DSPRelated.com
Forums

Perceived aliasing in small ratio SRC

Started by somenoob June 21, 2006
in article xuqdnTxw4f0N3ATZnZ2dnUVZ_tmdnZ2d@giganews.com, somenoob at
sserpy@hotmail.com wrote on 06/21/2006 08:54:

> I'm pretty new to the world of frequency domain as proven by this
question:
> > Is there any correlation between the quality of resampler required to > make
aliasing "imperceptible" and the magnitude of sampling rate change?
> > I'm currently playing with various resampling algorithms, running some
44.1kHz
> content through them (with lots of high and low frequencies) and I
noticed
> increasing the sampling rate change made it easier to hear the
differences in
> quality between the various methods. Does this imply that
as the sampling
> rate change decreases one can get away with a lower
quality algorithm, or that
> I simply got lucky with my given content and
ratios I happened to pick? it depends on how the reconstruction polyphase filter coefficients are computed as a function of the instantaneous time that comes out the phase accumulator (that increments very slowly). if it's asychronous (ASRC), it also depends on how aware the algorithm is of precisely where it is relative to the past and upcoming input and output clock signal edges. -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge."
in article UaydnY_QoscMMwTZnZ2dnUVZ_oidnZ2d@giganews.com, somenoob at
sserpy@hotmail.com wrote on 06/21/2006 16:39:

> Thanks to all for your responses. While I�m trying to learn more about
the
> theory -- I do believe my original question can be answered
independently of
> my test implementation as those details appear to be
misleading people on what
> I�m really after. All I want to know is if
there is any correlation between
> the order of filter required to make
aliasing imperceptible when resampling
> and the magnitude of sampling rate
change. not all FIR coefficient sets will have precisely the same magnitude response nor necessarily have a perfectly phase-linear (constant delay) spectrum for all polyphases or fractional-delays. a higher order FIR filter can be used to make the sliding modulation of frequency response less in effect. -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge."
somenoob wrote:

> Thanks to all for your responses. While I�m trying to learn > more about the theory -- I do believe my original question can > be answered independently of my test implementation as those > details appear to be misleading people on what I�m really after. > All I want to know is if there is any correlation between the > order of filter required to make aliasing imperceptible when > resampling and the magnitude of sampling rate change.
Please be aware that those slanted quote marks below don't display correctly everywhere as they are beyond the ASCII range and your post does not specify a character set.
> It sounds to me from Martin�s comment: �There is no simple > connection to the decimation ratio� that the answer is �no�.
That's not the same thing. If you consider that the transition band of a windowed-sinc filter comes from convolving the ideal sinc response with the window spectrum, you see that in this case its width is indeed independent of the corner frequency, but I'd like to emphasize that this is a property of that particular design procedure. I haven't done much with FIRs at all, but there is certainly a dependence in Parks-McClellan IIR design. Martin -- Quidquid latine scriptum sit, altum viditur.
robert bristow-johnson wrote:
> in article UaydnY_QoscMMwTZnZ2dnUVZ_oidnZ2d@giganews.com, somenoob at > sserpy@hotmail.com wrote on 06/21/2006 16:39: > > > Thanks to all for your responses. While I'm trying to learn more about > the > > theory -- I do believe my original question can be answered > independently of > > my test implementation as those details appear to be > misleading people on what > > I'm really after. All I want to know is if > there is any correlation between > > the order of filter required to make > aliasing imperceptible when resampling > > and the magnitude of sampling rate > change. > > not all FIR coefficient sets will have precisely the same magnitude response > nor necessarily have a perfectly phase-linear (constant delay) spectrum for > all polyphases or fractional-delays. a higher order FIR filter can be used > to make the sliding modulation of frequency response less in effect.
A smaller sample rate change might require more "phases" or fractional delays if the number in the denominator is bigger. Thus a small difference might well be more likely to expose a bad tap in a big table. IMHO. YMMV. -- rhn A.T nicholson d.0.t C-o-M
in article 1150942879.139304.232360@g10g2000cwb.googlegroups.com, Ron N. at
rhnlogic@yahoo.com wrote on 06/21/2006 22:21:

> robert bristow-johnson wrote: >> >> not all FIR coefficient sets will have precisely the same magnitude response >> nor necessarily have a perfectly phase-linear (constant delay) spectrum for >> all polyphases or fractional-delays. a higher order FIR filter can be used >> to make the sliding modulation of frequency response less in effect. > > A smaller sample rate change might require more "phases" or > fractional delays if the number in the denominator is bigger. > Thus a small difference might well be more likely to expose > a bad tap in a big table.
that is an interesting way to think about it. i've usually thunk that, at least for audio, that we would never need to have more 512 phases if you could linearly interpolate between them for fractional delays that are not a multiple of 1/512. but then you want the phase or group delay to be as close to the constant delay for this phase, and you want the magnitude to be a minimally varying gain. -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge."
Martin Eisenberg wrote:

> there is certainly a dependence in
elliptic
> IIR design.
Martin -- Quidquid latine scriptum sit, altum viditur.