From: Patent Tactics, George Brock-Nannestad


there can be no doubt that by digital means it is possible to emulate any 
analog filter function to sufficient precision, provided you define your 
frequency range of interest, use the appropriate sampling rate and the 
appropriate digital filter function. This is rarely done, however, and that 
is why you sometimes have audible differences. The thing you lose in digital, 
unless you choose to model that in as well, is the latency in the dielectric 
in capacitors used in the analog RIAA and similar filters. This means that 
the digital version will be more precise, also on the time axis. Analog RIAA 
networks are usually less specified than what you have to do with digital to 
even make it work. With digital you have to specify, specify, specify, i.e. 
you have to start with a higher knowledge than merely picking an analog 
circuit from a shelf.

from an earlier mail by Tom Fine:

"With my Denon high-output moving-coil cartridge, static discharge can
completely short out the signal, the waveform looks like the needle is 
jumping the groove but I don't think that's what's happening. I think the 
coils are completely saturated by the static charge and take a few 
milliseconds to recover as the charge passes through the cable. Not positive
that's exactly what's happening, but there can be complete loss of signal 
after the pop, with very fast rise time on the pop, almost instant fall-off, 
then a short "blank" period, the resumption of signal. In that case, 
the best one can do is remove the annoying pop, but there's no underlying
audio content to recover so it can appear to the ear as a small dropout 
(small enough that it must be listened carefully to hear, but still 

To me that sounds like simple pre-amp saturation. Many constructions have 
such high loop gain and such heavy feedback, that the amplifier gets very 
slow in getting its precise response. Get a better preamp.

I would also like to mention that a tick reproduced by a velocity-type 
transducer (dynamic, or ELP for that matter) is reproduced by the derivative 
of the little grain that causes the tick, i.e. it is a waveform going sharply 
positive and then sharply negative with a risetime that corresponds to a 
bandwidth of not less than 50 kHz. That speaks for having a high sampling 
rate when working with digital. If you apply RIAA to that before removing the 
tick, it will have become much softer and so more difficult to detect. It is 
a matter of compromise: most prefer to work with sound that sounds realistic, 
so they do not like the sound straight from the pickup linear preamp. But 
then the ticks do not stand out so much any longer.

I shall be away for a few days, so if I do not respond it is not because I am 
dubstruck by any response. However, I am certain that those ARSCLIST readers 
who work professionally with digital programming in audio may present the 
matter much more cogently.

Kind regards,



Andrew Hamilton wrote:
> ---- Clark Johnsen <[log in to unmask]> wrote: 
> > FWIW I pretty much agree with what Tom Fine says...
> > EQ can be fixed in the digital domain if it was done incorrectly, but
> > results won't be as good as if correct equalization had been used prior
> to
> > digitization.
> I wonder in what way the digital de-RIAA would differ from the analog, since
> both apply eq which alters the amplitude and phase.   Perhaps its a matter
> of precision of the group delay.  But, in fairness to the pluggos, out
> there, the filters of digital eq _are_ using the z axis and apply that
> electrician's imaginary constant, j (root-1), in addition to the other
> familiar elements of the electrical syntax of analog filtering, such as pi. 
> Gone also are the days of DSP at mere CD sampling rates and wordlengths.  
> Personally, the analog domain holds more appeal sonically and conceptually. 
> However, I'm willing to accept that double-blind ABX'ing of variously
> "unmastered" material - as in de-emphasized through divers means - could
> result in completely wrong choices.  (L:
> > Ticks and pops must be removed by hand.
> When Masterfonics engineers would transfer from disc to CD, the transfer was
> often "flat," using proper loading of the cartridge.   The un-deemphasized
> audio was easier to de-Click by hand, according to our Webboard host, since
> the clicks  stuck out like a sore thumb, or possibly cactus thorn...  Then
> the Sonic Studio HD 48 bit de-RIAA could be applied to the deClicked audio
> with the illusion of increased results.  It's true that the bandwidth is
> limited in digital audio, but one can get close to vinyl with high sample
> rate filtering, and the resolution of double-precision 24 and 32 bit audio
> is capable of ultra-precise error (l;.
> > It can be an artistic judgment
> > whether they belong in or out, e.g. bows striking music stands. I use
> > iZotope RX 2. On really bad disks automatic click reduction can be used
> on
> > a conservative setting to get rid of the worst half of the noises,
> leaving
> > less manual work.  (On the worst recordings it can take an hour's work
> to
> > clean up a minute of music.)  This software can also fake tape dropouts,
> > sometimes up to 1/4 second. It can even fake glissando and vibrato that
> are
> > missing, but this doesn't always work correctly.
> I tried deClicking my personal copy of All Thing Must Pass.  The task proved
> endless.  I heartily agree about the problem with taking out too much, since
> strange forests can be left of the remaining inaccessible trees.  Talking to
> an eq designer earlier this year, he told me that the problem with capturing
> vinyl isn't the de-emphasis of the cutting eq.   It's that the pops and
> clicks are near square waves, in terms of their rise times, and digital can
> only capture sines.  So, it changes the attack and harmonics of the surface
> noise and makes it ugly, whereas the same noises on the vinyl are easy to
> ignore and might even be exciting>?<   Incidentally, he said this is also
> why digital fails to deliver, musically, even though it does great
> telephony, since the combined harmonic overtones of a full orchestra during
> a fortissimo would approach a square wave on an oscilloscope, but the CD
> can't make that happen.   So, for all its accuracy and quietude, its just a
> stomp box (LPCM).
> Cheers,
>      Andrew
> > On Tue, Apr 10, 2012 at 8:49 AM, Tom Fine
> <[log in to unmask]>wrote:
> > 
> > > Read the article, it was in the ARSC Journal a while back, maybe a year
> or
> > > two. Too technical to summarize here, at least for this non-EE.
> > >
> > > ----- Original Message ----- From: "Andrew Hamilton"
> <[log in to unmask]>
> > > To: <[log in to unmask]>
> > > Sent: Tuesday, April 10, 2012 8:39 AM
> > > Subject: Re: [ARSCLIST] Recording_78rpm_records
> > >
> > >
> > >  Gary has also written for ARSC Journal advocating analog playback and
> EQ
> > >>>>> of
> > >>>>> grooved media, rather
> > >>>>> than "flat playback" and software EQ, and has specified the
> technical
> > >>>>> reasons why analog EQ works
> > >>>>> differently from DSP EQ.
> > >>>>>
> > >>>>
> > >>
> > >>
> > >> What is it about minimum phase digital audio equalization, such as
> that
> > >> found in a de-RIAA or RIAA plugin, that Mr. Galo feels is not up to
> snuff?
> > >>
> > >>
> > >>
> > >>
> > >>
> > >> Andrew
> > >>
> > >>