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Mastermind, use the discussed volume change technique then because the Elephant currently cannot be changed without the unfortunate quality losses.

Andrew, thanks for the tip.  That makes sense.  However, I'm not sure every mastering engineer understands this topic (to speak about a marketing point).


Hi Aleksey,

can you confirm that with use of elephant one cannot get into the trouble of achieving +0dB values after D/A-conversion. if not, maybe it is possible to implement a routine which ensures that +0dB levels are never reached even after D/A conversion.

I think the transparent way elephant works is quite favourable.  However, even if it is technically not possible to just set a ceiling value if you want to keep the quality, maybe elephant could display a guess of the achieved effective output, since the output value you set in elepühant obviously is not the effective output (which is imo a bit confusing)

best, drjee


I would still be interested in the +0dB issue and if elephant prevents this, but I realised that this thread was on V1.1 and since V2 there are brickwall limiting options. (somjetimes it's good to have a look in the manual)

best, drjee


Nobody can ensure limited signal would never go above 0 dbFS.  This depends on the digital-to-analog converter used.  But with Elephant's EL-2, AIGC-3 and AIGC-4 modes with 4x oversampling enabled you can minimize the probability of such overshoots happening.

thanks for this clarification.  I am just wondering then how this RME metering tool indicates that a file is 0dBFS+?  As much as I understood from the Nielsen/Lund Paper (and that's probably not much) it has to do with lowpass filtering applied with D-A conversion. wouldn't it be possibile to "simulate" this and to ztake it into account? or, maybe, we should just assume that D-A converter manufacturers become more aware of this problem.  It seems that some gear-manufactorer already have.

best, drjee


Since actual DAC implementations use different low-pass filters, emulating this is not possible.  Probably RME metering tool emulates filters used on the RME hardware.  But that means this tool is useless if you try to use it with another soundcard/audio playback device.

that's sad, but makes sense. so let's hope that converters improve...

My $0.02.

Old analog amplifiers have a responce curve which tapers off near the top of it's power range, meaning that as you approach or even slightly exceed the rated maximum power you get saturation/compression typical of an overdriven guitar amp, or the various Tube/Valve simulation plugins now available.

Digital amps found it older/cheaper equipment can clip as you get to the maximum output.  The use of filters in D/A converters probably goes along way to prevent this, but it's going to depend on the quality of the equipment.

Basically, old or cheap digital HiFi gear may bawk at heavily loudness maximised material because it was simply never tested with such signal loads.  Newer and/or more expensive hardware should not have this problem.

Having said that, I have heard that there can be problems encoding to MP3 if the signal regularly peaks to 0db.

And as AV said, it's just going to depend on the equipment so there really isn't any way of knowing unless you want to test every possible HiFi/Walkman/MP3 Player...

I guess I'm not saying anything you don't know, and I am no expert so I could be wrong on some points, but maybe someone reading this thread in the future may appreciate the clarification.

I use AIG3/AIG4 and set to -0.02.


My understanding is that if a digital-to-analog converter is doing its job correctly then it will produce greater than 0dBFS analog output if it is presented with digital data that is overly compressed, compressed with a bad algorithm, or otherwise artificial.  The thing that needs to be changed is not the reconstruction filter but for the post d-a converter analog components to be able to handle these higher levels without distortion (or for mastering engineers to stop overcompressing).

my understanding is

a) as long as we stay in the ditigal world: no problem

b) but we can't since out ear can be fed digitally (so far)

c) there will not always be a problem it will only occur under specific circumstances. these are complex. it is, put very simple, a combination of loudnes maximitzing resuklting in hidh floored waveform and d-a converters apllying low-pass filter. this is crutial. no low-pass filter, no problem. the point is that a square wave is composed from sinus waves. if you filter a spqare wave siganl, the result is a different compoisition of these sinus waves and this may result in values obove 0 dB.  Manufacturers of D/A converters could know that (and they know it for sure).  They could also take care of it by providing a headroom of 3 dB (which according to my reading is the highest possible value of 0dBFS+ values). but this will destroy their statiscis, ie they will have a signal to noise ratio lowered by 3 BD then.  Consumers will think this gear is worse.  They will not be aware of the fact that the will have of slightly better signal to noise ration at the cost the danger of distortet signals. is is what I understood.  I might be wrong.

drjee

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