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[Patrik T]

Sure, sure Mr Danko. But if that rock peaked lower - for starters - then that rock would surely make a better mp3.

- 3 dB instead of -0.3. Or no limiter instead of a limiter. Or tiny processing instead of endless processing chains.

The "style" of music is secondary - it's how it GETS there. A bad sounding mp3 begins to crap out at the recording stage and no that much at the end (mastering, whatever) stage.

I simply can not justify myself to spend my precious lifetime on finding solutions for making bad sounding music sound good at low digital resolutions. But many others do spend their days that way, hunting codecs, "solutions" and claiming this and that.

Either it sounds acceptable or not. It is mp3!!! I don't spend time trying to enjoy mp3's through my Lavry converter feeding my fairly flat speakers. I play mp3 through consumer things and how it sounds there is all that should matter.


Best Regards
Patrik

[Tomas Danko]

Patrik T wrote on Tue, 20 November 2007 13:37

Sure, sure Mr Danko. But if that rock peaked lower - for starters - then that rock would surely make a better mp3.

I agree, it surely would.

Patrik T wrote on Tue, 20 November 2007 13:37

- 3 dB instead of -0.3. Or no limiter instead of a limiter. Or tiny processing instead of endless processing chains.

Again, it would sound even better indeed.

Patrik T wrote on Tue, 20 November 2007 13:37

I simply can not justify myself to spend my precious lifetime on finding solutions for making bad sounding music sound good at low digital resolutions. But many others do spend their days that way, hunting codecs, "solutions" and claiming this and that.

Once more I agree completely with your point of view. I also just pick a tool and move on with other things.

Patrik T wrote on Tue, 20 November 2007 13:37

Either it sounds acceptable or not. It is mp3!!! I don't spend time trying to enjoy mp3's through my Lavry converter feeding my fairly flat speakers. I play mp3 through consumer things and how it sounds there is all that should matter.

I do the very same thing regarding mp3 files and consumer things.


With that said, looking at what happens inside the lossy codec will show that a classical piece will suffer a whole lot less than a busy heavy metal track. Limiter or no limiter. Regardless of post processing and mangling at the mastering stage, mp3's do lend themselves better to some music than others, by default.

Regards,

Danko

[Patrik T]

So where are the logic and natural conclusions? This topic has extended over four pages already. With attached graphs (whassup with that dB-scale btw?). Will there be four pages more?

Will there be new software companies trying to battle the clipped dragons?

"Smoothness guaranteed!"
"Look here how good we sound!"

Will there be money made?


I guess so.


Cause it all makes the world go around, doesn't it? The fear of sounding "bad".


BR
Patrik

[zmix]

Jeff,
Can you run a separate test to determine if these artifacts (below) are the result of the "Smart Encoding Adjustments" or the "Filter below 10hz" processing?

cerberus wrote on Fri, 02 November 2007 22:03

[cerberus]

hi chuck;

both those options cause some of the distortions. the "smart" causes a greater amount
of "waveform excursions" near the top of the scale.  the distortions appear to become
much more severe if the two options are applied together.

jeff

[cerberus]

Patrik T wrote on Mon, 19 November 2007 22:55

I might add that the CD is peaking around -6 dBFS (classic chamber music).

so you did not wish to make use of the full dynamic range that is possible to reproduce at 16 bits?
or do you think that your reconstructed analog waveform likely rises nearer to zero?

Patrik T wrote on Tue, 20 November 2007 08:37

I simply can not justify myself to spend my precious lifetime on finding solutions for making bad sounding music sound good at low digital resolutions. But many others do spend their days that way, hunting codecs, "solutions" and claiming this and that...

and yet despite all that, which seems well and good; i would say that this thread
seems to have truly captured your imagination.

Patrik T wrote on Tue, 20 November 2007 10:53

So where are the logic and natural conclusions? This topic has extended over four pages already. With attached graphs (whassup with that dB-scale btw?). Will there be four pages more? Will there be new software companies trying to battle the clipped dragons?

wow. i thought some of those were already answered here... however,
i didn't understand your "db scale" question.

jeff dinces

[Patrik T]

cerberus wrote on Wed, 21 November 2007 05:52

i didn't understand your "db scale" question.

Aight.

Well, to my eye it seems your dB scale on the right side of those graphs stretches between...+0.2 and -0.2? Is that around some kind of "ceiling"? Under/over the maximax?

From what I recall, when trying out things, a wav that was limited to -0.6 dB fisted out peaks at around +2 dB when re-imported into a DAW (after encoded to mp3).

On the meters that was. Ear judged things in other ways.

So...logic tells me I'd need something peaking -3 dB (not - 0.3 dB) to ensure there would be no overs in any possible case. Hence the dB-scale question. I mean - why bother measuring things if the scale is not enough to tell what is happening?


Best Regards
Patrik

[cerberus]

the images depict a zoomed-in view of the top of the waveform. zero on that scale
represents fixed point zero; a.k.a. convertus maximus.

those views are zoomed in on the top section of the rendered waveform to show more
information. such as: that the various mp3 encoders, even the fraunhofer ones,
are -not- identical to each other. this does not affect that they all are
clipping it like a lawnmower.

jeff dinces

[Patrik T]

cerberus wrote on Tue, 27 November 2007 08:05

the images depict a zoomed-in view of the top of the waveform. zero on that scale represents fixed point zero; a.k.a. convertus maximus.

Well, that's what I thought.

Without trying to debate things, just try -3 dB instead of -0.3 dB on the limiter, print a wav of it and make it mp3 and then reimport the mp3 and watch the peak meters in your floating point DAW.

It is surprising how much lossy encoding might rumble a DA.



/Patrik