Amplitude Compression

Started by bogfrog August 4, 2008
mboigner wrote:
>> Is this what you had in mind for the reference point? > > No. I will try to explain again: > > 1) What is the reference (in the paper) when they speak about dB values? > dB is always relative, for example if you speak from gain of a system your > reference is the input of the system. If we speak from dBmicrovolt (dBu) > the reference is 1uV. I think that they speak from dB_full_scale which > means that their reference is 1 (= full scale). So when they speak from > -46.4dB => 0.0048 linear (if rms). > > 2) I had a short look through the paper and saw that a reference point IS > missing. To print your IO curve you need to know at exactly ONE input power > the output power your system deliveres (or you know the gain, which is > output power-input power)
All audio dB values are by convention relative to 0dB=digital full-scale (0dBFS). Hence all working dB values are negative. The only exception to this is in pro mixing systems where they like to keep some quasi-analog "headroom" available, and define (say) -18dBFS as nominal 0dB. That seems unlikely in this case. The dB values (and ratios) listed in that paper seem more than a little arbitrary - they have no special audio significance that I can identify. I see that they list Winamp in the references; I would not be surprised if they simply ran some audio through a Winamp plugin, fiddled a bit and just read off whatever values the parameters displayed. There is at least one winamp compressor plugin that does offer a dual-knee model so they can, say, boost low sigs as well as reduce high-level ones. It is a relatively small component of the paper as a whole, but IMO they are nevertheless remiss in not giving more comprehensive details of what they used. Even the question of whether they used (if available) a"soft knee" at the transition points is surely relevant to the topic. Richard Dobson
On Tue, 05 Aug 2008 08:23:28 -0500, John O'Flaherty
<quiasmox@yeeha.com> wrote:

>On Tue, 05 Aug 2008 03:20:25 -0500, "bogfrog" <aj00mcgraw@gmail.com> >wrote: > >> >>> Doesn't the paper describing this compression give a mathematical >>>specification of what is meant? >> >> >>Nope, the 3 conditions I listed are all that it gives. It's just one out >>of a list of different signal degradations, simulated to test for >>robustness. >> >>In fact, if you want, you can take a look at the paper here: >> >>http://www.cs.northwestern.edu/~pardo/courses/eecs352/papers/audio%20fingerprint%20-%20haitsma.pdf >> >>Take a look at the beginning of section 4.4 (page 5 of the .pdf). > > I don't think the interpretation you gave is that intended, because >it is such an unlikely thing to happen to a signal as a natural >degradation. That leaves either instantaneous compression as I >described, or intentional gain compression as described in Richard >Dobson's post. I think what is intended was more likely the >instantaneous version, because no attack/decay parameters are given >for a gain shift, and the paper seems to be trying to give a complete >description of what was done. I would also reject a dB interpretation >of the numbers in the table, since they are expressed as ratios, and >dB are already ratios. > In my opinion, they intend a gain curve that rises from a base value >of 1.61 at -oo, changes slope to 1/1.73 = 0.578 at -46.4 dB, and >changes slope again to 1/8.94 = 0.111 at -28.6 dB. That would mean >calculating the net gain by seeing where you are on the curve, as I >described in my post. >For example, > The threshold points in voltage are >-46.4 dB : 0.00478 V >-28.6 dB : 0.03715 V > >For an input sample at -20 dB, the voltage would be +/- 0.1 V. The >degraded sample would have an instantaneous amplitude of >(.00478 * 1.73) + (0.03715 - .00478) * 0.578 + (0.1 - .03715) * 0.111 >= +/- 0.03395 V. > >That's my opinion, at any rate. I see that the authors' email >addresses are in the paper. Since the paper itself is in English, >though they appear to be Dutch, you might consider emailing them to >ask exactly what they meant.
The others who pointed out the need for a reference are correct. The calculations I gave assume dBV, quite possibly an unwarranted assumption. Apart from that, the method may be right. -- John
Thanks for the input, everyone.  I'll take some time to digest what's been

I'm an undergrad student, and even though I've finished implementing the
algorithms in this paper, I am not required to do every single robustness
test, even though I'd like to.  The other stuff is easier (filtering,
resampling, noise, etc), so maybe I'll finish implementing all those first
and come back to this.

Thanks again!