Algebraic minimum-phase variant of sinc function
- KVRAF
- Topic Starter
- 4030 posts since 7 Sep, 2002
Eh... Unfortunately, that 'sinc-filtered dirac spike' after resampling becomes unstable... and beside that it has zeros on the unit circle (it's a low-pass filter). so, it can't be used for deconvolution.
On the other hand, it can be converted into min-phase filter, but this will induce an additional phase shift.. and so, the final response will be definitely longer. maybe there's another way available...
On the other hand, it can be converted into min-phase filter, but this will induce an additional phase shift.. and so, the final response will be definitely longer. maybe there's another way available...
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Christian Schüler Christian Schüler https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=48996
- KVRist
- 266 posts since 23 Nov, 2004 from Hamburg, Germany
Aleksey Vaneev wrote: from this equation you can't say [a0 a1 a2] = [b0 b1 b2], but you should say:
[a0 a1 a2] * H(z) = [b0 b1 b2]
You can't throw H(z) from the equation.
Read my post! The output filtered by the b coeffs equals the input filtered by the A coeffs:
Y * B = X * A
- KVRAF
- Topic Starter
- 4030 posts since 7 Sep, 2002
This equation does not make sense, sorry. (makes sense if Y=H(z) and X = 1, or vice versa.Christian Schüler wrote: Y * B = X * A
Last edited by Aleksey Vaneev on Sat Mar 10, 2007 5:54 am, edited 1 time in total.
- KVRAF
- Topic Starter
- 4030 posts since 7 Sep, 2002
Seems like I've found a way to overcome that 'sinc-filtered dirac spike' unstability. From my previous deductions, filter is unstable in IIR topology if it contains zeros on the unit cycle (zeros in the frequency response) AND discontinuities in phase response. It is obvious that the steep sinc low-pass filter has a discontinuity in its phase response at cutoff point.
So, to solve this problem (and make that 'sinc-filtered dirac spike' usable in the IIR topology) the filter should be converted to a more relaxed one: for example, it should attenuate stop-band by some given value (e.g. 100 dB), and the transition band should be as wide as possible while still providing a good bandwidth and minimum amount of aliasing after resampling.
So, to solve this problem (and make that 'sinc-filtered dirac spike' usable in the IIR topology) the filter should be converted to a more relaxed one: for example, it should attenuate stop-band by some given value (e.g. 100 dB), and the transition band should be as wide as possible while still providing a good bandwidth and minimum amount of aliasing after resampling.
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Christian Schüler Christian Schüler https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=48996
- KVRist
- 266 posts since 23 Nov, 2004 from Hamburg, Germany
I'd say it makes all sense in the world, since H(z) = Y/X = A/B.Aleksey Vaneev wrote: This equation does not make sense, sorry.
Anyway, since there is only so much a biquad filter can do, resampling its coefficients and then truncating to biquad again is probably not optimal. An approximation algorithm (least squares, Chebychef, Remez) could give you a better choice of coeffs to match a filter curve.
Cheers
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- KVRist
- 499 posts since 11 Jul, 2004 from Southern California, USA
I agree.Christian Schüler wrote:I'd say it makes all sense in the world, since H(z) = Y/X = A/B.Aleksey Vaneev wrote: This equation does not make sense, sorry.
- KVRAF
- Topic Starter
- 4030 posts since 7 Sep, 2002
Of course, I agree as well.MackTuesday wrote:I agree.Christian Schüler wrote:I'd say it makes all sense in the world, since H(z) = Y/X = A/B.Aleksey Vaneev wrote: This equation does not make sense, sorry.
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- KVRist
- 499 posts since 11 Jul, 2004 from Southern California, USA
But Y is not being equated to A, nor is X to B. The input signal is already denoted in this equation by the variable 'X'. Y refers to the output signal. The roots of A and B are the zeros and poles of the transfer function, respectively.Aleksey Vaneev wrote:Of course, I agree as well.MackTuesday wrote:I agree.Christian Schüler wrote:I'd say it makes all sense in the world, since H(z) = Y/X = A/B.Aleksey Vaneev wrote: This equation does not make sense, sorry.But what the sense in equating Y to A, and X to B? There's no such entity as "input" or "output" in transfer function - it is a function that should be THEN multiplied by signal... That's why I'm talking about 'no sense'.
- KVRAF
- Topic Starter
- 4030 posts since 7 Sep, 2002
Allright. Probably I was thinking about a different thing...MackTuesday wrote:But Y is not being equated to A, nor is X to B. The input signal is already denoted in this equation by the variable 'X'. Y refers to the output signal. The roots of A and B are the zeros and poles of the transfer function, respectively.
Anyway, this is the same as:
Y = X * (A / B), where A/B = H(z)
which I cannot argue, of course.
(but I do not see a reason to think in these terms, because Y in practice is a result of multiplication of X by H(z). You can't obtain Y other way).
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Christian Schüler Christian Schüler https://www.kvraudio.com/forum/memberlist.php?mode=viewprofile&u=48996
- KVRist
- 266 posts since 23 Nov, 2004 from Hamburg, Germany
- KVRAF
- Topic Starter
- 4030 posts since 7 Sep, 2002
I'm not talking about causality.Christian Schüler wrote:For the theory, the filter does not need to be causal.
So, it is exactly because of IIR filter theory. Basic math operations mangling won't work here.
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- KVRer
- 1 posts since 23 Mar, 2007
The sinc impulse, is a sine, decaying at 1/T.
By superficial comparisons of the regular sinc impulse, and the minimal phase version, a different decay rate would seem sufficient at first guess. Have you tried this?
By superficial comparisons of the regular sinc impulse, and the minimal phase version, a different decay rate would seem sufficient at first guess. Have you tried this?
Starring : *** Luce & Gae ***
- KVRAF
- Topic Starter
- 4030 posts since 7 Sep, 2002
