I'd like to see DVDdoug defend his comment regarding phase response.

May I assure you that there no practical way to encode an mp3 file better than with a standard phase-preserving quantization of coefficients of a phase-preserving filter bank.

Quote from: pdq on 2010-11-07 17:21:38

There is nothing in the mp3 standard that says that you have to do anything of the kind. If current implementations DO preserve timing and phase then that may be because it is easiest to implement that way. If someone discovered a way to encode to mp3 such that the timing and phase were not preserved, but the result was peceptually transparent at lower bitrates, then everybody would switch to doing it that way.

May I assure you that there no practical way to encode an mp3 file better than with a standard phase-preserving quantization of coefficients of a phase-preserving filter bank.

Quote from: JapanAudio on 2010-11-07 05:37:59

I am not disputing this: To assess or compare lossy compression performance, you NEED to listen to the lossy files because it is difficult to determine what kind of noise may or may not be heard combined with the main track. I am only saying that listening to the discarded noise reveals the exact artifacts that are introduced by the encoder.

So what?  It doesn't tell you whether they are audible in context.

Quote from: Alexey Lukin on 2010-11-07 08:10:09

I disagree!
Mp3 encoding does preserve timing and phase information. Unlike parametric coders, Mp3 strives to preserve the waveform, including its phase.

There is nothing in the mp3 standard that says that you have to do anything of the kind. If current implementations DO preserve timing and phase then that may be because it is easiest to implement that way.

If someone discovered a way to encode to mp3 such that the timing and phase were not preserved, but the result was peceptually transparent at lower bitrates, then everybody would switch to doing it that way.

In fact, the only thing that the mp3 standard specifies is how to decode an mp3 file to wav.

....they're simply dulled out when normally they would sparkle (high pitch) through the audio.

So if you want to know exactly what kind of sparkle is being taken away from your music, i suggest hearing them alone.

Note that it might not give you insight on subjective performance

May I assure you that there no practical way to encode an mp3 file better than with a standard phase-preserving quantization of coefficients of a phase-preserving filter bank.

Quote from: greynol on 2010-11-07 18:37:55

I'd like to see DVDdoug defend his comment regarding phase response.

I'd like to hear some comments from people who have actually taken say a 44/16  400 Hz square wave .wav file, converted it into a 128 kb MP3, and then compared  the reconstructed .wav file to the original wav file.

Quote from: HibyPrime on 2010-11-07 05:43:04

Thats the goal of psychoacoustics, the fact that many people claim to hear the difference between lossless and lossy says that they haven't quite achieved that goal yet - at least at 320Kbps and under.

That's an incorrect generalisation - it's only true for a very few killer samples, and for a few people. In general, modern lossy codecs (MP3 / AAC / OGG) achieve perceptual transparency at much lower bit rates for the vast majority of people and for the vast majority of music. That's what all of the properly conducted tests have revealed, and that's why many people who do their own tests settle on LAME -V2 (around 220 kb/S) or even lower for their MP3 encoding.

Quote from: HibyPrime on 2010-11-07 05:43:04

When I started ripping/downloading all of my music in FLAC, I only did it because I had just a gotten bigger hard drive, and saw no reason to keep ripping in MP3.  I ran ABX tests in foobar and would consistently fail to tell a difference between +192Kbps MP3s and FLAC.  Over time I started to notice things I've never heard before in tracks I've had for a long time, but I would still fail ABX tests.  At this point you would assume that the only difference is all in my head, but when I heard the difference file in the youtube link it sounds a lot like the sounds that I occasionally notice in FLAC files.  Note I'm referring to the lower volume part of the clip, once it hits the louder part it just sounds like a mess.

It is illogical to claim that the difference file on Youtube sounds like the sounds you occasionally notice in FLAC files. If you can't ABX it then you clearly can't notice the difference, so it's all in your head i.e. it's a placebo effect.

Again, the psychoacoustic models in lossy codecs exploit the way your ears and brain perceive sound, and ABX is the only generally available method that measures your perception of the sound. Looking at the difference file, or listening to it in isolation, tells you nothing.

So you are saying that the mp3 standard disallows the use of parametric encoding?

Phase and timing aren't the same thing... Phase cannot be preserved, because you add noise that's reflected in both time and freq domains. But timing doesn't change with respect to this noise.

Its fair to say that phase information is neither totally lost, nor accurately preserved.

Quote from: JapanAudio on 2010-11-07 19:21:16

So if you want to know exactly what kind of sparkle is being taken away from your music, i suggest hearing them alone.

Great, but it doesn't tell you anything because you are ignoring the signal that is masking the "sparkle".

Quote from: JapanAudio on 2010-11-07 19:21:16

Note that it might not give you insight on subjective performance

Finally, you have got it. As has been said numerous times in this and other threads, listening to the difference gives you absolutely no insight into the subjective performance of the lossy codec. Only an ABX test can do that.

May I assure you that there no practical way to encode an mp3 file better than with a standard phase-preserving quantization of coefficients of a phase-preserving filter bank.

Actually, it is a perfectly logical statement.  If the goal of the MP3 encoder is to mask the sounds that you (hopefully) would never hear anyway, it is logical to assume that pin pointing those sounds in an ABX test would be nearly impossible.

when I heard the difference file in the youtube link it sounds a lot like the sounds that I occasionally notice in FLAC files

Not being able to pin point something does NOT mean it doesn't exist.

my argument is that the mathematical difference between a WAV and MP3 file of the same song can help to reveal the differences between lossy and lossless.  I am not arguing that either one is better.

I took first 4096 samples from "testcase.wav" included with LAME distribution, encoded them with lame -b 128, decoded with lame --decode and did FFT on both original and decoded. The phases of resulting matrices are far from identical. So the phase spectrum is not preserved, or am I missing something?

Quote from: pdq on 2010-11-07 19:10:30

So you are saying that the mp3 standard disallows the use of parametric encoding?

Yes, to the best of my knowledge.

I thought that mp3's intensity stereo was a form of parametric encoding?

Quote from: HibyPrime on 2010-11-07 20:16:49

Actually, it is a perfectly logical statement.  If the goal of the MP3 encoder is to mask the sounds that you (hopefully) would never hear anyway, it is logical to assume that pin pointing those sounds in an ABX test would be nearly impossible.

No, your original statement is illogical. You said:

Quote

when I heard the difference file in the youtube link it sounds a lot like the sounds that I occasionally notice in FLAC files

If you noticed those sounds in the FLAC file but not in the MP3 file then the codec would "fail" an ABX test. If it "passes" the ABX test (i.e. you can't detect the difference in a double blind test") then by definition you CAN'T hear those sounds in the FLAC file. Hence your statement is illogical (unless you are deliberately using a lossy codec that creates audible artifacts).

Quote from: HibyPrime on 2010-11-07 20:16:49

Not being able to pin point something does NOT mean it doesn't exist.

In a perceptual listening test it means almost exactly that. If you can't hear the difference, it effectively doesn't exist. The fact that it exists mathematically (when you subtract A from B) is a red herring.

Quote from: HibyPrime on 2010-11-07 20:16:49

my argument is that the mathematical difference between a WAV and MP3 file of the same song can help to reveal the differences between lossy and lossless.  I am not arguing that either one is better.

It can reveal the mathematical differences, but not the audible differences. You are completely ignoring the fact that the noise that is removed is removed precisely because it is inaudible when played in the presence of the rest of the reconstructed signal. If it were audible as part of the original signal then the codec would "fail" an ABX test.

my argument is that the mathematical difference between a WAV and MP3 file of the same song can help to reveal the differences between lossy and lossless.

I think you are misunderstanding what I mean by small differences here.  I'm not talking about a difference where you can just simply skip back 20 seconds in the song and hear it all over again.  I'm talking about the kind of differences that are so small you can't be sure it's actually there.

If we defined reality by what can be easily understood, seen and measured, science would grind to a hault and very little progress would be made.  Just because it is too small to be easily measured, does not mean it can't be heard.  If a computer can 'hear' it with no problems, why is it so hard for you to accept that a person, on some level, can?

Also, you keep saying that the things that are removed are removed because they are inaudible.  That is, again, the goal of applied psychoacoustics and can be proven that it has not been met millions of times over.  As proof to that statement, I think it would be safe to say that anyone in the world with average hearing (and say, 5 years old or older) can hear the difference between a 32kBps MP3 and a 16/44.1 lossless file.  To say that the information removed is inaudible, I assume you were trying to imply that it occurs at higher bit-rates.

At what rate does this occur?

4.  Error signal PCM-320kbps is inaudible, explaining the absence of audible discrepancies described in (2), thus supporting transparency in this instance.
  5.  Error signal PCM-V5 sounds exactly like the noisy 'hush' described in (5), supporting the fact that artifacts are indeed exposed by this method, in this instance.

Quote from: JapanAudio on 2010-11-08 00:52:24

4.  Error signal PCM-320kbps is inaudible, explaining the absence of audible discrepancies described in (2), thus supporting transparency in this instance.
  5.  Error signal PCM-V5 sounds exactly like the noisy 'hush' described in (5), supporting the fact that artifacts are indeed exposed by this method, in this instance.

Yeah this isn't valid reasoning.  You can't conclude #5 (4 is pretty doggy too) because masking exists.  Or to put it another way, lossy audio would be pretty much useless if #5 actually worked.  But MP3s can sound pretty good, so obviously 5 must be wrong

Ok, so here is what I did, hopefully someone more knowledgeable than I will decide if this methodology is sound.

I took a lossless track, encoded it into an mp3 (128k) and opened both files up in audacity.  After lining up the tracks (it appears the timing information isn't changed, when encoded, even on a sample by sample basis) I inverted one, then mixed the two tracks.  I now have a single track consisting of the difference information, correct? Good.  I saved this file as a WAV.

I then opened up the original 128k MP3 in audacity along with the new difference WAV and played them side by side, after lining up as best as possible.  Is it correct in assuming that I should now be listening to the original lossless track + any distortion added by the encoder?  If so, then I can now simply check and uncheck the mute button (on the difference track) during playback and have an instant transition between what is close to lossless and what is lossy.

I haven't yet tried this method at higher than 128k (believe me I will be soon enough), but if correct, this method very audibly reveals the differences at 128Kbps.

Ok, so here is what I did, hopefully someone more knowledgeable than I will decide if this methodology is sound.