What I'd like to do is have C++ source that contains a function
similar to this:

#typedef some_type VECTOR; // arrays of floats muy bueno...
#typedef FREQUENCY unsigned int;
#typedef ERRCODE int; // an enum, really
#define IN
#define OUT

ERRCODE boxcar(IN VECTOR *before, INOUT VECTOR *after, IN FREQUENCY
hipass, IN FREQUENCY lowpass);

where:

- hipass is always < lowpass, and
- "after" must accept an equal number of elements to "before";

The application is pulse shaping of audio data to convert the
shaped pulses into sets of MIDI note on/note off events. I can
do this with COTS audio editors very easily, but I want an
"inline" solution. This will not be in-real-time, it will
be purely batch operation against .wav files on a Windows platform.

Dev tools are not Microsoft. I would rather use MingW.

If it is just a matter of translating the moment-generating function
how does that work?

--
Les Cargill

Les Cargill wrote:
Les Cargill wrote:

What I'd like to do is have C++ source that contains a function
similar to this:

#typedef some_type VECTOR; // arrays of floats muy bueno...
#typedef FREQUENCY unsigned int;
#typedef ERRCODE int; // an enum, really
#define IN
#define OUT

ERRCODE boxcar(IN VECTOR *before, INOUT VECTOR *after, IN FREQUENCY
hipass, IN FREQUENCY lowpass);

where:

- hipass is always < lowpass, and
- "after" must accept an equal number of elements to "before";

The application is pulse shaping of audio data to convert the
shaped pulses into sets of MIDI note on/note off events. I can
do this with COTS audio editors very easily, but I want an
"inline" solution. This will not be in-real-time, it will
be purely batch operation against .wav files on a Windows platform.

Dev tools are not Microsoft. I would rather use MingW.

If it is just a matter of translating the moment-generating function
how does that work?

--
Les Cargill


Well, of course I found... a comp.dsp post ... containing
this...

"... An example of a recursive FIR filter is the "boxcar"
filter, which just averages over a number of samples (y is output, x
is input):

y[n] = 1/N (x[n] + x[n-1] + ... + x[n-(N-1)] ).

Its recursive form is

y[n] = y[n-1] + 1/N (x[n] - x[n-N])."

How does "N" relate to frequency, presumably relative to Fs? Is
this "one end" of the boxcar?


It doesn't, much. An averager (which is what a "boxcar" filter is
understood to be - all coefficients the same) gives you, well, the
average level (amplitude) of that block. Add a sqrt(x[n...]^^2)) and
you get an rms (power) measure of the block. This might be used as an
envelope follower, e.g. in a (musical) compressor algorithm. It is of
course a form of low-pass filter; in the limit, you are left with
whatever DC content the signal has (sum all samples of N periods of a
sinusoid and the result is zero, as the negative values cancel out the
positive ones).

N above does not directly relate to frequency, as such - it signifies
the number of samples in the block (or length of filter) - you sum 1024
samples and divide the result by 1024. However, it implies the amount of
lowpass filtering; clearly a short averager will reveal more HF activity
than a long one. In the limit, a two-sample averager (y[n] = 0.5 * (x[n]
+ x[n-1])) has a frequency response the shape of the first quadrant of a
cosine, i.e. kills any energy at Nyquist, passes DC unchanged. All
introductory dsp textbooks will show how this works.

The recursive form given above eliminates duplicated computations -
rather than add the same samples together each time, you just add the
newest (x[n]) and remove the oldest (x[n-N]).

How this relates to your original question I don't understand. Are you
trying to extract blocks of audio as mini "samples", with an envelope,
to be played via MIDI? Do you want to detect discrete (successive)
pitches in an audio file and separate them out (Melodyne-style)? It
might be easier if you just describe what exactly you want to do, and
others on this list can then tell you how that might be done, and with
what.

Richard Dobson

Richard's suggestion of an envelope sounds like what you're describing
rather than a high and lowpass filter. You will find code for envelope
followers and synth related stuff on musicdsp.org. They're very common
for synths and keyboards to give you an attack-sustain-release type of
thing.

Fully Half Baked wrote:
Richard's suggestion of an envelope sounds like what you're describing
rather than a high and lowpass filter. You will find code for envelope
followers and synth related stuff on musicdsp.org. They're very common
for synths and keyboards to give you an attack-sustain-release type of
thing.


I've tried a few free "drum replacer" plugins, and I think
what I object to about them is that they seem to *be* envelope
followers, with a triggered threshold. When I applied a
low pass/high pass filter on data sets I'm trying to transform,
the results were much "cleaner" - you had nice tall spikes that
would be relatively easy to detect by a state machine.

I know one has a "filter" window, but I didn't spend enough time
with it to actually construct a bandpass filter - the idea for the
boxcar filter was later, thinking about pulse shaping and all
that.

--
Les Cargill

Full stop - I found the mother lode!

1) I needed a Chebyshev filter
2) This gens 'C' source from a wizard.

http://www-users.cs.york.ac.uk/~fisher/mkfilter/trad.html

Les Cargill wrote:
..

Full stop - I found the mother lode!

1) I needed a Chebyshev filter
2) This gens 'C' source from a wizard.

http://www-users.cs.york.ac.uk/~fisher/mkfilter/trad.html


If you are happy to stick to a single sample rate, sure. But if not, and
you find you need to ~design~ the filter as well as simply use one, the
journey is not yet complete.


Richard Dobson

Fully Half Baked wrote:
Richard's suggestion of an envelope sounds like what you're describing
rather than a high and lowpass filter. You will find code for envelope
followers and synth related stuff on musicdsp.org. They're very common
for synths and keyboards to give you an attack-sustain-release type of
thing.

I've tried a few free "drum replacer" plugins, and I think
what I object to about them is that they seem to *be* envelope
followers, with a triggered threshold. When I applied a
low pass/high pass filter on data sets I'm trying to transform,
the results were much "cleaner" - you had nice tall spikes that
would be relatively easy to detect by a state machine.

I know one has a "filter" window, but I didn't spend enough time
with it to actually construct a bandpass filter - the idea for the
boxcar filter was later, thinking about pulse shaping and all
that.

--
Les Cargill

Richard Dobson wrote:
Les Cargill wrote:
..

Full stop - I found the mother lode!

1) I needed a Chebyshev filter
2) This gens 'C' source from a wizard.

http://www-users.cs.york.ac.uk/~fisher/mkfilter/trad.html


If you are happy to stick to a single sample rate, sure. But if not,
and you find you need to ~design~ the filter as well as simply use
one, the journey is not yet complete.


Richard Dobson



Excellent point. I'm not 100% sure whether I will truly need to fully
generalize this or not - I don't think this productizes*, and I value
my dilletante status at this writing I realize, of course, that
what you espouse is The Right Thing to Do.

*might a good open source thing - not sure. It could certainly
be implemented as a VST, but I really just want the thing
for myself.

I also have to get that code working - it isn't as I speak. The output
veers off into nAn land - I stepped on something.

--
Les Cargill

On Nov 1, 6:18 pm, Les Cargill <lcargil... at (no spam) comcast.net> wrote:
Fully Half Baked wrote:
Richard's suggestion of an envelope sounds like what you're describing
rather than a high and lowpass filter. You will find code for envelope
followers and synth related stuff on musicdsp.org. They're very common
for synths and keyboards to give you an attack-sustain-release type of
thing.
I've tried a few free "drum replacer" plugins, and I think
what I object to about them is that they seem to *be* envelope
followers, with a triggered threshold. When I applied a
low pass/high pass filter on data sets I'm trying to transform,
the results were much "cleaner" - you had nice tall spikes that
would be relatively easy to detect by a state machine.

I know one has a "filter" window, but I didn't spend enough time
with it to actually construct a bandpass filter - the idea for the
boxcar filter was later, thinking about pulse shaping and all
that.

--
Les Cargill

Is this something like a beat detector? If so it's not something I've
looked into but I've seen mention of it so I'm pretty sure there will
be known methods if you search around. There are DSP librarys around
that will more than likely have bandpass filters that you can plug
your values into. If you google "DSP library" you will probably find
them.

Les Cargill wrote:
..

Is this something like a beat detector? If so it's not something I've
looked into but I've seen mention of it so I'm pretty sure there will
be known methods if you search around. There are DSP librarys around
that will more than likely have bandpass filters that you can plug
your values into. If you google "DSP library" you will probably find
them.


Yes, it is a lot like that. But it's really more like a
software MIDI trigger.


"Onset detection".

Richard Dobson