the cows die of old age and you will achieve nothing. You can MEASURE that there are small changes if you use the right tools. This is absolutely standard, so well established that you might as well worry whether or not hydrogen will forget how to fuse and produce helium as to fret with it any further.

It depends on the source material. I could make a 16-bit wav with high dynamic range variations on the audio that would be audibly affected by replaygaining it. The replaygain process would reduce the dynamic range in a way that when reversed would result in an audible increase of the noise floor and loss of the lowest level detail. In real world, in most cases this won't happen, but you can't say it won't ever happen.

Now, in case of 24-bit wavs I don't think I could make a wav where an audible difference could result as a consecuence of replaygaining. And 32 bit floating point (and higher) wavs can be replaygained in a in practice lossless way for all purposes.

ReplayGaining .wav files clearly isn't lossless, and isn't reversible.

It's also not as "lossy" as, say, mp3 encoding.

I'm still happy calling it "lossy", but for those who want to draw a distinction, call it "near lossless".

This matches the situation with some "lossless" audio codecs which can be forced to reduced the bitrate by operating in a "near lossless" mode, which sacrifices the accuracy of the LSBs in order to reduce the bitrate - but not nearly as dramatically as, say, mp3.


KikeG just beat me to an important point: if you have a carefully noise shaped dithered 16-bit file with a huge dynamic range, then a slight level change (implemented without dither, or with optimal but spectrally flat dither) could introduce audible noise into the quiet parts, if you turn the volume up when listening.

The chances of it happening are small, even in an intentionally contrived "worse case" situation, but they are there.


I think AndyH-ha is wrong about "the real world". Most people who own a computer will be familiar with the concept of "lossless", even if they can't name it. It means no loss, which implies no change, which implies I can do something as many times as I want without causing any problem what so ever.

So I can zip and unzip documents, copy them from memory to disk to network to where ever, pass them on to friends and let them to the same multiple times - and if so much as a single character changes in the document, I'll say that there's a fault somewhere - because these operations are lossless.

Similarly, if ReplayGaining .wav files was lossless, I could ReplayGain them as often as I want, to whichever loudness I choose. I could pass the files onto friends and let them do the same. The file could pass through the hands of every person in the world, each one changing the gain to a different level, and when I get the file back I could just change it back, and have my original file.

This is clearly nonsense. If I tried this game of ReplayGain Chinese Wispers with a .wav file, I'd be lucky to get any music back at all! However, if I tried it with an mp3 file, there would have to be a fault somewhere to prevent me from recovering my original file at the end of it.

There are "degrees of lossy", and you might argue that some of them are "near lossless" - but there is only 1 "lossless" - it has a very strict, clear definition.

Cheers,
David.
P.S. thank you to those who gave useful answers to the OP - I'd wavegain before burning a CD and not think anything of the "loss".