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In Reply to: RE: Some DVD-A not delivering Hi Def Content posted by antoneb on March 27, 2010 at 12:38:28
"a characteristic sound of a lot of digital recordings using FIR filters is a subtle chirp or pre ring?"
The pre-ringing induced by the ADC's anti-aliasing filter (and not the DAC's anti-imaging filter, which can be proven not to ring at all when stimulated with a correctly band-limited signal!) is situated above 20kHz. It is on its own not audible by the majority of people. There are voices that it could be detectable by some: when the pre-ringing extends in time beyond the temporal width of the highest cochlear filter (~400 us), then it might trigger undue desensitization of the inner ear, which would sound as a form of compression. This of course only with pathological test signals, as in normal music there is so much going on simultaneously that an isolated pre-ring has no chance to surface.
"I paper I read on a FIR filtering a square wave at 2khz resulted in a very noticeable chirp, vs a negative feedback filter."
Yes. But that is entirely predictable and hardly a secret.
"Have the filter designers found a way to throw away the negative portion of the impulse response and keep phase linear?"
That would be mathematically impossible.
"My conversation about zero phase filters happened about 4 years ago sky walkers in house acoustician. Who was responsible for all if the modernization of the screening rooms post THX dude. There seemed to be a lot of dis satisfaction with FIR filters at least in his circle. "
This must be seen in its context, which most probably was NOT anti-aliasing and anti-imaging at the edge of the audible band.
FIR pre-ringing is detrimental if it is applied in the audible band. This happens for instance in the bandpass filters used in perceptual coding (i.e. MP3, AC3, DTS, ...) although eventually all induced pre-ringing cancels as the several bands are summed again upon decoding.
Another important case is that of digital loudspeaker crossovers: if FIRs are used then the low-pass and high-pass sections both pre-ring. This, again, is canceled when the signals are acoustically summed. Now this summing is non-perfect as the two speakers are not coincident, and the acoustic paths of the sounds are not identical. In this scenario pre-ringing can become clearly audible.
regards,
Darth
bring back dynamic range
Follow Ups:
That's some good stuff.
Perhaps it was in the context of speaker x-overs that we were discussing. It seems unfortunate since the possible benefit of eliminating electronic phase shift (at the expense of latency) from the speaker x-over would be desirable.
How does one predict the impulse ring range? Looking at an impulse prediction graphs it looks almost like a damped reflected signal except that its symmetric around its peak.
https://ccrma.stanford.edu/~jos/fp/Linear_Phase_Really_Ideal.html
Like in this students paper. I'm just getting back into preparing for engineering so I can't quite follow the equations. Do you know what determines the time and damping of the reflected energies?
I don't know if you can call it reflected but it looks that way from the impulse response (regenerated?). And wouldn't that reflected energy be as broad band as your filtered signal.
Anyhow this is the kind of stuff that interests me. I hope to study more after I finish my Pre Requisites.
Anyhow thanks for that its good stuff!
You can regard the inverse of the width of the transition band as a rough measure of the impulse's width in the time domain. I.e. a short transition band (steep filter) gives rises to long ringing.Best is indeed to grab a tool like Matlan or Scilab (freeware) and have a look at actual impulses. It may be enlightening to rectify the signal and then take 20*log, to put them in the dB world ...
You may want to read this too
http://www.stereophile.com/features/106ringing/index.html
And you may also want to do a search for the presentations of James Johnston ('JJ') of AT&T and DTS.
Good luck with the studies...
Edits: 03/28/10
A problem I see is that there is no standard that specifies how the bits in a file are supposed to be interpolated during playback. This leaves the mastering engineer with no reference playback. It's like the days of LP before the RIAA came along and standardized the equalization.
I've been mastering 44/16 material on the assumption that a linear phase brick wall filter will be used for playback. (I use a minimum phase anti-alias filter that cuts off below Nyquist and has a wide transition band.) This is consistent with most DACs which use on-chip brick wall linear phase filtering. But these recordings sound a bit blurred and soft when played on audiophile DACs that use less aggressive filtering, presumably because this mitigates some of the ringing of brick wall anti-alias filtering common on many CDs. I don't find this a satisfactory solution.
It's clear enough from theory what the solution should be: tradeoffs between ringing, pre-or post, and transient response should be left with the mastering engineer and playback should simply interpolate between the samples with as close an approximation to perfect sinc filtering as is practical. This leaves the artistic control where it is belongs, with the producer of the recording. But that's not the way things seem to be going, at least not with high end gear. Of course, some might consider this to be an advantage, as it gets the audiophile couch potatoes up of their butts to tweak their DAC settings. :-)
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
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