I suggest the cubic spline interpolator. It expresses the underlying function
as piecewise trinomial so that the maxima/minima can be computed by solving
binomial equations. It is also known to be close to the ideal sync
interpolation alias-wise.
Xue
From: Paul
The trick is to count the total number of operations, not for each incoming
sample as the window moves.
The algorithm maintains a buffer that pushes at the back and pops at both front
and back. Each sample is pushed onto the buffer and popped out of it exactly
once. If R samples are popped
Just copy an mp3 file from one drive to another and there you have this
perfectly neutral wire.
Don't use itunes though.
-Original Message-
From: Theo Verelst
Sent: Wednesday, April 09, 2014 6:44 AM
To: A discussion list for music-related DSP
Subject: [music-dsp] Does a neutral wire
But, soft-clipping is not going to change periodicity, is it?
So if you soft-clip a sine wave, be it polynomial or not, the outcome is
periodical at the same period, so contains only perfect harmonics. It
cannot behave in the folded alias way one usually suspect.
Xue
On 02/11/2013 06:36,
On 29/10/2013 08:42, robert bristow-johnson wrote:
Rob, i think what Thilo is referring to is the subsample positioning
of an entire grain when it is launched in the PSOLA system. i have
heard this done both ways (grain positioning accurate to subsample
precision vs. accurate only to sample
one paradigm of asymmetric windows is to convolve a symmetric one (like the
Hann) with a filter. So what they actually do is 1) inverse-filter the
sound; 2) PSOLA it with Hann window; 3) filter the outcome. Seems there are
plenty of papers discussing using the linear-predictive filter in 1) and
? and is it better to use asymmetrical windows?
Ross.
On 23/10/2013 2:05 AM, Robert Bielik wrote:
Wen Xue skrev 2013-10-22 16:53:
One issue I find with 2N is that if you downshift by more than one
octave you get gaps between the grains.
Exactly. This is the point :) Otherwise you won't get
One issue I find with 2N is that if you downshift by more than one octave
you get gaps between the grains. In such case I'm thinking you may use
something like 3N or 4N or 5N so that the output grains also have ample
coverage on the time axis. For example if you choose the smallest kN larger
Many thanks Rob.
I'm somewhat puzzled by the grain size being possibly smaller than N (i.e.
2MN), for that means we lose whole pieces of data (every N-2M per N) in
time domain. Maybe I'm slow to see the truth but right now it just doesn't
feel right to me.
Is there some well accepted
Maybe a beginner's question here:
when pitch-synchronized OLA is used to modify speech pitch, do we resample
the original signal or not?
In the traditional view that pitch_shifting = time_scaling + resampling, the
overlapped parts should be resampled, so that the wave shape of each period
literally novelty is something going on now that wasn't before - see now
perfectly that matches the idea of note onsets. ideally a complete onset
detector shall test that 1) a note exists at t1 and 2) it hadn't existed at
t0 to signal a positive between t0 and t1.
for some reason most of
In the old days the text to songs must be verse, as opposed to prose, in
that they needed comply with such things as measure and rhythm, which
would automatically make some sense when adapted to music with the same
matrices. This still governs many songs today, although there's the
tendency of
I think one can trust the compiler to handle a/3.14 as a multiplication. If
it doesn't it'd probably be worse to write a*(1/3.14), for this would be a
division AND a multiplication.
-Original Message-
From: Nigel Redmon
Sent: Saturday, March 09, 2013 5:15 PM
To: A discussion list
On 01/03/2013 20:18, Theo Verelst wrote:
...
- *All* filtering you can do, either analog or digital, will
inevitably have phase shifting as a consequence, no matter what people
will try to tell you about correcting networks (check out the
theory and
preferably do your homework: ALWAYS
Not that I pretend to know much theory -- but I think these filters don't
add up simply because they're not designed to do so. If one wants these
filters to add up he has to patch them in some way. But at this stage it's
already complicated by the uncooperative design.
Some observation on
On 01/03/2013 00:29, robert bristow-johnson wrote:
On 2/28/13 5:44 PM, Wen Xue wrote:
Not that I pretend to know much theory -- but I think these filters
don't add up simply because they're not designed to do so.
Linkwitz-Riley filters *do* add up to an all-pass filter and they are
designed
They have to agree upon some measurement of emotion before making a point at
each other. Maybe a geometric average of blood pressure and heart rate? I
remember someone making a connection between music-induced emotion with
goosebumps on the forearm - he literally counted the bumps!
I think in his serial LP-HP topology you're meant to use your parameter to
control the cut-off frequencies, not the gains. It makes possible to have
the filter be either LP, BP or HP at ANY time. If you use parallel LP-BP-HP
and tune the gains, it's very likely that at some point the filter's
-dsp-boun...@music.columbia.edu] On Behalf Of Wen Xue
Sent: Tuesday, January 22, 2013 4:31 PM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Overlap-add settings for pitch detection?
Hi Danijel,
the choice of window size has much to do with what your pitched source is
like
Hi Danijel,
the choice of window size has much to do with what your pitched source is
like. Usually the window should be long enough to include plenty of cycles
to fight against noise, but short enough to finish before any dramatic pitch
change. So if the pitch is known to be stable, the
Somehow I feel it's the correlated case that deserves more attention. Things
being uncorrelated simply means their correlation coefficients are zero;
but things being correlated these can be anything from -1 to 1 but zero.
You probably don't want to handle all these cases with a same set of
Hi Danijel,
I don't think there's a standard way to it, for people take different
approaches to suit their detectors and the signals. But roughly there are
three schemes:
1. Adapt the pitch detector to multi-channel inputs. In a Bayesian detector,
for example, this is done by replacing the
To only way to guarantee precision is to use enough bits for intermediate
results. Given your running sum formulation, the worst-case quantization
error for any N is
0.5*Pi + 0.25*Pm*(N+2)(N-1)/N
where Pi is the precision of inputs (the summed signals) and Pm is that of
the partial sum
I played it out of my laptop speaker and picked it with my laptop mic.
Surprisingly (or maybe not for some) the second half comes back some 5 times
stronger in partial amplitudes than the first half. I have not observed any
additional harmonic. I assume that shows speaker distortion is not an
It's easy to imagine a 1000hz and a 1200hz generate a 200hz, or a 1200hz and
a 1400hz generate another 200hz, for there is the common divisor. But, say,
will a 500hz and a 900hz generate a 400hz?
xue
-Original Message-
From: Richard Dobson
Sent: Thursday, December 06, 2012 1:27 PM
As far as I could remember, with sampled signals we always try to forget
values *between* samples, for they are always uniquely determined by values
*at* the samples, so if we get these right, those must be right as well.
The problem with mimicking an analogue filter with digital is that the
as the
moving average output is kept below.
Xue
-Original Message-
From: Domagoj Saric
Sent: Wednesday, August 01, 2012 9:39 AM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] DC blocking (again :)
On 31.7.2012. 12:54, Wen Xue wrote:
5ms moving-average doesn't sound very right
: Monday, May 14, 2012 10:36 AM
To: music-dsp@music.columbia.edu
Subject: Re: [music-dsp] Window presum synthesis
Hi, everyone, apologies for the delay...was on a short vacation ;)
On 25 April 2012 02:22, Wen Xue mr.x@gmail.com wrote:
Yes, it's very right that you can't recover a and b from a+b
Time-aliasing is just another formulation of delay-add. If you look at the
definition of convolution y=x*h in terms on y(t)=... then it's clearly
time-aliasing. In your example it's a convolution with a pulse train where
the pulse period is K=N/4, provided the same treatment is applied to all
be
observed, a similar illusion.
-Original Message-
From: Domagoj Šarić
Sent: Monday, April 23, 2012 11:40 AM
To: A discussion list for music-related DSP
Subject: Re: [music-dsp] Window presum synthesis
On 23 April 2012 10:52, Wen Xue mr.x@gmail.com wrote:
Time-aliasing is just another
Pre- and post- windows do not have to be identical.
Post-windowing is more about eliminating discontinuities at the ends of
a frame. It has nothing to do with DFT so one doesn't care about the
spectral qualities.
I use Hamming forpre-windowing and {Hann divided by Hamming} for
The sine window is exactly the square-root of the Hann. (I've seen Hann
window called square-sine window in some textbooks.)
M or M-1 determine the actual window size (time taken by the window function
to go from 0 to 0, not to be confused with DFT size). If you prefer to use
same window and
Filtering a signal x(t) with a LP filter H(z) then subtract the result from
x(t) itself is equivalent to filtering x(t) with a filter 1-H(z), which is
a HP filter only if H(j2*pi*f) is close to 1 in the pass band (i.e. unit
gain and zero phase). Otherwise the result after subtraction will still
Subtracting the LP part makes sense only if the LP filter is zero-phase. I
believe the typical way is to directly construct a series of steep band-pass
filters to cover the whole frequency range. This is very flexible but
usually means the individual parts do not accurately add up to the
Is FramesRecorder(...) a macro or a function call?
And why multiply vol=1.0 when everything is floating-point already?
-Original Message-
From: music-dsp-boun...@music.columbia.edu
[mailto:music-dsp-boun...@music.columbia.edu] On Behalf Of StephaneKyles
Sent: 30 August 2011 12:13
To: 'A
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