From: Anthony Airon Oetzmann Anthony.Oetzmann@epost.de To: Peter Olufsen po@dsinet.dk Dato: 6. december 2000 01:49 Emne: Re: Sv: [mad-user] Attenuation, clipping

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1. Is it correct that the clipping appears in the decoder even though the file was encoded "correct" without clipped samples ?

You're thinking lossless compression here, 'cause you won't get a signal different from the original with lossless compression such as Monkeyaudio. However, MP3 reconstructs an audio signal from less than the original signal plus noise. Add for example one frequency x and another frequency 2*x, then you'll have some areas that are boosted, and some that are attenuated. When 2*x is filtered out again(or gets discarded in some places, because the encoding algorythem thought you won't hear it) then some places will get attenuated that were peaks and some will be boosted that weren't peaks before. And sometimes in complex signals that are mastered to use the maximum of the 16 bit resolution on a CD and are mastered to extreme loudness(single releases are mangled this way a LOT), the signal's reconstruction overshoots, because the stuff that got filtered out in the encoding process meant that the tightly controlled frequencies of the signal will have some things missing. You get some peaks and attenuations you previously didn't have. You can check this easily, by taking a ripped track and using an EQ in a sample editor to EQ 100Hz by -6 dB for example. You almost always get clipped samples, because of the great energy that lower frequencies take and thus command more amplitude and heavier changes to the signal.

Ok, i get it !

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3. I think it would be nice to have some form of auto-attenuation (and also auto normalize-to-peak or even amping/compressing), but to make it perfect wouldn't you have to scan all your files first, or could you make some setup for that ?

I have testet some files and they typically have clipping about 1db, but i found one file with 6db clipping in one sample (?), and if you auto-attenuate that the result is not good, so you should have some form of setup to prevent that. I have also tested some auto-amp-plugin and the result wasen't good because you could hear that the sound-level was adjusted when the songs changed in level.

I propose to set a minimum attenuation level of -2 dB. That seems reasonable for most material. Automatic attenuation would set in with a look-ahead algorhythem, and perhaps a good limiter could be used in cases of less than five clipped samples.

I agree.

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4. Would it be possible to use MAD's 24/32-bit output with some DSP-plugin like DFX ?

Or do you loose the advantage from MAD when you use DSP ?

MAD needs to do this itself. DSP plugins get the decoded signal. The signal path is DECODER PLUGIN -> (optional DPS Plugins->) OUTPUT PLUGIN

But it would seem like a good idea that MAD delivered its 32-bit output to DFX (or some other DSP-plugin) instead of MAD going 16->32->16 and DFX doing the same thing, or am i wrong here ?

Peter