Tue Dec 19 06:34:46 CET 2006

Update of /cvsroot/pure-data/pd/doc/3.audio.examples
In directory sc8-pr-cvs1.sourceforge.net:/tmp/cvs-serv27825

Modified Files:
      Tag: branch-v0-39-2-extended
	A06.frequency.pd E03.octave.divider.pd E07.evenodd.pd 
	E10.complex.FM.pd 
Added Files:
      Tag: branch-v0-39-2-extended
	D13.additive.qlist.pd D14.vibrato.pd F14.wave.packet.pd 
	H01.low-pass.pd H02.high-pass.pd H03.band-pass.pd 
	H04.filter.sweep.pd H05.filter.floyd.pd 
	H06.envelope.follower.pd H07.measure.spectrum.pd 
	H08.heterodyning.pd H09.ssb.modulation.pd H10.measurement.pd 
	H11.shelving.pd H12.peaking.pd H13.butterworth.pd 
	H14.all.pass.pd H15.phaser.pd H16.adsr.filter.qlist.pd 
	I01.Fourier.analysis.pd I02.Hann.window.pd I03.resynthesis.pd 
	I04.noisegate.pd I05.compressor.pd I06.timbre.stamp.pd 
	I07.phase.vocoder.pd I08.pvoc.reverb.pd 
	I09.sheep.from.goats.pd I10.phase.bash.pd J01.even.odd.pd 
	J02.trapezoids.pd J03.pulse.width.mod.pd J04.corners.pd 
	J05.triangle.pd J06.enveloping.pd J07.oversampling.pd 
	J08.classicsynth.pd J09.bandlimited.pd buttercoef3.pd 
	butterworth3~.pd filter-graph1.pd filter-graph2.pd 
Log Message:
added in new patches from HEAD and fixed most bugs listed in  1602617

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#X text 81 461 <-- Q (selectivity);
#X text 88 5 ANOTHER SWEEPING FILTER EXAMPLE;
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#X obj 15 291 *~ 2;
#X obj 15 315 -~;
#X text 119 268 trick to;
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#X text 294 616 updated for Pd version 0.39;
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#X obj 31 362 max 61;
#X text 82 409 smooth & convert to Hz.;
#X obj 47 482 max 3;
#X text 105 483 at least 3;
#X text 11 28 Here's an approximate reconstruction of an old riff by
Pink Floyd. Because we're filtering a waveform with odd partials \,
it's easier to pick out the partials in the filtered sound than if
we had had both even and odd ones.;
#X text 78 527 rejection of the stop bands without having;
#X text 79 509 Put two vcf objects in series for better;
#X text 77 545 to make the passband excessively narrow.;
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--- NEW FILE: J02.trapezoids.pd ---
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#X text 138 30 -- PHASES (percent) --;
#X text 164 180 AMPLITUDES (percent);
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#X text 31 3 Making trapezoidal waves from sawtooth waves;
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#X text 4 476 If the amplitudes sum to zero (with negative ones to
balance positive ones) \, the slope of each linear segment becomes
zero. Otherrwise \, the segments have just enough slope to make up
for the three jumps ane get to the same starting value after each cycle..
;
#X text 4 408 Here we combine three sawtooth waves with controllable
relative phases and amplitudes (in percent \, between -100 and 100.)
Each sawtooth wave gives rise to one jump (upward or downward) per
cycle.;
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--- NEW FILE: H08.heterodyning.pd ---
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#X text 109 12 MORE ON MEASURING SPECTRA: HETERODYNING;
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#X text 56 248 signal to;
#X text 58 268 analyze;
#X text 51 44 Another method for picking out the strengths of partials
in a sound is heterodyning. We guess the frequency of a partial (as
in the previous patch) but this time we multiply by a complex exponential
to frequency-shift the partial down to zero (DC).;
#X text 47 126 Then a low-pass filter (applied separately on the real
and imaginary parts) removes all but the DC component thus obtained.
The result is two audio signals (which we take snapshots of) holding
the real and imaginary parts of the complex amplitude of the partial
we want. Compared to the previous method \, this had the advantage
of reporting the phase of the partial as well as its frequency.;
#X text 240 358 modulate;
#X text 237 394 to DC;
#X text 154 321 <-- test frequency;
#X text 236 376 test frequency;
#X text 132 471 low-pass filter;
#X text 55 596 real;
#X text 59 611 part;
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#X text 105 670 magnitude;
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Index: A06.frequency.pd
===================================================================
RCS file: /cvsroot/pure-data/pd/doc/3.audio.examples/A06.frequency.pd,v
retrieving revision 1.1.1.1
retrieving revision 1.1.1.1.14.1
diff -C2 -d -r1.1.1.1 -r1.1.1.1.14.1
*** A06.frequency.pd	9 May 2003 16:03:50 -0000	1.1.1.1
--- A06.frequency.pd	19 Dec 2006 05:34:43 -0000	1.1.1.1.14.1
***************
*** 59,60 ****
--- 59,61 ----
  #X connect 31 0 1 0;
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+ #X connect 31 0 29 1;

--- NEW FILE: I07.phase.vocoder.pd ---
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#X text 229 417 back up 1/2 window;
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by "location". If "speed" is nonzero \, "location" automatically precesses.
;
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#X text 91 587 "back" window 1/4 cycle behind "front" one;
#X text 137 205 computation period (msec) for overlap of 4;
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#X text 326 275 loop to precess the location according;
#X text 325 291 to the "speed" parameter.;
#X text 611 31 if location changes \, update number box;
#X text 610 50 in main window via "location-set" \, but;
#X text 613 69 taking care to limit frequency of updates.;
#X text 756 462 reflect control changes;
#X text 756 479 in main window.;
#X text 754 344 setting location by hand;
#X text 752 362 sets speed to zero.;
#X text 760 653 misc controls;
#X text 496 527 "rewind" control takes us;
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#X text 272 5 recall previous output amplitude. Its phase will be added
to the phase difference we measure from two windows in the the recorded
sound.;
#X obj 121 69 *~;
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#X obj 141 245 lrshift~ 1;
#X obj 133 269 lrshift~ -1;
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#X text 247 66 divide by the magnitude to make a unit-magnitude complex
amplitude (phase only). The 1e-20 is to prevent overflows. q8_rsqrt~
is reciprocal square root.;
#X text 247 165 Take FT of the window in back. Multiply its conjugate
by the normalized previous output. The result has the magnitude of
the input sound and phase (previous output phase) minus (back window
phase).;
#X text 249 370 Normalize again \, this time taking care to salt each
channel with 1e-15 so that we get a unit complex number even if everything
was zero heretofore.;
#X text 288 427 Now take the FT of the forward window and multiply
it by the unit complex number from above. The magnitude will be that
of the forward window and the phase will be the previous output phase
plus the phase difference between the two analysis windows -- except
that if "lock" is on \, they will be modified to agree progressively
better with the inter-channel phase relationships of the input.;
#X text 249 242 If "lock" is on \, encourage neighboring channels to
stay in phase by adding the two neighboring complex amplitudes. The
result will tend toward the channel with the strongest amplitude. If
the phase relationships between channels in the output and those in
the input are in parallel \, then neighboring channels of the quotient
will all have the same phase and this will not change any phases. (lrshift
shifts the signal to the left or right depending on its argument.)
;
#X text 387 560 'set' message to block;
#X text 390 577 allows variable size;
#X text 259 126 Read two windows \, one 1/4 length behind the other
\, of the input sound \, with Hann window function (see inside).;
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#X restore 55 480 pd fft-analysis;
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#X restore 440 504 pd phase-tables;
#X obj 494 338 s transpo;
#X text 164 364 hundredths;
#X text 493 294 in cents;
#X text 389 359 normal;
#X obj 56 517 output~;
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#X obj 378 214 +~ 0.5;
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#X obj 37 88 r window-size;
#X obj 38 173 /;
#X obj 127 142 samplerate~;
#X obj 38 251 s window-sec;
#X obj 177 204 swap;
#X obj 177 228 /;
#X obj 177 252 s window-hz;
#X obj 49 201 * 1000;
#X obj 49 228 s window-msec;
#X obj 38 115 t f b f;
#X msg 173 92 resize \$1;
#X obj 173 116 s \$0-hann;
#X obj 330 105 r window-hz;
#X msg 382 130 0;
#X obj 330 131 t f b;
#X text 15 8 calculate Hann window table (variable window size) and
constants window-hz (fundamental frequency of analysis) \, window-sec
and window-msec (analysis window size in seconds and msec).;
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#X msg 285 383 \; read-sample ../sound/voice.wav;
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#X obj 276 42 openpanel;
#X obj 276 67 s read-sample;
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#X msg 229 536 ../sound/voice2.wav;
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#X text 460 438 <- record;
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#X obj 55 407 s location;
#X obj 167 407 s speed;
#X obj 262 386 bng 15 250 50 0 empty empty empty 0 -6 0 8 -262144 -1
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#X obj 262 408 s rewind;
#X msg 345 336 200;
#X msg 345 358 100;
#X msg 345 380 20;
#X text 386 335 contract;
#X text 390 380 expand;
#X obj 493 407 s lock;
#X text 494 277 detune;
#X text 55 330 location;
#X text 52 346 (stops;
#X text 57 361 motion);
#X text 165 348 motion in;
#X text 232 464 read input sound;
#X text 103 7 PHASE VOCODER FOR TIME STETCHING AND CONTRACTION;
#X text 604 479 length \, msec;
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#X msg 607 351 2048;
#X msg 607 373 4096;
#X obj 607 395 s window-size;
#X text 607 274 window size \,;
#X text 607 289 samples;
#X text 648 306 <- set;
#X text 100 306 ------- location controls -------;
#X text 660 419 (check);
#X obj 345 407 s auto;
#X text 23 35 This patch takes a sound \, analyzes windows in it both
for channel magnitude and for phase precession in each channel (compared
to another operlapping window). The real-time output recreates the
same magnitudes and phase precession \, althought the phases themselves
are in general different. You can control either the location or its
motion (setting location stops motion \, while setting a non-zero motion
causes the location to change automatically). "Rewind" goes back to
the beginning. You can use different window sizes (use the message
boxes - the number box is for readout). The "lock" feature forces phase
coherency between neighboring channels \, which makes a more present
sound but can add artifacts to the sound. Look in "pd fft-analysis"
to see the workings.;
#X text 483 568 updated for Pd version 0.39;
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#X text 466 458 file ->;
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#X connect 53 0 11 1;

--- NEW FILE: J03.pulse.width.mod.pd ---
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#X text 57 9 CLASSICAL PULSE WIDTH MODULATION;
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#X text 24 314 This patch demonstrates pulse width modulation \, which
is accomplished simply by subtracting two sawtooth waves at a varying
phase difference. Here their frequencies are set to differ by 1/5 Hz.
so that the relative phase wanders continuously.;
#X text 570 457 ---- 0.02 seconds ----;
#X text 524 487 updated for Pd version 0.39;
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#X text 219 60 <-- start/stop graphing;
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--- NEW FILE: H12.peaking.pd ---
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#X text 35 195 This subpatch loads initial;
#X text 31 219 values in number boxes.;
#X msg 22 91 \; \$1-pole 60 \; \$1-zero 20;
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#X text 554 481 gain=0;
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#X text 594 182 5;
#X text 611 492 0;
#X text 599 423 1;
#X text 596 596 updated for Pd version 0.39;
#X text 183 10 PEAKING FILTER;
#X floatatom 406 366 3 0 180 0 - #0-pole -;
#X text 415 328 angle;
#X text 399 344 (degrees);
#X obj 460 435 sin;
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#X text 266 332 pole and zero;
#X text 284 347 radii (%);
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#X text 21 34 To get a peaking filter \, start with a shelving filter
but rotate the pole and zero to the point on the unit circle you want
to amplify or attenuate. The rpole~ and rzero~ filters are replaced
with their complex-valued siblings \, cpole~ and czero~. These filters
take a (real \, imaginary) pair to filter and another (real-imaginary)
pair to specify the pole or zero. As for rpole~ and rzero~ \, the coefficients
may change at audio rate.;
#X text 22 162 The outputs of cpole~ and czero~ are also in the form
of a (real-imaginary) pair. Both outlets of cpole~ are connected to
czero~ in this example \, but then since we want a real-valued filter
\, we only take the real part of the (complex) output of czero~.;
#X text 23 246 Here the pole and zero radii (p and q) control the center-frequency
gain by the formula (1-q)/(1-p). The closer to 1 the radii \, the narrower
the band affected. The non-peak gain \, (1+q)/(1+p) \, is close to
1 as long as p and q are at least 50% or so.;
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#X connect 34 0 29 3;

Index: E10.complex.FM.pd
===================================================================
RCS file: /cvsroot/pure-data/pd/doc/3.audio.examples/E10.complex.FM.pd,v
retrieving revision 1.1
retrieving revision 1.1.14.1
diff -C2 -d -r1.1 -r1.1.14.1
*** E10.complex.FM.pd	23 Sep 2003 00:18:12 -0000	1.1
--- E10.complex.FM.pd	19 Dec 2006 05:34:43 -0000	1.1.14.1
***************
*** 44,48 ****
  #X text 262 212 bang to graph once;
  #X obj 16 494 t b f;
! #X obj 19 295 tabwrite~ E10-signal;
  #X obj 208 295 tabwrite~ E10-spectrum;
  #X text 72 536 set carrier multiplier and modulation multipliers after
--- 44,48 ----
  #X text 262 212 bang to graph once;
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#X obj 450 259 butterworth3~ 80 100000 0;
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#X text 257 102 test sinusoid:;
#X text 272 116 cos;
#X text 325 115 sin;
#X text 397 97 output of filter;
#X text 398 113 we're testing;
#X text 31 103 index and time to next step;
#X text 39 82 ----- from filter-graph1's 3 outlets: -------;
#X text 117 193 low-pass filters;
#X text 118 177 cutoff freq. for;
#X obj 368 360 swap;
#X obj 620 215 t b;
#X text 583 184 clear filters;
#X text 582 198 to start;
#X text 578 452 cbeck if any table;
#X text 577 467 is specified for phase;
#X text 577 483 (don't compute it if;
#X text 578 498 not.);
#X text 31 3 filter-graph2: measures frequency and phase response of
a filter \, which should be driven by a "filter-graph1" object. We
need the three outputs of filter-graph1 \, plus the filter output.
;
#X text 438 55 1: table name for frequency response;
#X text 518 39 creation arguments:;
#X text 438 71 2 (optional): table name for phase response;
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#X text 610 382 updated for Pd version 0.39;
#X text 691 145 frequency;
#X text 631 141 0;
#X text 814 144 44100;
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#X text 35 195 This subpatch loads initial;
#X text 31 219 values in number boxes.;
#X msg 22 91 \; \$1-pole 80;
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#X text 575 127 gain=0;
#X text 574 327 phase=0;
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#X text 44 202 <-- compute;
#X text 34 266 index;
#X text 104 -6 ALL-PASS FILTERS;
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#X obj 346 287 / 100;
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#X text 341 240 pole (%);
#X text 14 20 The all-pass filter has a phase response that depends
on its coefficient \, and a flat frequency response. The coefficient
(p) gives the location of the pole. There is a zero at 1/p \, unless
p=0. If p=0 the filter is effectively a one-sample delay. Negative
values of $p$ are allowed \, as long as p is between -1 and 1;
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--- NEW FILE: H15.phaser.pd ---
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#X text 448 562 updated for Pd version 0.39;
#X text 167 -1 PHASER;
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#X text 147 32 test sound for phaser;
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#X obj 72 501 +~;
#X text 23 17 The phaser ranks \, along with fuzz and wah-wah \, as
one of the great guitar pedals. A phaser simply adds an all-passed
copy of the signal to the original \, making phase reinforcement and
cancellation at frequencies that depend on the all-pass coefficients.
In this example the coefficients range from 0.88 to 0.98 \, controlled
by a phasor~ object (no relation). The phasor~ is converted to a symmetrical
triangle wave (abs($v1-0.5)) and then ranged appropriately.;
#X obj 250 417 phasor~ 0.3;
#X text 22 158 Many variations of this have been invented. A deeper
effect can be obtained by using 12 all-pass filters and adding the
outputs of the 4th \, 8th. and 12th one to the original. Various stereo
configurations are possible. Some people use 6 instead of the 4 stages
used here. Controls can be added to change the frequency of sweeping
and the range of the all-pass coeefficients.;
#X obj 250 449 expr~ 1 - 0.03 - 0.6*abs($v1-0.5)*abs($v1-0.5);
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#X connect 17 0 9 1;
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#X connect 17 0 11 1;

--- NEW FILE: filter-graph1.pd ---
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#X msg 114 122 \; pd dsp 1;
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#X text 166 7 filter-graph1 -- generate sinusoids to test a filter
;
#X text 168 23 arg 1: number of steps - arg2: frequency range;
#X text 40 53 This \, together with its companion filter-graph2 \,
measure a filter's frequency and phase response. Here we count from
0 to n-1 (where n is the table size) and output the index and a complex
sinusoid at each frequency to test.;
#X text 222 192 fudge to estimate settling time;
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#X connect 36 0 34 0;
#X connect 36 0 32 1;

--- NEW FILE: D14.vibrato.pd ---
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#X obj 63 367 sqrt;
#X text 572 461 <-- octave up;
#X msg 460 416 \; trigger 1 60;
#X msg 460 453 \; trigger 1 72;
#X text 550 494 <-- release;
#X text 556 512 is optional;
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#X text 236 438 since we'll multiply \,;
#X text 235 453 vibrato output should;
#X text 235 470 be centered at 1 \, not 0;
#X text 273 384 multiply by vib depth;
#X obj 391 361 / 6923;
#X text 62 425 apply vibrato;
#X text 66 453 fourth;
#X text 69 469 power;
#X text 97 537 waveform;
#X text 96 517 simple;
#X text 457 354 4/(exp(log(2)/1200)-1);
#X text 461 335 conversion factor is;
#X text 384 295 vibrato depth;
#X text 383 312 in cents;
#X text 228 274 vibrato speed;
#X text 227 291 in Hertz;
#X obj 28 392 adsr 0 100 200 100 300;
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#X text 88 9 USING ADSRS FOR PORTAMENTO AND ADDING VIBRATO TOO;
#X text 43 30 Portamento can be treated as a special case of an ADSR
envelope \, with 100 percent sustain. Vibrato is properly computed
in units of pitch \, but it's also possible to do the job without having
to convert from pitch to frequency units at the audio rate. To do this
we just raise the "pitch" to the fourth power \, so that it acts pseudo-exponentially.
Rather than add vibrato to the ADSR output \, we multiply a signal
which controls relative frequency. The relative frequency change is
one plus an oscillator.;
#X text 439 626 updated for Pd version 0.39;
#X text 45 185 The table below holds 6 cycles of vibrato with small
variations to get a not-exactly-repeating vibrato. We thus have to
divide vibrato frequency by six. You can just use a sine or triangle
wave if you prefer.;
#X text 573 426 <-- middle C;
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#X text 35 195 This subpatch loads initial;
#X text 31 219 values in number boxes.;
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#X text 108 34 SHELVING FILTER;
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#X text 616 327 0;
#X text 604 258 1;
#X text 16 58 This patch demonstrates using the raw filters \, rpole~
and rzero~ (raw \, real-valued one-pole and one-zero filters) \, to
make a shelving filter.;
#X text 14 109 If the pole is at p and the zero is at q \, the gain
at DC is (1-q)/(1-p) and the gain at Nyquist is (1+q)/(1+p). If the
pole location is close to plus or minus one \, this can give large
gains unless q is in the same vicinity. (try \, for example \, p=90%
\, q=70%).;
#X text 11 191 The crossover region varies from DC to Nyquist as p
and q decrease from 100% to -100%.;
#X text 278 241 pole;
#X text 334 241 zero;
#X text 383 263 (in hundredths);
#X text 610 387 updated for Pd version 0.39;
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#X text 124 387 <-- cutoff (pitch units);
#X text 135 434 <-- cutoff (Hertz);
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#X text 348 582 updated for Pd version 0.39;
#X text 88 459 low-pass filter;
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#X text 408 528 --- 0.02 sec ---;
#X text 28 30 This and the following patches show how to use filters
in Pd \, starting with the simplest one: the one-pole low-pass filter.
Here we test it with an input of white noise. The lop~ object does
the filtering. Its left inlet takes an audio signal to be filtered
\, and its right inlet takes messages to set its cutoff frequency in
Hertz.;
#X text 26 129 The lop~ object is normalized to pass DC (the lowest
frequency) with a gain of one. Higher frequencies are progressively
more and more attenuated. The lower the cutoff frequency \, the lower
the total power of the filtered noise. If you graph the output you'll
see that the waveform gets smoother (and smaller overall) as the cutoff
frequency is lowered.;
#X text 28 243 At the cutoff frequency the gain is about -3 dB \, and
above that the gain drops a further 6 dB per octave. (Sometimes one
uses the word "rolloff" instead of "cutoff" to emphasize the gradual
way the gain drops off with frequency.);
#X text 108 353 white noise \, test signal;
#X text 185 6 ONE-POLE LOW-PASS FILTER;
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#X text 18 179 boxes.;
#X text 16 161 This subpatch loads initial values in number;
#X msg 84 83 \; \$1-pit 60;
#X connect 0 0 1 0;
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#X connect 2 0 5 0;
#X restore 129 582 pd loadbang;
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--- NEW FILE: H04.filter.sweep.pd ---
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#X text 126 9 SWEEPING FILTERS;
#X obj 44 193 phasor~;
#X obj 59 351 +~;
#X floatatom 81 326 5 0 100 0 - #0-offset -;
#X floatatom 60 222 5 0 0 0 - #0-speed -;
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#X floatatom 75 404 5 0 1000 0 - #0-q -;
#X obj 44 426 vcf~;
#X obj 59 375 tabread4~ mtof;
#X text 127 403 <-- Q (selectivity);
#X text 115 182 sawtooth;
#X text 116 198 oscillator;
#X text 112 221 <-- sweep speed;
#X text 137 245 LFO for sweep;
#X text 134 274 <-- sweep depth;
#X text 131 326 <-- base center frequency;
#X text 103 350 add base to sweep;
#X text 192 375 convert to Hz.;
#X text 97 144 <-- pitch;
#X obj 43 457 output~;
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#X obj 60 244 phasor~;
#X obj 60 298 *~;
#X text 294 496 updated for Pd version 0.39;
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#X text 18 209 This subpatch loads initial values in number;
#X msg 85 83 \; \$1-pitch 48 \; \$1-speed -2 \; \$1-depth 27 \; \$1-offset
56 \; \$1-q 2;
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#X restore 168 491 pd startup;
#X text 14 109 Note the different effects of negative and positive
sweep speeds.;
#X text 13 32 If you want actively changing center frequencies \, use
"vcf~" instead of "bp~". The vcf~ module takes an audio signal to set
center frequency. (Q is still set by messages though.) Vcf is computationally
somewhat more expensive than bp~.;
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#X text 393 465 --- 0.02 sec ---;
#X text 24 31 Many synthesis algorithms and transformations can have
outputs with a zero-freqency component (commonly called DC for "direct
current"). These are inaudible and sometimes cause distortion in audio
output devices \, or when converting to fixed-point soundfile formats.
It is often desirable to filter an audio signal to remove its DC component.
;
#X text 23 147 The simplest way to do this is to use a one-pole low-pass
filter \, tuned to a low frequency such as 3 Hertz \, and to subtract
its output from the original. This difference is called a one-pole
\, one-zero high-pass filter \, and it is used so often that Pd provides
one in the "hip~" object.;
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#X text 88 407 high-pass filter;
#X floatatom 74 366 5 0 0 0 - - -;
#X msg 74 296 0;
#X text 110 245 sinusoidal test signal;
#X text 71 270 add "DC";
#X text 112 296 zero for no filtering;
#X msg 74 319 3;
#X text 109 320 3 (or so) to remove DC;
#X text 114 343 higher freqencies affect;
#X text 154 359 the audible part of;
#X text 154 375 the signal as well.;
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Index: E07.evenodd.pd
===================================================================
RCS file: /cvsroot/pure-data/pd/doc/3.audio.examples/E07.evenodd.pd,v
retrieving revision 1.1
retrieving revision 1.1.14.1
diff -C2 -d -r1.1 -r1.1.14.1
*** E07.evenodd.pd	23 Sep 2003 00:18:12 -0000	1.1
--- E07.evenodd.pd	19 Dec 2006 05:34:43 -0000	1.1.14.1
***************
*** 39,47 ****
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! #X text 16 11 This patch loads a sequence of pitches into array1. The
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--- 39,47 ----
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! #X text 16 11 This patch loads a sequence of pitches into E07. The
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#X text 30 242 sample loop for;
#X text 30 260 test signal;
#X text 35 321 pair of allpass;
#X text 34 338 filters to make;
#X text 34 356 90 degree phase;
#X text 32 373 shifted versions;
#X text 238 323 <-- shift frequency;
#X text 310 356 cosine and sine waves;
#X text 55 7 SINGLE SIDEBAND MODULATION;
#X text 300 7 (AKA FREQUENCY SHIFTING);
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#X text 352 547 updated for Pd version 0.39;
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#X text 123 438 <-- complex multipier;
#X text 122 455 (calculates real part);
#X text 309 371 to form the real and;
#X text 309 387 imaginary part of a;
#X text 309 404 complex sinusoid;
#X text 43 37 The signal sideband modulator gives you only one sideband
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both a positive and negative sideband). You can set the shift frequency
positive to shift all frequencies upward \, or negative to shift them
downwards.;
#X text 42 117 The technique is to filter the input into two versions
\, 90 degrees out of phase \, which can be interpreted as the real
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You can then form the (complex) product of this with a (complex) sinusoid
to modulate upward or downward in frequency.;
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#X text 42 213 The "Hilbert~" object is an abstraction in pd/extra.
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#X text 504 163 real part;
#X text 489 398 imaginary part;
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#X text 94 166 <- frequency;
#X text 133 182 (as multiple;
#X text 135 198 of SR/64 \, the;
#X text 133 215 fundamental);
#X text 170 345 of a cycle;
#X text 431 638 updated for PD version 0.39;
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#X text 127 315 <- phase in;
#X text 161 331 hundredths;
#X text 113 264 <- frequency \, Hz.;
#X text 87 415 given the real and imaginary part;
#X text 88 431 of a complex-valued signal. Here;
#X text 87 447 the imaginary part is zero (the;
#X text 86 400 fft~ computes the Fourier transform \,;
#X text 186 541 real and imaginary;
#X text 186 557 outputs are graphed;
#X text 185 574 separately.;
#X text 86 464 input is real-valued). The output;
#X text 85 482 is a (real \, imaginary) pair for each;
#X text 86 500 frequency from 0 to 63 (in units of;
#X text 87 520 SR/64).;
#X text 145 -36 The "fft~" object has separate inlets for the real
and imaginary parts of a complex-valued signal and outputs its Fourier
transform \, again using separate outlets for the real and imaginary
part. The transform is done on one block of samples (here the block
size is 64 \, Pd's default.) The outputs give the complex amplitudes
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tuned to the fundamental frequency of the analysis \, 1/64th of the
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is two harmonics symmetric about the Nyquist frequency. Fractional
frequencies spill across harmonics. Changing the initial phase rotates
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#X text 26 -24 ANALYSIS;
#X text 27 -42 FOURIER;
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#X text 58 142 audio;
#X text 133 140 lp freq;
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#X text 68 10 3-pole \, 3-zero butterworth lp/hp/shelving filter. Args:
lp freq \, hp freq \, normalize-hi. Inlets: input signal \, lo freq
\, hi freq \, hi norm \, reset.;
#X text 70 75 For high-pass: set LP freq =0 and hi/lo to 1;
#X text 70 56 For low-pass: set HP freq >= SR/2 and hi/lo to 0;
#X text 69 92 Shelving: HP and LP specify shelving band. Gain difference
is about HP/LP cubed (so HP=2LP should give about 18 dB \, for example.)
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#X text 170 541 0 if a neighbor is clean;
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#X text 472 568 0 if all neighbors clean;
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#X text 362 148 normalize the amplitudes;
#X text 439 253 add neighboring amplitude to this one;
#X text 437 269 and take squared magnitude of result -;
#X text 437 286 do this for both the left neightbor and;
#X text 436 303 the right one;
#X text 94 82 forward real Hann-windowed FT;
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#X text 594 366 adjust threshold to quadratic;
#X text 594 382 units and scale;
#X text 142 389 total incoherence;
#X text 496 414 compare incoherence with the threshold;
#X text 532 511 multiply by left and right;
#X text 531 529 neighbors \, so 0 if any of;
#X text 531 546 the 3 is "clean".;
#X text 497 429 If greater (dirty) \, the "clip" outputs;
#X text 498 444 1 \, otherwise (if clean) \, zero.;
#X text 161 583 add to let in channels;
#X text 159 597 for either criterion;
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#X text 354 563 updated for Pd version 0.39;
#X text 11 212 Two separate thresholds may be adjusted to listen to
the "clean" or "dirty" part of the signal. You'll hear anything less
incoherent than the clean threshold \, OR more incoherent than the
dirty one.;
#X text 13 35 This patch applies a very simple coherence test to distinguish
between sinusoids and noise in an input signal. It works very imperfectly
(since noise is random \, no matter what test we place on it it will
sometimes spoof its way in.) Here we just test that neighboring channels
are 180 degrees (pi radians) out of phase \, as they should be in the
main lobe in response to a sinusoid. If any three channels are so arranged
\, all three are considered as contributing to a sinusoid. To do this
we make an "incoherence" measure which is zero if the phase relationship
is perfect and progressively larger otherwise.;
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#X text 119 4 3-pole (or zero) Butterworth filter coefficient calculator
;
#X text 145 109 "theta" in units of pi/2;
#X text 211 138 conjugate pair of pole/zero locations:;
#X text 197 155 real part: (1-r*r)/(1+r*r-2rcos(th));
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#X text 290 5 DENOISER;
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#X text 25 14 This table ($0-mask) is the average power measured in
each channel of the spectrum \, presumed to represent the noise floor.
;
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#X text 80 322 amplitudes (dB);
#X text 68 26 This patch attempts to scrub the noise floor from a sample
in two steps. First using the "make-mask" message (which is caught
in the "fft-analysis" window) \, you estimate the background spectrum.
You would normally do this at a moment when only the background noise
is audible. Then \, turn on "masking" (to 15 by default \, but try
other values) and the patch will try to clean the background noise
out of a signal.;
#X text 67 149 For this demonstration \, you control the amplitudes
of a looping sample and a filtered noise source. Normally you'd hit
"calculate noise mask" with only hte noise turned on \, then turn both
the noise and the sampler on \, and also "masking" \, to see if the
patch can clean the noise out of the signal. Open the "fft-analysis"
window to see the algorithm \, or the "insample" window to change samples
\, or "mask-table" to see the current mask (the average signal power
of the noise to clean out of the signal).;
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#X obj 15 218 rfft~;
#X obj 36 140 tabreceive~ \$0-hann;
#X obj 14 306 *~;
#X obj 56 306 *~;
#X obj 15 356 sqrt~;
#X obj 14 498 tabwrite~ \$0-magnitude;
#X obj 23 386 loadbang;
#X obj 23 470 metro 250;
#X obj 23 449 tgl 15 0 empty empty empty 0 -6 0 8 -262144 -1 -1 1 1
;
#X msg 31 411 \; pd dsp 1;
#X obj 15 8 block~ 512;
#X text 225 131 tabreceive~ outputs array contents \,;
#X text 225 149 constantly \, every block. Here it's;
#X text 223 169 used to get the Hann window to;
#X text 225 187 multiply by the input.;
#X text 120 7 block~ object does no computation but declares this;
#X text 120 24 window to be operating at a different block size from
;
#X text 122 58 Fourier transform.;
#X text 121 40 the parent window. This determines the size of the;
#X text 76 99 The inlet~ automatically re-blocks to the new block size.
;
#X obj 15 332 +~;
#X text 94 308 Take the magnitude by squaring real and imaginary part
\, adding and taking square root.;
#X text 110 424 periodically graph the output. It appears every 512
samples (about 12 milliseconds) but we only update the graph 4 times
per second. The graph is back on the main (parent) window.;
#X text 82 215 forward real FFT. Like "fft~" \, but only one inlet
(for the real part) and only the first half of the output signals are
used. (The others are determined by symmetry: they're complex conjugates
of the first half \, in reverse order.) This takes 1/2 the CPU time
of "fft".;
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#X connect 2 0 4 1;
#X connect 2 1 5 0;
#X connect 2 1 5 1;
#X connect 3 0 0 1;
#X connect 4 0 22 0;
#X connect 5 0 22 1;
#X connect 6 0 7 0;
#X connect 8 0 10 0;
#X connect 8 0 11 0;
#X connect 9 0 7 0;
#X connect 10 0 9 0;
#X connect 22 0 6 0;
#X restore 26 289 pd fft-analysis;
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#X array \$0-magnitude 256 float 0;
#X coords 0 256 255 0 256 100 1;
#X restore 287 208 graph;
#X text 110 6 WINDOWING AND BLOCKING FOURIER TRANSFORMS;
#X obj 25 264 osc~;
#X floatatom 25 218 5 0 0 0 - - -;
#X obj 25 240 * 10;
#X text 305 559 updated for Pd version 0.39;
#X text 349 183 magnitude;
#X text 284 311 0;
#X text 522 311 255;
#X text 273 297 0;
#X text 255 253 128;
#X text 254 203 256;
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#X restore 278 401 graph;
#X msg 156 415 0;
#X obj 50 464 osc~;
#X obj 50 416 samplerate~;
#X obj 50 487 *~ -0.5;
#X obj 50 510 +~ 0.5;
#X obj 42 535 tabwrite~ \$0-hann;
#X text 264 393 1;
#X text 257 511 0;
#X text 273 524 0;
#X obj 50 440 / 512;
#X obj 42 393 bng 15 250 50 0 empty empty empty 0 -6 0 8 -262144 -1
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#X text 321 373 Hann window;
#X text 98 462 period 512;
#X text 40 368 recalculate Hann;
#X text 75 383 window table;
#X text 100 233 tens of Hz.;
#X text 80 215 <- frequency \,;
#X text 98 270 click here and;
#X text 170 286 <- see;
#X text 21 32 In this example we use a sub-patch ("pd fft-analysis")
to re-block the Fourier transform to 512 points. The signal is multiplied
by the Hann window function (which is just a raised cosine.) The magnitude
\, which is computed in the sub-patch \, is graphed below in this window.
The point at 255 corresponds to just below the Nyquist frequency. Phase
isn't shown \, and unlike the previous patch we don't control the initial
phase of the oscillator. (For fun \, try drawing other window functions
with the mouse...);
#X text 459 527 511;
#X connect 3 0 0 0;
#X connect 4 0 5 0;
#X connect 5 0 3 0;
#X connect 14 0 15 1;
#X connect 15 0 17 0;
#X connect 16 0 23 0;
#X connect 17 0 18 0;
#X connect 18 0 19 0;
#X connect 23 0 15 0;
#X connect 24 0 16 0;
#X connect 24 0 14 0;
#X connect 24 0 19 0;

--- NEW FILE: H03.band-pass.pd ---
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#X obj 43 278 mtof;
#X floatatom 43 255 5 0 150 0 - #0-pit -;
#X obj 32 446 output~;
#X obj 32 225 noise~;
#X text 95 254 <-- cutoff (pitch units);
#X text 106 301 <-- cutoff (Hertz);
#X floatatom 43 303 5 0 0 0 - - -;
#X text 330 494 updated for Pd version 0.39;
#X obj 121 414 metro 250;
#X obj 121 394 tgl 15 0 empty empty empty 0 -6 0 8 -262144 -1 -1 0
1;
#X text 139 391 graphing on/off;
#N canvas 0 0 450 300 graph2 0;
#X array H03-graph 882 float 2;
#X coords 0 1 882 -1 200 140 1;
#X restore 375 290 graph;
#X text 399 432 --- 0.02 sec ---;
#X text 98 224 white noise \, test signal;
#X obj 32 361 bp~;
#X text 73 363 band-pass filter;
#X obj 121 439 tabwrite~ H03-graph;
#X floatatom 54 331 5 0 1000 0 - #0-q -;
#X text 106 329 <-- q;
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#X obj 85 16 loadbang;
#X obj 85 40 bng 15 250 50 0 empty empty empty 0 -6 0 8 -262144 -1
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#X obj 85 59 f \$0;
#X text 18 179 boxes.;
#X text 16 161 This subpatch loads initial values in number;
#X msg 85 83 \; \$1-pit 72 \; \$1-q 1;
#X connect 0 0 1 0;
#X connect 1 0 2 0;
#X connect 2 0 5 0;
#X restore 139 482 pd loadbang;
#X text 154 8 RESONANT (BAND-PASS) FILTER;
#X text 26 129 The two controls specify \, first \, the center frequency
\, and second \, the sharpness of the filter \, commonly called "q".
If you increase q to 10 or 20 \, you will see a drop in total signal
power \, and moreover \, you'll see and hear the resonant frequency
more clearly in the result.;
#X text 28 30 A simple resonant band-pass filter is provided in the
bp~ object. Resonant filters can be tuned to a specific "center frequency"
and then will pass that frequency while attenuating other frequencies
(the further from the center frequency \, the more attenuation). This
patch uses a white noise source to demonstrate bp~.;
#X connect 0 0 6 0;
#X connect 1 0 0 0;
#X connect 3 0 14 0;
#X connect 6 0 14 1;
#X connect 8 0 16 0;
#X connect 9 0 8 0;
#X connect 14 0 2 0;
#X connect 14 0 2 1;
#X connect 14 0 16 0;
#X connect 17 0 14 2;

--- NEW FILE: H10.measurement.pd ---
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#X restore 639 200 graph;
#X text 621 56 1;
#X text 633 342 0;
#X text 615 265 pi;
#X text 608 195 2pi;
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#X floatatom 33 249 5 0 0 0 - - -;
#X text 621 -8 2;
#X text 104 -6 MEASURING FILTER FREQUENCY AND PHASE RESPONSE;
#X text 610 382 updated for Pd version 0.39;
#X text 691 145 frequency;
#X text 631 141 0;
#X text 814 144 44100;
#N canvas 876 177 375 255 startup 0;
#X obj 22 24 loadbang;
#X obj 22 48 bng 15 250 50 0 empty empty empty 0 -6 0 8 -262144 -1
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#X obj 22 67 f \$0;
#X text 35 195 This subpatch loads initial;
#X text 31 219 values in number boxes.;
#X msg 22 91 \; \$1-freq 3000 \; \$1-q 3;
#X connect 0 0 1 0;
#X connect 1 0 2 0;
#X connect 2 0 5 0;
#X restore 285 350 pd startup;
#X floatatom 238 257 5 0 10000 0 - #0-freq -;
#X floatatom 249 280 3 0 999 0 - #0-q -;
#X text 12 18 You can use the "filter-graph1" and "filter-graph2" abstractions
as shown to test filters. Connect them as shown with a filter between
them. Try varying the parameters and/or substituting other filters.
;
#X text 575 127 gain=0;
#X text 574 327 phase=0;
#X obj 25 226 filter-graph1 100 44100;
#X obj 227 310 bp~;
#X text 44 202 <-- compute;
#X text 34 266 index;
#X text 290 254 <-- center frequency;
#X text 288 279 <-- "Q";
#X text 9 86 "filter-graph1" takes as arguments the number of points
to graph and the frequency range. "filter-graph2 takes as arguments
the name of a table to hold the (frequency dependent) gain \, and another
\, if specified \, for the phase.;
#X text 8 153 You can edit this patch to replace "bp" with any other
filter you're curious about.;
#X connect 7 0 21 0;
#X connect 16 0 22 1;
#X connect 17 0 22 2;
#X connect 21 0 0 0;
#X connect 21 0 8 0;
#X connect 21 1 0 1;
#X connect 21 1 22 0;
#X connect 21 2 0 2;
#X connect 22 0 0 3;

--- NEW FILE: J07.oversampling.pd ---
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#N canvas 158 4 728 420 16x 0;
#X obj 21 151 *~ 0.064;
#X obj 21 174 rpole~ 0.93538;
#X obj 21 197 *~ 0.00431;
#X obj 21 220 cpole~ 0.96559 0.05592;
#X obj 21 246 cpole~ 0.96559 -0.05592;
#X obj 21 269 *~ 0.125;
#X obj 21 292 rzero~ -1;
#X obj 21 315 rzero~ -1;
#X obj 21 338 rzero~ -1;
#X obj 21 66 phasor~;
#X obj 204 29 block~ 1024 1 16;
#X obj 21 31 inlet;
#X obj 21 372 outlet~;
#X text 170 151 These objects make a 3-pole \, 3-zero Butterwirth low-pass
filter with cutoff at 15kHz (assuming 44100 sample rate.) The filter
was designed using the "buttercoef3" abstraction introduced in patch
H13.butterworth.pd in this series.;
#X connect 0 0 1 0;
#X connect 1 0 2 0;
#X connect 2 0 3 0;
#X connect 3 0 4 0;
#X connect 3 1 4 1;
#X connect 4 0 5 0;
#X connect 5 0 6 0;
#X connect 6 0 7 0;
#X connect 7 0 8 0;
#X connect 8 0 12 0;
#X connect 9 0 0 0;
#X connect 11 0 9 0;
#X restore 23 148 pd 16x;
#X floatatom 23 111 7 0 0 0 - - -;
#X obj 109 149 phasor~;
#X obj 22 194 output~;
#X obj 108 194 output~;
#X obj 23 83 mtof;
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#X text 131 18 UPSAMPLING TO CONTROL FOLDOVER;
#X text 56 57 <-- pitch;
#X text 126 250 not;
#X text 22 265 sampled;
#X text 26 249 16x up-;
#X text 20 293 The "pd 16x" subpatch at left contains a phasor~ object
\, but is locally upsampled by a factor of sixteen. Without upsampling
\, partials as low as 24 Khz. fold back over into the audible range.
With upsampling \, the first audibly folding over partial is at almost
700 Hz \, 29 times higher. The relevant partials will be 29 times \,
or almost 30 dB \, quieter when upsampled.;
#X text 21 403 A third-order Butterworth filter is used inside the
"pd 16x" subpatch - without that \, the internal signal would fold
over as it gets downsampled at the outlet~ object.;
#X text 324 464 Updated for Pd version 0.39;
#X connect 0 0 3 0;
#X connect 0 0 3 1;
#X connect 1 0 0 0;
#X connect 1 0 2 0;
#X connect 2 0 4 0;
#X connect 2 0 4 1;
#X connect 5 0 1 0;
#X connect 6 0 5 0;

--- NEW FILE: I08.pvoc.reverb.pd ---
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#X obj 258 454 *~;
#X obj 356 456 *~;
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#X obj 258 479 -~;
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#X obj 93 84 rfft~;
#X obj 18 451 rifft~;
#X obj 576 334 rsqrt~;
#X obj 293 103 +~;
#X obj 484 383 *~;
#X obj 56 499 sig~ 0.0002;
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#X obj 50 341 outlet~;
#X obj 50 183 -~;
#X obj 50 226 clip~ 0 1;
#X obj 50 204 *~ 1e+20;
#X obj 196 98 inlet~;
#X text 137 213 stronger than;
#X text 139 228 old one;
#X obj 274 202 -~;
#X obj 288 177 lrshift~ 1;
#X obj 274 250 clip~ 0 1;
#X obj 274 228 *~ 1e+20;
#X obj 450 202 -~;
#X obj 450 250 clip~ 0 1;
#X obj 450 228 *~ 1e+20;
#X obj 464 177 lrshift~ -1;
#X obj 50 283 *~;
#X obj 50 312 *~;
#X text 135 199 1 if new signal;
#X text 55 73 new;
#X text 203 70 old;
#X text 51 12 Choose whether to replace the "lod" signal with the "new"
one. The "new" one must be stronger than the old one and also must
be stronger than its two neighboring channels;
#X text 267 283 1 if we're louder than neighbor;
#X connect 0 0 2 0;
#X connect 0 0 9 0;
#X connect 0 0 8 0;
#X connect 0 0 12 0;
#X connect 0 0 15 0;
#X connect 2 0 4 0;
#X connect 3 0 16 0;
#X connect 4 0 3 0;
#X connect 5 0 2 1;
#X connect 8 0 11 0;
#X connect 9 0 8 1;
#X connect 10 0 16 1;
#X connect 11 0 10 0;
#X connect 12 0 14 0;
#X connect 13 0 17 1;
#X connect 14 0 13 0;
#X connect 15 0 12 1;
#X connect 16 0 17 0;
#X connect 17 0 1 0;
#X restore 23 172 pd decision;
#X obj 576 356 *~;
#N canvas 276 481 755 363 divide-by-prev 0;
#X obj 283 99 inlet~;
#X obj 385 101 inlet~;
#X obj 284 249 outlet~;
#X obj 386 249 outlet~;
#X obj 107 251 outlet~;
#X obj 208 253 outlet~;
#X obj 250 180 *~;
#X obj 217 180 *~;
#X obj 182 181 *~;
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#X restore 603 192 pd divide-by-prev;
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#X text 46 28 switch between two pairs of inputs. If first inlet is
one \, take the left-hand pair \, otherwise the right-hand one.;
#X text 15 140 switch;
#X text 92 76 pass this if one;
#X text 269 77 pass this if zero;
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#X text 46 28 switch between two pairs of inputs. If first inlet is
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#X msg 665 293 set \$1;
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#X msg 800 433 set \$1 4;
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#X text 20 206 choose whether to;
#X text 18 224 punch in new (amplitude \,;
#X text 16 243 increment) pair;
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#X text 361 6 previous output amplitude \, encoding both magnitude
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#X text 453 87 previous phase increment (unit-magnitude complex number)
;
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#X text 363 482 propagate amplitudes by multiplying in the;
#X text 361 499 increments \, which advance the phase and drop;
#X text 365 514 magnitude according to revtime.;
#X text 608 370 normalize increments between 0 and;
#X text 606 388 1 according to revtime.;
#X text 78 453 IFFT and output;
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#X text 131 9 PIANO REVERB;
#X text 418 236 reverb time;
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#X text 23 25 This is a phase vocoder acting as a reverberator. The
sound is more coherent (less "whispered") than a real room or a standard
delay-based reverberator.;
#X text 25 80 The technique is to "punch" the incoming sound into channels
where (1) there's a peak \, and (2) the incoming sound drowns out whatever
might already be there. If the sound already in any channel is louder
than the input the input for that channel is ignored.;
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#X restore 68 266 pd test-sound;
#X text 56 217 short tone;
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#X text 24 164 For each window \, the amplitude in each channel is
propagated by a constant phase increment and multiplied downward by
a gain that determines the "reverb time".;
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#X text 15 8 calculate Hann window table (variable window size) and
constants window-hz (fundamental frequency of analysis) \, window-sec
and window-msec (analysis window size in seconds and msec).;
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#X text 307 383 Updated for Pd version 0.39;
#X text 26 389 reverb in;
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--- NEW FILE: J04.corners.pd ---
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#X text 410 208 ---- 0.02 seconds ----;
#X text 354 676 updated for Pd version 0.39;
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#X text 129 26 -- PHASES (percent) --;
#X text 140 267 AMPLITUDES (percent);
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#X text 30 3 Making waveforms with corners using parabolic waves;
#X text 14 499 Here we combine three parabolic waves (in the same way
as \, two patches ago \, we combined sawtooth waves). The parabolic
wave is obtained from the sawtooth wave (assuming it runs from -0.5
to 0.5) by the formula: y=x*x/2 - 1/12. This is normalized so that
the corner has a slope change of minus one unit per cycle \, and adjusted
to remove any DC component.;
#X text 12 593 In general \, the segments of the result will be curved
\, but if the three magnitudes sum algebraicly to zero \, the segments
will be linear.;
#X text 371 67 0.25;
#X text 362 184 -0.25;
#X text 14 644 Note the reduced scale of the graph (from -0.25 to 0.25)
compared to the previous examples.;
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#X text 75 15 BAND-LIMITED SAWTOOTH GENERATOR USING A TRANSITION TABLE
;
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#X text 39 657 Now any time we wish to make a discontinuity in the
output signal \, we make it look exactly like the bandlimited square
wave looks. We do this by reading through the table we recorded \,
carefully adding a "digital" \, non-band-limited \, sawtooth to "array1"
so that the discontinuities in the two cancel out and what you have
left is the transition in the table.;
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#X text 242 138 back the phase up one sample;
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#X msg 13 195 \; pd dsp 1;
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#X text 292 216 twice the table length;
#X text 280 193 period is 2000 samples \,;
#X text 80 369 This one is used - first and third harmonics only.;
#X text 28 644 This alternate one puts in harmonics 1 \, 3 \, and 5
;
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#X text 537 179 ----- 1002 samples ----;
#X text 24 27 This network puts a half cycle of a band-limited square
wave into the table "array1.";
#X text 22 64 Logically the half-cycle is in samples 1 through 1000
\; samples 0 and 1001 are provided so that the 4-point interpolation
will work everywhere.;
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#X text 351 853 updated for Pd version 0.39;
#X text 37 515 A more sophisticated way to control foldover in sawtooth
waves is to replace the once-a-cycle jump with a bandlimited transition.
To get a band-limited transition we synthesize a band-limited square
wave and harvest the transition from the middle of the top half to
the middle of the bottom half. Here we use a square wave at SR/10 \,
so that only partials 1 and 3 fit below the Nyquist. The transition
should take 1/2 period \, or 5 samples. The table is calculated and
stored in the "transition-table" subpatch.;
#X text 40 767 The "band limit" controls how fast the transition table
is read. If it is set to the Nyquist frequency the table is read at
0.4 times the Nyquist \, or five samples a cycle. Lowering the band
limit cuts off the partials of the generated sawtooth wave at frequencies
below the Nyquist.;
#X connect 0 0 24 0;
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#X connect 1 0 22 0;
#X connect 1 0 22 1;
#X connect 1 0 28 0;
#X connect 2 0 5 1;
#X connect 3 0 2 1;
#X connect 4 0 14 0;
#X connect 5 0 6 0;
#X connect 6 0 12 0;
#X connect 7 0 4 0;
#X connect 7 0 3 0;
#X connect 8 0 9 0;
#X connect 9 0 0 0;
#X connect 11 0 25 0;
#X connect 12 0 13 0;
#X connect 13 0 23 0;
#X connect 14 0 5 0;
#X connect 14 0 1 1;
#X connect 16 0 17 0;
#X connect 17 0 7 0;
#X connect 20 0 26 0;
#X connect 23 0 1 0;
#X connect 24 0 2 0;
#X connect 25 0 8 0;
#X connect 26 0 21 0;
#X connect 27 0 28 0;
#X connect 31 0 32 0;
#X connect 32 0 16 0;

--- NEW FILE: H07.measure.spectrum.pd ---
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#X floatatom 145 654 5 0 0 0 - - -;
#X obj 44 565 bp~;
#X obj 44 536 bp~;
#X obj 55 467 mtof;
#X floatatom 55 490 7 0 0 0 - - -;
#X floatatom 98 520 3 0 999 0 - #0-q -;
#X floatatom 55 447 7 0 150 0 - #0-pitch -;
#X obj 145 586 env~ 4096;
#X obj 45 370 *~ 0;
#X obj 44 395 +~ 1;
#X obj 145 608 + 0.5;
#X obj 145 631 int;
#X text 12 41 In this example we use two cascaded bandpass filters
to troll for partials in Jonathan Harvey's famous bell sample.;
#X text 16 233 You can hear partials around 48 \, 51.3 \, 55 (faint!)
\, 57 (fainter!) \, 60 \, two beating partials around 65 \, 67 \, 69
\, 70.9 \, 71.75 \, 72.6 \, 74 \, 74.65 \, 75.6 \, 77 \, 81.2 \, 84.6
\, 86.5 \, and probably many more. There's also one down at 36 \, but
it's easier to see it on the meter than hear it.;
#X text 124 447 <-- center pitch;
#X text 120 463 (shift-drag to fine tune);
#X text 131 491 <-- center frequency;
#X text 138 520 <-- Q (filter selectivity);
#X obj 44 614 output~;
#X text 341 680 updated for Pd version 0.39;
#X text 14 82 Note that filters can give unexpected level changes.
The bp~ object is designed to have roughly unit gain at the pass band
\, so the higher you set "Q" the more amplitude is lost. You can correct
for this by pushing the output amplitude \, but be sure to remember
to reset the output amplitude before you reduce Q again. I set the
Q to 100 and the output amplitude to 110 or 120 (with the room gain
way down.) Then holding the shift key \, slowly drag the center pitch
upward listening for modes.;
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#X msg 59 102 \; readfile symbol \$1-array \; \$1-totsamps 143718 \;
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#X msg 53 361 read -resize ../sound/bell.aiff \$1;
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#X connect 5 0 9 0;
#X connect 6 0 5 0;
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#X coords 0 1 155947 -1 200 150 1;
#X restore 396 322 graph;
#X obj 45 322 r \$0-loopf;
#X obj 45 346 phasor~;
#X obj 44 419 tabread4~ \$0-array;
#X obj 89 370 r \$0-totsamps;
#X text 109 12 MEASURING SPECTRA USING BANDPASS FILTERS;
#X connect 1 0 7 0;
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#X connect 1 0 18 1;
#X connect 2 0 1 0;
#X connect 3 0 4 0;
#X connect 4 0 2 1;
#X connect 4 0 1 1;
#X connect 5 0 2 2;
#X connect 5 0 1 2;
#X connect 6 0 3 0;
#X connect 7 0 10 0;
#X connect 8 0 9 0;
#X connect 9 0 25 0;
#X connect 10 0 11 0;
#X connect 11 0 0 0;
#X connect 23 0 24 0;
#X connect 24 0 8 0;
#X connect 25 0 2 0;
#X connect 26 0 8 1;

Index: E03.octave.divider.pd
===================================================================
RCS file: /cvsroot/pure-data/pd/doc/3.audio.examples/E03.octave.divider.pd,v
retrieving revision 1.2
retrieving revision 1.2.10.1
diff -C2 -d -r1.2 -r1.2.10.1
*** E03.octave.divider.pd	2 Feb 2004 12:13:22 -0000	1.2
--- E03.octave.divider.pd	19 Dec 2006 05:34:43 -0000	1.2.10.1
***************
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--- NEW FILE: F14.wave.packet.pd ---
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#X obj 272 261 mtof;
#X text 267 218 freq.;
#X obj 272 357 *~;
#X text 394 231 bandwidth;
#X obj 397 274 mtof;
#X obj 396 366 *~;
#X text 437 358 divide by fundamental;
#X text 149 220 (MIDI units);
#X obj 136 720 output~;
#X text 91 2 WAVE PACKETS AS ALTERNATIVE TO PAF;
#X obj 79 421 phasor~;
#X obj 471 479 +~ 0.5;
#X obj 471 504 wrap~;
#X text 442 424 second phase signal;
#X text 443 439 out of phase from;
#X text 441 456 first one;
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#X obj 372 621 clip~ -0.5 0.5;
#X obj 396 390 max~ 1;
#X obj 205 285 - 12;
#X text 375 755 updated for Pd version 0.40.;
#X text 20 122 The patch is almost exactly like B13 (the overlapping
sample) except that \, instead of using tabread~ we just use cos~ \,
and that we control pulse width (for bandwidth) as well as wavetable
transposition (for center frequency).;
#X text 18 23 The stretched wavetable method is an alternative to the
PAF generator \, slightly more expensive in processing time but with
two advantages: first \, it is not patent encumbered (PAF patent runs
out in 2011) and second \, it can be generalized to use samples instead
of sinusoids to make complex spectral shapes.;
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--- NEW FILE: I05.compressor.pd ---
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#X obj 137 381 *~ 0.00065;
#X obj 137 225 +~ 1e-20;
#X obj 136 262 q8_rsqrt~;
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#X text 31 5 As in the previous patch \, this works by multiplying
each channel of the Fourier analysis by a real number computed from
the magnitude. If the magnutude is "m" \, the correction factor is
1/m \, but only to an upper limit controlled by the "squelch" parameter.
;
#X text 211 174 squared magnitude;
#X text 219 225 protect against divide-by-zero;
#X text 223 261 quick 8-bit-accurate reciprocal square root;
#X text 222 277 (done by table lookup - about 0.25% accurate);
#X text 193 351 limit the gain to squelch*squelch/100;
#X text 238 381 normalize for 1024-point \, overlap-4 Hann;
#X text 151 409 multiply gain by real and complex part;
#X text 152 429 of the amplitude;
#X text 130 137 outputs complex amplitudes;
#X msg 461 139 \; pd dsp 1 \; window-size 1024 \; squelch 10 \; squelch-set
set 10;
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#X text 21 28 test signal: looped sample playback;
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#X restore 223 313 pd insample;
#X text 362 406 updated for Pd version 0.39;
#X text 56 43 Here we divide each complex channel in the Fourier analysis
by its own magnitude to "flatten" the spectrum. The "squelch" control
limits the amplitude boost the algorithm will apply. If infinite \,
you'll get a white spectrum. If less \, the louder parts of the spectrum
will be flattened but the quieter ones will only be boosted by the
squelch value.;
#X text 73 6 DYNAMIC RANGE COMPRESSION BY FOURIER ANALYSIS CHANNEL
;
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#X text 62 281 <- record;
#X text 276 365 sample length \, msec;
#X msg 292 183 ../sound/bell.aiff;
#X msg 292 208 ../sound/voice.wav;
#X msg 292 233 ../sound/voice2.wav;
#X text 91 197 <- squelch;
#X text 295 161 change input sound;
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#X obj 177 204 swap;
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#X obj 49 201 * 1000;
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#X text 15 8 calculate Hann window table (variable window size) and
constants window-hz (fundamental frequency of analysis) \, window-sec
and window-msec (analysis window size in seconds and msec).;
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--- NEW FILE: J06.enveloping.pd ---
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#X obj 47 52 phasor~;
#X text 126 2 ENVELOPE GENERATORS FROM LINE SEGMENTS;
#X obj 19 514 output~;
#X text 610 698 updated for Pd version 0.39;
#X obj 46 98 *~;
#X obj 11 165 -~;
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#X text 639 514 ----- 0.02 second ----;
#X text 86 24 <-- frequency (Hz.);
#X text 636 322 ----- 0.5 second ------;
#X text 107 72 <-- slope of rise segment. Just multiply this by the
phase to make the segment.;
#X text 129 140 Subtract this to make the phasor cross zero at the
desired point of the cycle.;
#X text 107 173 <-- slope of decay segment.;
#X text 112 190 multiply the phasor (with the zero crossing shifted
as above) by the desired slope \, negating it so the segment points
downward.;
#X text 63 244 minimum of rise and decay segments (makes a triangle
wave);
#X obj 17 267 clip~ 0 1;
#X text 109 266 clip the triangle wave to between 0 and 1 \, to make
the sustain and silent regions.;
#X text 108 121 <-- Duty cycle (end of decay segment as % of cycle.)
;
#X text 60 304 <-- click to graph envelope shape;
#X text 91 456 <-- click to graph audio waveform;
#X text 172 364 this makes a quick-and-dirty triangle wave;
#X text 172 382 as described in the previous patch. It's;
#X text 172 419 to listen to.;
#X text 97 511 You can make a phasor-generated envelope generator using
"min" and "clip" to combine line segments. Here a rise segment starts
at phase 0 \, and a decay segment passes through zero at a controllable
point (the "duty cycle" \, as a percentage of a cycle.) Each has a
controllable slope (in units per cycle). The resulting triangle wave
(the minimum of the rise and decay segments) is limited to 1 \, thus
making a flat "sustain" segment (unless the rise and decay segments
meet at a value less than one \, in which case there is none). Limiting
below by 0 prevents us from following the decay segment into negative
values. Reasonable values to start with are 6 Hz. frequency \, rise
and decay slope 10 \, duty cycle 75%.;
#X text 173 401 used here so we'll have something;
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#X text 35 195 This subpatch loads initial;
#X text 31 219 values in number boxes.;
#X msg 22 91 \; \$1-lf 80 \; \$1-hf 150 \;;
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#X text 553 359 gain=0;
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#X text 610 370 0;
#X text 598 301 1;
#X text 575 435 updated for Pd version 0.39;
#X text 186 -4 BUTTERWORTH FILTER;
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#X text 232 318 poles;
#X text 288 318 zeros;
#X text 24 20 The butterworth filter can be configured for low-pass
\, high-pass \, and shelving \, depending on the placement of the poles
and zeros. For low-pass \, the poles are placed to set the cutoff frequency
and the zeros are at -1 (the Nyquist). Leaving the poles fixed and
moving the zeros then gives shelving filters. In this example \, the
actual filtering is relegated to an abstraction (butterworth3~) which
takes frequencies corresponding to the pole and zero placement.;
#X text 24 147 The butterworth3~ abstraction computes filter coeffients
using control messages \, and so it is not suitable for continuously
time-varying Butterworth filters. For that \, it is often appropriate
to use time-saving approximations \, but precisely which approximations
to use will depend on the way the filter is to be used.;
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#X obj 44 255 output~;
#X text 333 501 Updated for Pd version 0.39;
#X text 151 7 THE CLASSIC SUBTRACTIVE SYNTH SOUND;
#X obj 152 132 *~;
#X obj 151 102 +~ 0.2;
#X obj 151 156 *~ 2000;
#X obj 108 221 *~;
#X obj 43 218 *~;
#X obj 41 122 mtof;
#X obj 41 13 r \$0-note;
#X obj 41 62 makenote 1;
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#X obj 42 192 vcf~ 3;
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#X text 31 323 Now that we can make reasonably high-quality classic
waveforms using upsampling \, we combine an upsampled oscillator with
a "vcf" filter and ADSR generators to control the filter resonant frequency
and the amplitude to make the classic subtractive synthesis sound.
Send an "s \$0-note" object a (pitch \, duration) pair to play a note.
(Classic VC synths did not have velocity sensitive keyboards!) You
can add controls to change the parameters of the ADSR envelopes and/or
the vcf~ "Q" parameter. THe oscillators' waveforms and tuning relationship
is controlled by other parameters set within the "pd 16x" window.;
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--- NEW FILE: I06.timbre.stamp.pd ---
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#X text 458 425 of control;
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#X text 115 159 filter input;
#X text 438 282 control source;
#X text 434 332 Fourier transform;
#X text 28 17 Internal workings of the timbre stamping algorithm. First
the "filter input" is treated as in the compressor patch \, multiplying
each channel amplitude by one over its modulus (but limited by the
"squelch" parameter.) It is then multiplied by the modulus of the channel
amplitude for the control source (which is Fourier analyzed in parallel
with the filter input.);
#X text 145 422 multiply the two amplitude;
#X text 143 439 factors (for compression;
#X text 145 455 and to apply new timbre);
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#X text 137 12 CORT&ZACK's SECRET;
#X text 27 422 filter;
#X text 29 437 input;
#X text 232 441 source;
#X text 233 422 control;
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#X msg 157 278 ../sound/bell.aiff;
#X msg 157 303 ../sound/voice.wav;
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#X text 255 231 change input sounds;
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#X text 21 28 test signal: looped sample playback;
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#X restore 87 415 pd test-signals;
#X text 104 393 <- record ->;
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#X text 15 8 calculate Hann window table (variable window size) and
constants window-hz (fundamental frequency of analysis) \, window-sec
and window-msec (analysis window size in seconds and msec).;
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#X restore 334 455 pd hann-window;
#X text 509 412 sample lengths \,;
#X text 510 427 msec;
#X text 27 35 This is a Fourier-based "vocoder" (perhaps better called
a "timbre stamp") like the one the Convolution brothers use. The "control
source" is analyzed to get its spectral envelope \, which is then stamped
onto the "filter input" by adjusting the amplitudes of its Fourier
transform. The "filter input" is first whitened by the compression
algorithm from the previous patch in this series. The best value of
"squelch" to use depends critically on what kind of sounds are used
for the filter input and the control source.;
#X text 402 498 updated for Pd version 0.39;
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--- NEW FILE: J01.even.odd.pd ---
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#X text 570 475 ---- 0.02 seconds ----;
#X text 528 567 updated for Pd version 0.39;
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#X text 41 -1 Splitting a sawtooth wave into even and odd harmonics
;
#X obj 24 29 phasor~ 100;
#X text 87 58 remove DC bias;
#X text 132 29 original sawtooth;
#X text 144 173 180-degree-out-of-phase;
#X text 147 188 sawtooth;
#X text 145 212 form the sum and difference;
#X obj 23 224 +~;
#X obj 59 223 -~;
#X text 4 408 This patch splits a sawtooth wave into its even and odd
harmonics. The wrap~ object is used to make the phased copy. Adding
and subtracting this to and from the original gives the results shown
and heard. (Listen to the two outputs separately \, then together.)
;
#X text 102 291 output level;
#X text 93 367 for sum;
#X text 95 350 output level;
#X text 100 308 for difference;
#X text 157 77 <-- click to graph;
#X msg 148 97 \; pd dsp 1;
#X obj 138 247 tabwrite~ \$0-difference;
#X obj 138 270 tabwrite~ \$0-sum;
#X obj 138 138 tabwrite~ \$0-phasor;
#X text 4 491 This is a classic technique for gaining separate control
over the even and odd harmonics in a synthetic sound. It can also be
used conceptually to understand the harmonic content of a square wave
in terms of that of a sawtooth \, or vice versa.;
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--- NEW FILE: I03.resynthesis.pd ---
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#X text 85 88 The inlet~ now re-uses 3/4 of the previous block \, along
with the 128 new samples.;
#X text 221 141 window function as before.;
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#X obj 77 283 /~ 768;
#X text 98 301 divide by 3N/2 (factor of N because rfft and rifft aren't
normalized \, and 3/2 is the gain of overlap-4 reconstruction when
Hann window function is applied twice.);
#X text 120 216 Just to show we're doing something \, we multiply each
channel by a gain controlled by an array in the main window. The control
is quartic-scaled for easy editing.;
#X obj 78 251 *~;
#X text 92 357 Multiply the (complex-valued) spectrum amplitudes by
the (real-valued) gain-and-normalization-factor;
#X obj 15 399 rifft~;
#X text 89 396 Real-valued inverse Fourier transform. This uses only
the first N/@ points of its inputs \, supplying the rest by symmerty
(so it's OK that rfft~ obly puts out those N/2 points.) There's only
one outlet because the output is real-valued.;
#X obj 16 566 outlet~;
#X text 88 499 Multiply by the Hann window function again \, necessary
because the operation we performed might result in a signal that doesn't
go smoothly to zero at both ends.;
#X text 89 566 This repackages the output into 64-sample chunks for
the parent window. Since we're operating with an overlap \, the outlet~
object performs an overlapped sum of the blocks computed in this window.
;
#X text 129 8 block~ object specifies vector size of 512 and overlap
four. This window now computes blocks of 512 samples at intervals of
128 samples computed on the parent patch.;
#X connect 0 0 2 0;
#X connect 1 0 0 0;
#X connect 2 0 4 0;
#X connect 2 1 5 0;
#X connect 3 0 0 1;
#X connect 4 0 18 0;
#X connect 5 0 18 1;
#X connect 9 0 10 0;
#X connect 9 0 10 1;
#X connect 10 0 16 0;
#X connect 10 0 16 1;
#X connect 11 0 20 0;
#X connect 12 0 11 1;
#X connect 13 0 4 1;
#X connect 13 0 5 1;
#X connect 16 0 13 0;
#X connect 18 0 11 0;
#X restore 26 289 pd fft-analysis;
#X text 290 362 updated for Pd version 0.39;
#N canvas 35 66 592 433 Hann-window 0;
#N canvas 0 0 450 300 graph1 0;
#X array \$0-hann 512 float 0;
#X coords 0 1 511 0 200 120 1;
#X restore 293 249 graph;
#X msg 171 263 0;
#X obj 65 312 osc~;
#X obj 65 264 samplerate~;
#X obj 65 335 *~ -0.5;
#X obj 65 358 +~ 0.5;
#X obj 57 383 tabwrite~ \$0-hann;
#X text 279 241 1;
#X text 272 359 0;
#X text 288 372 0;
#X obj 65 288 / 512;
#X obj 57 241 bng 15 250 50 0 empty empty empty 0 -6 0 8 -262144 -1
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#X text 336 221 Hann window;
#X text 113 310 period 512;
#X text 90 215 recalculate Hann;
#X text 125 230 window table;
#X obj 57 146 loadbang;
#X msg 79 179 \; pd dsp 1;
#X text 40 27 The Hann window is now recomputed on 'loadbang' to make
the file smaller (it doesn't have to be saved with the array.);
#X text 474 375 511;
#X connect 1 0 2 1;
#X connect 2 0 4 0;
#X connect 3 0 10 0;
#X connect 4 0 5 0;
#X connect 5 0 6 0;
#X connect 10 0 2 0;
#X connect 11 0 3 0;
#X connect 11 0 1 0;
#X connect 11 0 6 0;
#X connect 16 0 11 0;
#X connect 16 0 17 0;
#X restore 192 318 pd Hann-window;
#X obj 27 323 output~;
#X obj 25 264 noise~;
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#X msg 192 264 const 0;
#X obj 192 293 s \$0-gain;
#X text 138 0 FOURIER RESYNTHESIS;
#X text 6 218 0;
#X text 6 159 1;
#X text 19 228 0;
#X text 516 231 22K;
#X text 270 261 <- reset gain;
#X text 224 148 GAIN;
#X text 21 24 Using Fourier resynthesis you can take an incoming sound
\, operate on its spectrum \, and hear the result. Here we start with
white noise and apply a frequency-dependent gain \, which works as
a graphic equalizer. There are N/2 = 256 points \, each spaced SR/512
Hz. apart (although their frequency ranges overlap). Open the "fft-analysis"
patch to see the workings.;
#X connect 0 0 3 0;
#X connect 0 0 3 1;
#X connect 4 0 0 0;
#X connect 6 0 7 0;

--- NEW FILE: H16.adsr.filter.qlist.pd ---
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#X obj 12 219 r trigger;
#X obj 12 437 *~;
#X obj 12 330 *~ 0.01;
#X obj 12 365 *~;
#X obj 12 395 *~;
#X obj 59 359 r pitch;
#X obj 59 409 mtof;
#X floatatom 59 384 4 0 0 0 - - -;
#X floatatom 36 271 4 0 0 0 - - -;
#X obj 36 246 r level;
#X floatatom 110 271 4 0 0 0 - - -;
#X obj 110 246 r attack;
#X floatatom 195 271 4 0 0 0 - - -;
#X obj 195 246 r decay;
#X floatatom 270 271 4 0 0 0 - - -;
#X floatatom 364 271 4 0 0 0 - - -;
#X obj 270 246 r sustain;
#X obj 364 246 r release;
#X obj 499 158 r note;
#X msg 500 236 \; trigger 1;
#X obj 602 225 del;
#X msg 602 247 \; trigger 0;
#X obj 14 166 qlist;
#X obj 14 7 r qlist;
#X msg 35 34 bang;
#X msg 35 59 rewind;
#X obj 42 88 r tempo;
#X floatatom 42 113 4 0 0 0 - - -;
#X msg 42 138 tempo \$1;
#X obj 499 201 t b f;
#X obj 550 198 s pitch;
#X obj 624 176 r duration;
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#X floatatom 499 181 4 0 0 0 - - -;
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#X obj 542 380 r sustain2;
#X obj 638 380 r release2;
#X obj 59 434 tabosc4~ array1;
#X floatatom 218 365 4 0 0 0 - - -;
#X obj 12 481 vcf~;
#X floatatom 119 487 4 0 0 0 - - -;
#X obj 119 462 r q;
#X obj 12 305 adsr 0 0 0 0 0;
#X obj 268 443 adsr 0 0 0 0 0;
#X obj 294 400 / 69.23;
#X obj 218 390 mtof;
#X obj 218 415 sqrt;
#X obj 218 440 sqrt;
#X obj 176 335 r filter;
#X obj 219 493 *~;
#X obj 219 518 *~;
#X obj 268 468 +~ 1;
#X obj 218 465 *~;
#X text 118 214 ADSR for amplitude:;
#N canvas 0 258 703 380 otherstuff 0;
#X obj 289 86 loadbang;
#X obj 418 85 loadbang;
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#X msg 418 115 \; qlist read qlist2.txt;
#X msg 289 111 \; level 100 \; attack 20 \; decay 300 \; sustain 70
\; release 300 \; duration 300 \; pitch 72 \; filter 38 \; level2 49
\; attack2 19 \; decay2 300 \; sustain2 17 \; release2 700 \; q 3 \;
tempo 4;
#X connect 0 0 4 0;
#X connect 1 0 3 0;
#X restore 134 560 pd otherstuff;
#X text 87 33 <--start loop;
#X text 104 61 <--stop loop;
#X text 90 113 <--set tempo;
#X text 257 562 <--loadbangs and table;
#X msg 447 517 \; qlist read qlist2.txt;
#X text 441 493 click to reload qlist2.txt;
#X obj 12 509 output~;
#X text 229 19 This is an analog-synth sound made using a wavetable
oscillator and a "vcf~' object. Unkike the "floyd" example earlier
\, we use a qlist object to do the sequencing. This can also be adapted
to make a keyboard synth.;
#X text 227 85 The qlist reads the file \, "qlist2.txt" \, which contains
four "note" messages and a message at the end that restarts the qlist
at the beginning. The "note" messages are translated into a pitch change
and triggers for the ADSRs.;
#X text 667 551 updated for Pd version 0.39;
#X text 379 305 ADSR for filter. Here \, it works better to make the
envelope modify a constant "filter pitch"--so the "filter" receive
gets the "mtof" treatment and the ADSR is an offset in halftones.;
#X text 231 1 ANALOG_STYLE SYNTH USING QLIST;
#X connect 0 0 50 0;
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#X connect 2 0 3 0;
#X connect 2 0 3 1;
#X connect 3 0 4 0;
#X connect 3 0 4 1;
#X connect 4 0 1 0;
#X connect 5 0 7 0;
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#X connect 7 0 6 0;
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#X connect 10 0 50 2;
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--- NEW FILE: D13.additive.qlist.pd ---
#N canvas 233 179 667 449 12;
#X obj 16 182 osc-voice amp1 pit1;
#X obj 16 206 osc-voice amp2 pit2;
#X obj 16 230 osc-voice amp3 pit3;
#X obj 16 254 osc-voice amp4 pit4;
#X obj 16 278 osc-voice amp5 pit5;
#X obj 16 302 osc-voice amp6 pit6;
#X obj 16 326 osc-voice amp7 pit7;
#X obj 16 350 osc-voice amp8 pit8;
#X obj 464 343 qlist;
#X msg 394 185 stop;
#X msg 524 300 read qlist.txt;
#X obj 524 255 loadbang;
#X text 258 164 start;
#X text 395 161 stop;
#X text 534 279 reread file;
#X msg 467 199 rewind;
#X msg 535 199 next;
#X msg 251 212 tempo 100 \, bang;
#X msg 250 188 tempo 1 \, bang;
#X text 82 11 USING QLIST TO SEQUENCE AN OSCILLATOR BANK;
#X text 479 178 single step;
#X obj 532 392 r #;
#X text 28 49 Here is an eight voice additive synthesis patch controlled
by a qlist. Open a text editor on the file \, "qlist.txt" \, to see
how the oscillators' amplitudes and frequencies are specified. The
abstraction \, "osc-voice" \, shows an effective way to make patches
react to qlists but also to mousing.;
#X text 234 391 this is where qlist comments go:;
#X obj 16 380 output~;
#X text 394 423 updatged for Pd version 0.39;
#X connect 0 0 1 0;
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#X connect 2 0 3 0;
#X connect 3 0 4 0;
#X connect 4 0 5 0;
#X connect 5 0 6 0;
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#X connect 7 0 24 1;
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#X connect 16 0 8 0;
#X connect 17 0 8 0;
#X connect 18 0 8 0;

--- NEW FILE: I10.phase.bash.pd ---
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#X obj 7 153 *~;
#X obj 7 98 *~;
#X obj 7 70 inlet~;
#X obj 7 125 rfft~;
#X obj 7 227 *~;
#X obj 7 306 rifft~;
#X obj 43 99 tabreceive~ \$0-hann;
#X obj 7 200 q8_sqrt~;
#X obj 337 158 samplerate~;
#X obj 328 124 bang~;
#X obj 337 183 t f b;
#X obj 444 4 loadbang;
#X obj 636 19 r window-size;
#X obj 636 65 block~;
#X obj 337 207 osc~;
#X msg 381 207 0;
#X obj 8 333 *~;
#X obj 44 334 tabreceive~ \$0-hann;
#X obj 9 363 outlet~;
#X obj 77 281 r window-size;
#X obj 7 281 /~ 1000;
#X msg 636 41 set \$1 2;
#X obj 387 267 r \$0-start;
#X obj 328 320 spigot;
#X msg 387 292 1;
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#X obj 328 493 del;
#X obj 364 395 / 2;
#X obj 364 418 - 1;
#X msg 443 28 \; pd dsp 1 \; window-size 1024 \; pitch 48 \; specshift
0;
#X text 96 196 magnitude of FT;
#X text 18 249 align partials to middle of window;
#X text 39 232 alternate every other sign to;
#X text 383 122 control computations to do every frame;
#X text 414 180 set sample rate of the oscillator to;
#X text 416 195 Nyquist (here we're operating at twice;
#X text 418 211 the global "samplerate~" because of;
#X text 417 228 the overlap-2 blocking.) Also set phase;
#X text 417 244 to zero at beginning of frame.;
#X text 424 287 When analysis starts \, set a delay to;
#X text 425 304 one frame minus a sample (i.e. \, just;
#X text 424 321 one 64-sample block before the next;
#X text 423 338 frame) which is synchronized with the;
#X text 423 352 first frame emerging from outlet~ at;
#X text 499 368 left. In the parent window;
#X text 497 385 this is used to start;
#X text 497 402 recording synchronously.;
#X text 14 384 output phase-aligned frames;
#X text 395 514 output a bang to start recording;
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#X restore 22 459 pd fft;
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#X floatatom 583 330 0 0 0 0 - specshift-set -;
#X obj 583 353 s specshift;
#X obj 407 443 s loco;
#X obj 586 400 s pitch;
#X obj 407 346 pack 0 100;
#X obj 588 453 output~;
#X text 214 -1 PHASE BASHING;
#X text 455 515 updated for Pd version 0.39;
#X floatatom 199 389 5 0 0 0 - #0-samp-msec -;
#X text 197 403 sample length \, msec;
#X msg 198 288 ../sound/bell.aiff;
#X msg 198 313 ../sound/voice.wav;
#X msg 198 338 ../sound/voice2.wav;
#X text 201 266 change input sound;
#X obj 198 364 s read-sample;
#N canvas 190 43 657 626 test-signal 0;
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#X obj 88 536 outlet~;
#X msg 88 360 0 \, \$1 \$2;
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#X obj 211 260 t b b f;
#X obj 152 225 t b f;
#X obj 88 406 tabread4~ \$0-sample;
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#X msg 402 379 1;
#X msg 401 404 0;
#X obj 87 473 hip~ 5;
#X obj 88 444 +~;
#X text 73 123 play sample;
#X text 71 143 once;
#X obj 482 420 s \$0-start;
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#X restore 22 436 pd test-signal;
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#X restore 82 311 graph;
#X obj 378 165 osc~;
#X obj 378 190 *~ -0.5;
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#X obj 37 88 r window-size;
#X obj 38 173 /;
#X obj 127 142 samplerate~;
#X obj 38 251 s window-sec;
#X obj 177 204 swap;
#X obj 177 228 /;
#X obj 177 252 s window-hz;
#X obj 49 201 * 1000;
#X obj 49 228 s window-msec;
#X obj 38 115 t f b f;
#X msg 173 92 resize \$1;
#X obj 173 116 s \$0-hann;
#X obj 330 105 r window-hz;
#X msg 382 130 0;
#X obj 330 131 t f b;
#X text 15 8 calculate Hann window table (variable window size) and
constants window-hz (fundamental frequency of analysis) \, window-sec
and window-msec (analysis window size in seconds and msec).;
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#X msg 28 290 1024;
#X msg 28 312 2048;
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#X obj 19 184 soundfiler;
#X text 118 379 read a sample;
#X obj 38 378 loadbang;
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#X obj 75 99 symbol \$0-sample;
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#X msg 38 402 \; read-sample ../sound/voice.wav;
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#X text 33 384 ---analyze---;
#X text 20 399 sample;
#X obj 94 489 tabwrite~ \$0-nophase;
#X obj 21 492 output~;
#X msg 415 384 0 \, 400 4000;
#X msg 415 419 0 \, 400 10000;
#X text 47 18 This patch takes an incoming sound \, does an overlap-2
FFT analysis of it \, and bashes the phases of the spectra so that
when regenerated the components will all have zero phase at the middle
of each window. You can use the windows as waveforms and cross-fade
them at will without getting phase modulation. This might be useful
for making synthetic instruments that mimic the spectral variation
of recorded sounds.;
#X text 398 305 (hundredths of sec);
#X text 401 289 location in sample;
#X text 420 365 normal speed;
#X text 422 403 slow;
#X text 458 262 ------ playback -------;
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#X obj 302 367 samphold~;
#X obj 508 290 phasor~;
#X obj 174 288 wrap~;
#X obj 510 68 r pitch;
#X obj 510 140 mtof;
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#X text 45 426 offset into;
#X text 50 440 sample;
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#X text 204 393 next block;
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#X text 60 248 grain size;
#X text 62 264 in samples;
#X text 97 383 grain size;
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#X text 165 248 fractional;
#X text 164 265 part of loc;
#X text 295 224 integer part of loc;
#X text 328 247 middle of block;
#X text 310 290 cvt to samples;
#X text 522 265 run two copies 180 degrees out of phase;
#X text 29 589 window shaped;
#X text 27 604 by raised cos;
#X text 265 522 weighted sum of;
#X text 265 538 2 windows;
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#X text 223 44 read location in sec/100;
#X obj 200 120 samplerate~;
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#X text 113 162 read location \, blocks;
#X obj 260 89 unpack;
#X msg 630 52 set \$1;
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#X msg 771 55 set \$1;
#X obj 630 30 r specshift;
#X text 723 190 1/(block size);
#X obj 630 76 s specshift-set;
#X obj 770 78 s pitch-set;
#X text 607 104 analysis overlap was 2 so our;
#X text 606 120 block size is (window size)/2;
#X text 12 -1 OVERLAPPED \, WINDOWED SAMPLE PLAYBACK;
#X text 357 0 - with controls for pitch \, location \, and spectral
shift;
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#X restore 589 428 pd playback;
#X text 585 290 spectral shift;
#X text 583 306 (hundredths of;
#X text 646 323 octave);
#X text 126 398 live;
#X text 45 141 You can save the analyses and needn't be running the
FFT patch to do the resynthesis. You can read a sample \, select window
size \, and press "sample" to analyze it \, or else analyze a "live"
input. You'll hear the phase-bashed sample as the analysis runs. You
can regenerate the sound with specified pitch \, sample location \,
and spectral shift \, using the "playback" controls.;
#X text 83 278 analysis;
#X text 80 264 (redo;
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--- NEW FILE: J05.triangle.pd ---
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#X text 381 257 ---- 0.02 seconds ----;
#X text 350 505 updated for Pd version 0.39;
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#X text 11 51 frequency;
#X text 126 50 SLOPES (percent);
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#X text 30 4 Making waveforms with corners by specifying line segment
slopes;
#X text 136 67 up;
#X text 209 68 down;
#X text 29 317 Occasionally a second method for making corners is more
convenient. Here we specify the slopes of the rising and falling segments
(as always \, in units per cycle). We then make a triangle wave with
a corner at (0 \, 0) and another one \, placed somewhere within the
cycle. The slopes of the two lines determine the second point \, which
will have an x value of t/(s+t) (if we let s denote the rising slope
and t the falling one \, both as positive numbers). The y value is
st/(s+t). If we wish instead to specify the corner location (x \, y)
(with x in cycles \, 0<x<1) we set s = y/x and t = y/(1-x). The DC
value is y/2.;
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--- NEW FILE: H06.envelope.follower.pd ---
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#X text 162 12 ENVELOPE FOLLOWER;
#X text 22 33 An envelope follower measures the mean square power of
an signal as it changes over time. (You can convert mean square power
to RMS ampitude or to decibels if you wish.) The term "mean square"
means simply that the signal should be squared \, and then averaged.
The averageing is done using a low-pass filter such as lop~.;
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#X floatatom 93 444 3 0 100 0 - #0-lop -;
#X obj 61 356 +~;
#X text 187 317 <-- frequency of second oscillator;
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#X text 335 361 built-in envelope;
#X obj 354 491 dbtorms;
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#X text 35 195 This subpatch loads initial;
#X text 31 219 values in number boxes.;
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#X restore 217 598 pd startup;
#X text 115 414 square the signal;
#X text 124 440 <-- responsiveness;
#X text 159 501 take snapshot;
#X text 108 548 convert to RMS;
#X text 327 599 updated for Pd version 0.39;
#X text 334 381 follower for comparison;
#X text 107 466 low-pass filter;
#X text 114 573 output;
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#X text 159 517 every 1/4 second;
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#X obj 376 414 env~;
#X text 20 242 The env~ object at right \, which is a built-in envelope
follower using a higher-quality low-pass filter than lop~ \, is shown
for comparison. Its output is artificially slowed down to match the
homemade one at left.;
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#X text 204 358 <-- on/off;
#X text 20 128 Here we're adding two oscillators so the result should
be an RMS of one if the second oscillator is on \, 0.707 otherwise.
Note two effects: first \, the more responsive the envelope follower
\, the less accurate the result (but the faster it responds). Second
\, if the two oscillators are tuned close to each other their beating
affects the nombers coming out.;
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