get rid of obsolete audio examples

svn path=/trunk/; revision=3005

17 files changed, 0 insertions, 2499 deletions

diff --git a/pd/doc/3.audio.examples/H01.intro.filters.pd b/pd/doc/3.audio.examples/H01.intro.filters.pd
deleted file mode 100644
index 4a64a0b5..00000000
--- a/pd/doc/3.audio.examples/H01.intro.filters.pd
+++ /dev/null
@@ -1,72 +0,0 @@
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-#X text 9 11 Filters;
-#X msg 87 130 \; pd dsp 1 \; metro bang;
-#X msg 191 135 \; pd dsp 0;
-#X text 15 417 Original;
-#X text 122 417 Low pass;
-#X text 210 417 High pass;
-#X text 298 416 Band pass;
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-#X text 97 110 start;
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-#X text 51 236 <-- change frequency;
-#X text 294 247 <-- center/rolloff frequency;
-#X text 374 297 <-- Q;
-#X text 548 341 This graph holds 4 arrays for the input and three filter outputs.;
-#X text 375 389 RMS amplitudes of the original signal and the three filter outputs;
-#X text 746 536 updated for Pd version 0.26;
-#X obj 334 482 metro 1000;
-#X text 29 33 Pd provides low \, high \, and band pass filters \, as shown here. By changing the test frequency \, the filter frequency \, and the "Q" value \, you can see how these filters affect the amplitude and phase of incoming signals;
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-#X text 456 434 Notice how the phase of the graphed sinusoids slips back and forth... this is because graphing always starts on the nearest 64-sample boundary to the time the metronome fires. If you run at 48K the slippage disappears \, because then the metronome fires every 48K samples \, which is a multiple of 64;
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diff --git a/pd/doc/3.audio.examples/H02.bandpass.pd b/pd/doc/3.audio.examples/H02.bandpass.pd
deleted file mode 100644
index 0ce9cc47..00000000
--- a/pd/doc/3.audio.examples/H02.bandpass.pd
+++ /dev/null
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-#X text 648 14 START;
-#X text 845 19 STOP;
-#X text 88 11 BANDPASS FILTERS;
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-to troll for partials in Jonathan Harvey's famous bell sample.;
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-#X text 14 82 Note that filters can give unexpected level changes.
-The bp~ object is designed to have roughly unit gain at teh pass band
-\, so the higher you set "Q" the more amplitude is lost. YOu can correct
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-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.;
-#X text 16 233 You can hear partials around 48 \, 51.3 \, 55 (faint!)
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-\, 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.;
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-#X text 281 450 (shift-drag to fine tune);
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diff --git a/pd/doc/3.audio.examples/H03.filter.sweep.pd b/pd/doc/3.audio.examples/H03.filter.sweep.pd
deleted file mode 100644
index 0d478275..00000000
--- a/pd/doc/3.audio.examples/H03.filter.sweep.pd
+++ /dev/null
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diff --git a/pd/doc/3.audio.examples/H04.filter.floyd.pd b/pd/doc/3.audio.examples/H04.filter.floyd.pd
deleted file mode 100644
index 02027117..00000000
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-#X text 277 354 convert to Hz.;
-#X text 35 6 ANOTHER SWEEPING FILTER EXAMPLE;
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diff --git a/pd/doc/3.audio.examples/H07.ssb.modulation.pd b/pd/doc/3.audio.examples/H07.ssb.modulation.pd
deleted file mode 100644
index 968b3b45..00000000
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-#X text 95 261 This is a pair of all-pass filters whose outputs somehow
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-know what phase relation they have with the original signal. I adapted
-this from a 4X patch by Emmanuel Favreau \, circa 1982;
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diff --git a/pd/doc/3.audio.examples/J03.qlist.pd b/pd/doc/3.audio.examples/J03.qlist.pd
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index 58495ca1..00000000
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-#X text 82 11 USING QLIST TO SEQUENCE AN OSCILLATOR BANK;
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-#X text 35 61 Here is an eight voice additive synthesis patch controlled
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-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 258 453 this is where qlist comments go:;
-#X text 418 485 updatged for Pd version 0.34;
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-#X connect 7 1 8 0;
-#X connect 8 0 5 0;
-#X connect 9 1 4 1;
-#X connect 10 0 24 0;
-#X connect 11 0 1 1;
-#X connect 11 0 9 0;
-#X connect 12 0 10 0;
-#X connect 12 0 18 0;
-#X connect 14 0 22 0;
-#X connect 15 0 17 0;
-#X connect 15 0 20 0;
-#X connect 18 0 19 0;
-#X connect 21 0 22 1;
-#X connect 22 0 23 0;
-#X connect 22 0 23 1;
-#X connect 24 0 21 0;
-#X restore 39 406 pd output;
-#X msg 107 376 MUTE;
-#X text 150 375 <-- output amplitude;
-#X text 518 189 <-- attack;
-#X text 485 281 <-- release;
-#X obj 39 338 *~;
-#X obj 39 282 *~;
-#X obj 39 310 *~;
-#X obj 80 340 osc~ 440;
-#X text 88 9 ADSR envelope;
-#X text 35 29 You can use the quartic trick to simplify patches using
-ADSR envelopes. For amplitude control it's especially simple. It's
-usually good enough in practice to control amplitudes by ranging an
-ADSR from 0 to 1 (divide by 100 so your input range can be a comfortable
-0-100) \, then take fourth power:;
-#X obj 77 227 / 100;
-#X floatatom 77 197 3 0 100;
-#X msg 383 269 \; trigger 0;
-#X obj 39 166 unpack;
-#X floatatom 39 197 1 0 100;
-#X msg 383 175 \; trigger 1 100;
-#X text 503 229 <-- softer attack;
-#X msg 382 222 \; trigger 1 60;
-#X text 441 583 updated for Pd version 0.34;
-#X text 14 437 Note that the units aren't dB \; for most of the range
-0-100 it's about 0.4 dB per unit. If you want something closer to dB
-\, you can set the scale as 0-40 instead of 0-100 (just change "/ 100"
-to "/ 40") and then you'll get the response shown in the first patch
-in this section.;
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-#X obj 596 547 *~;
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-#X obj 605 441 sqrt;
-#X obj 605 466 sqrt;
-#X text 174 526 To use ADSR to control pitch \, you should;
-#X text 173 545 usually just use real pitch units like this-->;
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-#X connect 7 0 2 0;
-#X connect 8 0 9 0;
-#X connect 8 0 9 1;
-#X connect 9 0 7 0;
-#X connect 10 0 7 1;
-#X connect 13 0 32 1;
-#X connect 14 0 13 0;
-#X connect 16 0 17 0;
-#X connect 16 1 14 0;
-#X connect 17 0 32 0;
-#X connect 23 0 24 0;
-#X connect 23 0 24 1;
-#X connect 25 0 26 0;
-#X connect 26 0 27 0;
-#X connect 27 0 28 0;
-#X connect 28 0 31 1;
-#X connect 31 0 23 0;
-#X connect 31 0 23 1;
-#X connect 32 0 8 0;
-#X connect 32 0 8 1;
diff --git a/pd/doc/3.audio.examples/J05.vibrato.pd b/pd/doc/3.audio.examples/J05.vibrato.pd
deleted file mode 100644
index 78c38efd..00000000
--- a/pd/doc/3.audio.examples/J05.vibrato.pd
+++ /dev/null
@@ -1,158 +0,0 @@
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-#X obj 27 246 unpack;
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-#X text 463 668 updated for Pd version 0.34;
-#X obj 26 525 +~ 0.3;
-#X obj 26 551 cos~;
-#X obj 26 499 osc~;
-#X text 88 9 PORTAMENTO AND VIBRATO;
-#X obj 62 300 mtof;
-#X obj 62 325 sqrt;
-#X obj 62 350 sqrt;
-#X text 619 402 <-- midC;
-#X text 607 444 <-- octave up;
-#X msg 484 388 \; trigger 1 60;
-#X msg 483 435 \; trigger 1 72;
-#X text 584 488 <-- release;
-#X text 590 506 is optional;
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-#X floatatom 236 286 3 0 0;
-#X obj 236 313 / 6;
-#X obj 236 370 *~;
-#X floatatom 390 323 3 0 0;
-#X text 235 421 since we'll multiply \,;
-#X text 234 436 vibrato output should;
-#X text 234 453 be centered at 1 \, not 0;
-#X text 275 372 multiply by vib depth;
-#X obj 390 350 / 6923;
-#X text 28 35 Portamento can be treated as a special case of an ADSR
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-in units of pitch \, but it's also easy to add vibrato to the envelope--before
-raising it to the fourth power \, so that it acts pseudo-logarithmically.
-Rather than add to the ADSR output \, we multiply a signal which controls
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-;
-#X text 61 417 apply vibrato;
-#X text 65 445 fourth;
-#X text 68 461 power;
-#X text 96 529 waveform;
-#X text 95 509 simple;
-#X text 465 344 4/(exp(log(2)/1200)-1);
-#X text 469 325 conversion factor is;
-#X text 383 279 vibrato depth;
-#X text 382 296 in cents;
-#X text 233 245 vibrato speed;
-#X text 232 262 in Hertz;
-#X text 152 168 I made a table with 6 cycles of vibrato and made small
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-or triangle wave if you prefer.;
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-#X connect 11 0 12 0;
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-#X connect 29 0 27 0;
-#X connect 30 0 25 0;
-#X connect 31 0 36 0;
-#X connect 36 0 30 1;
-#X connect 50 0 24 0;
diff --git a/pd/doc/3.audio.examples/J06.adsr.sequenced.pd b/pd/doc/3.audio.examples/J06.adsr.sequenced.pd
deleted file mode 100644
index 26300054..00000000
--- a/pd/doc/3.audio.examples/J06.adsr.sequenced.pd
+++ /dev/null
@@ -1,217 +0,0 @@
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-#X obj 315 401 / 69.23;
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-#X text 402 300 ADSR for filter. Here \, I thought it better to make
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-the "/ 69.23" as compared to the previous patch.);
-#X text 141 5 USING QLIST TO MAKE SEQUENCES OF "NOTES";
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diff --git a/pd/doc/3.audio.examples/J09.up.downsampling.pd b/pd/doc/3.audio.examples/J09.up.downsampling.pd
deleted file mode 100644
index cf50f9b9..00000000
--- a/pd/doc/3.audio.examples/J09.up.downsampling.pd
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-#X obj 78 258 outlet~ lin;
-#X obj 78 38 inlet~ lin;
-#X text 172 257 "lin" is for linear upsampling;
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-#X text 160 39 "lin" has (still !) no meaning when downsampling;
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-#X text 718 314 a pd-bug !;
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-#X connect 20 0 53 0;
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-#X connect 23 0 52 0;
-#X connect 25 0 24 0;
-#X connect 26 0 35 1;
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-#X connect 29 0 34 0;
-#X connect 30 0 35 0;
-#X connect 32 0 31 0;
-#X connect 34 0 24 0;
-#X connect 34 1 31 0;
-#X connect 35 0 27 0;
-#X connect 36 0 40 1;
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diff --git a/pd/doc/3.audio.examples/K01.pulse.width.mod.pd b/pd/doc/3.audio.examples/K01.pulse.width.mod.pd
deleted file mode 100644
index 214d250a..00000000
--- a/pd/doc/3.audio.examples/K01.pulse.width.mod.pd
+++ /dev/null
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-#X msg 162 324 MUTE;
-#X obj 86 137 phasor~ 0;
-#X text 83 86 frequency;
-#X graph graph1 0 -1.02 882 1.02 668 179 868 49;
-#X array phasor1-output 882 float 0;
-#X pop;
-#X msg 191 77 bang;
-#X text 231 76 <-- click to graph;
-#X text 57 9 CLASSICAL PULSE WIDTH MODULATION;
-#X obj 102 196 phasor~ 0;
-#X obj 102 172 + 0.2;
-#X obj 86 246 -~;
-#X graph graph1 0 -1.02 882 1.02 668 316 868 186;
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-#X pop;
-#X obj 191 164 tabwrite~ phasor1-output;
-#X obj 191 222 tabwrite~ phasor2-output;
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-#X text 12 386 This patch demonstrates pulse width modulation \, which
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-phase difference. Here their frequencies are set to differ by 1/5 Hz.
-so that the relative phase wanders continuously.;
-#X text 669 459 ---- 0.02 seconds ----;
-#X text 665 498 updated for Pd version 0.34;
-#X text 203 325 <-- output;
-#X connect 0 0 5 0;
-#X connect 0 0 12 0;
-#X connect 2 0 3 1;
-#X connect 3 0 2 0;
-#X connect 4 0 3 2;
-#X connect 5 0 13 0;
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-#X connect 11 0 13 1;
-#X connect 11 0 16 0;
-#X connect 12 0 11 0;
-#X connect 13 0 17 0;
-#X connect 13 0 3 0;
diff --git a/pd/doc/3.audio.examples/K02.stereo.pd b/pd/doc/3.audio.examples/K02.stereo.pd
deleted file mode 100644
index 1c417df5..00000000
--- a/pd/doc/3.audio.examples/K02.stereo.pd
+++ /dev/null
@@ -1,87 +0,0 @@
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-#X msg 340 12 \; pd dsp 1;
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-#X text 361 45 ON;
-#X text 424 43 OFF;
-#X floatatom 123 196;
-#N canvas 159 26 618 383 output 0;
-#X obj 393 156 t b;
-#X obj 393 106 f;
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-#X msg 393 81 bang;
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-#X text 148 106 show level;
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-#X connect 9 1 4 1;
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-#X connect 11 0 1 1;
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-#X connect 12 0 10 0;
-#X connect 12 0 18 0;
-#X connect 14 0 22 0;
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-#X connect 15 0 20 0;
-#X connect 18 0 19 0;
-#X connect 21 0 22 1;
-#X connect 21 0 27 1;
-#X connect 22 0 28 0;
-#X connect 23 0 21 0;
-#X connect 26 0 27 0;
-#X connect 27 0 29 0;
-#X restore 61 224 pd output;
-#X msg 152 196 MUTE;
-#X text 186 195 <-- output amplitude;
-#X obj 59 111 phasor~ 0;
-#X text 56 45 frequency;
-#X text 331 323 updated for Pd version 0.26;
-#X text 57 9 CLASSICAL PULSE WIDTH MODULATION;
-#X obj 131 110 phasor~ 0;
-#X obj 59 134 -~ 0.5;
-#X obj 131 85 + 0.5;
-#X obj 131 135 -~ 0.5;
-#X obj 131 160 *~ -1;
-#X text 34 262 Here's what happens if you take the previous patch but \, instead of subtracting the two sawtooth waves \, we put them in two speakers with opposite phase.;
-#X connect 0 0 9 0;
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-#X connect 5 0 6 2;
-#X connect 6 0 5 0;
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-#X connect 9 0 14 0;
-#X connect 13 0 16 0;
-#X connect 14 0 6 0;
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diff --git a/pd/doc/3.audio.examples/K03.envelope.mod.pd b/pd/doc/3.audio.examples/K03.envelope.mod.pd
deleted file mode 100644
index 52551163..00000000
--- a/pd/doc/3.audio.examples/K03.envelope.mod.pd
+++ /dev/null
@@ -1,148 +0,0 @@
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-#X graph graph1 0 -1.02 44100 1.02 764 446 964 316;
-#X array cos-output 44100 float 0;
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-#X graph graph1 0 -1.02 44103 1.02 761 148 961 18;
-#X array sample-table 44103 float 0;
-#X pop;
-#X obj 593 579 loadbang;
-#X floatatom 96 489 0 0 0;
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-#X msg 214 64 \; pd dsp 1;
-#X obj 83 194 line~;
-#X obj 22 212 *~;
-#X obj 22 241 dac~;
-#X obj 83 171 pack 0 50;
-#X text 20 158 audio;
-#X text 93 110 show level;
-#X connect 0 0 4 0;
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-#X connect 6 0 1 0;
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-#X connect 9 1 4 1;
-#X connect 10 0 24 0;
-#X connect 11 0 1 1;
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-#X connect 12 0 10 0;
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-#X connect 14 0 22 0;
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-#X connect 21 0 22 1;
-#X connect 22 0 23 0;
-#X connect 22 0 23 1;
-#X connect 24 0 21 0;
-#X restore 67 517 pd output;
-#X msg 133 490 MUTE;
-#X text 176 490 <-- output amplitude;
-#X obj 67 460 hip~ 5;
-#X obj 47 408 tabread4~ sample-table;
-#X floatatom 46 50 0 0 0;
-#X text 85 49 <-- frequency (Hz.);
-#X floatatom 97 358 0 0 0;
-#X obj 97 385 * 441;
-#X obj 47 358 *~ 0;
-#X obj 47 383 +~ 1;
-#X text 271 274 <-- click to display output;
-#X msg 229 275 bang;
-#X text 136 357 <-- chunk size (100ths of a second);
-#X obj 588 471 adc~ 1;
-#X obj 588 495 hip~ 5;
-#X obj 588 545 tabwrite~ sample-table;
-#X msg 603 518 bang;
-#X text 635 518 <-- click here to record your own sample;
-#X text 704 583 v-- re-read the original sample;
-#X text 40 9 ENVELOPING YOUR LOOPING SAMPLER;
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-#X obj 88 259 clip~ -0.5 0.5;
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-#X graph graph1 0 -1.02 44100 1.02 763 311 963 181;
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-#X pop;
-#X obj 229 303 tabwrite~ cos-input;
-#X obj 88 284 cos~;
-#X obj 88 126 wrap~;
-#X obj 88 155 -~ 0.5;
-#X obj 88 332 *~ -0.5;
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-#X obj 229 329 tabwrite~ cos-output;
-#X floatatom 119 187 0 0 0;
-#X obj 67 433 *~;
-#X text 157 98 subtracting 0.5 and wrapping produces a sawtooth wave
-180 degrees out of phase from the original.;
-#X text 153 150 as before we subtract 1/2 again to center the sawtooth
-from -1/2 to 1/2.;
-#X text 153 188 <-- sharpness (at least 1);
-#X msg 593 608 read ../sound/voice.wav sample-table;
-#X obj 593 633 soundfiler;
-#X text 776 150 -- 44103 samples ---;
-#X text 767 447 ----- 1 second ------;
-#X obj 119 211 max 1;
-#X obj 46 77 phasor~;
-#X text 24 560 Here we apply an amplitude envelope to protect against
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-width modulation example \, except that the widening is applied to
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-#X text 23 647 To see the envelope \, put the phasor on 2 Hz \, select
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-;
-#X text 734 688 updated for Pd version 0.34;
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-#X connect 25 0 31 0;
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diff --git a/pd/doc/3.audio.examples/K04.even.odd.pd b/pd/doc/3.audio.examples/K04.even.odd.pd
deleted file mode 100644
index bfa950a9..00000000
--- a/pd/doc/3.audio.examples/K04.even.odd.pd
+++ /dev/null
@@ -1,116 +0,0 @@
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-#X text 148 106 show level;
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-#X msg 156 369 MUTE;
-#X text 13 430 A patch to split a sawtooth into even and odd harmonics
-ala Buchla. The wrap~ object folds its input into the interval [0 \,
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-#X text 631 567 ---- 0.02 seconds ----;
-#X text 577 613 updated for Pd version 0.34;
-#X text 193 369 <-- output;
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-#X connect 5 0 8 0;
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-#X connect 16 0 12 0;
-#X connect 18 0 19 2;
-#X connect 19 0 18 0;
-#X connect 20 0 19 3;
diff --git a/pd/doc/3.audio.examples/K05.bandlimited.pd b/pd/doc/3.audio.examples/K05.bandlimited.pd
deleted file mode 100644
index f49bcb68..00000000
--- a/pd/doc/3.audio.examples/K05.bandlimited.pd
+++ /dev/null
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-#X text 366 79 period is 2000 samples \, twice the table length;
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-#X connect 15 0 17 0;
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-#X connect 22 0 23 0;
-#X connect 22 0 23 1;
-#X connect 24 0 21 0;
-#X restore 177 469 pd output;
-#X msg 235 441 MUTE;
-#X text 276 440 <-- output amplitude;
-#X text 27 503 Patch to make an approximately band-limited sawtooth. This is useful if you intend to use sawtooth generators above about 200 Hz. \, perhaps to use any of the techniques shown in the previous four patches.;
-#X text 28 562 We generate a perfect square wave at Nyquist/6 \; this will have partials 1 \, 3 \, and 5 \, but the Nyquist frequency at partial 6 will cut off the rest of the partials. This is stored in array1 using the "make-table" subpatch.;
-#X text 64 34 BAND-LIMITED SAWTOOTH GENERATOR;
-#X obj 43 459 tabwrite~ array2;
-#X msg 44 435 bang;
-#X text 28 632 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.;
-#X text 338 737 updated for Pd version 0.26;
-#X text 41 409 graph output;
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diff --git a/pd/doc/3.audio.examples/K06.triangle.pd b/pd/doc/3.audio.examples/K06.triangle.pd
deleted file mode 100644
index ce8ac916..00000000
--- a/pd/doc/3.audio.examples/K06.triangle.pd
+++ /dev/null
@@ -1,169 +0,0 @@
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-#X text 92 12 DELAYS;
-#X obj 33 341 -~;
-#X obj 30 241 tabread4~ mtof;
-#X obj 33 317 *~ 3;
-#X obj 74 318 *~ 2;
-#X obj 49 266 phasor~;
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-#X obj 44 374 delwrite~ delay1 2000;
-#X floatatom 49 401 0 0 0 0 - - -;
-#X obj 49 426 delread~ delay1 1000;
-#X obj 30 47 loadbang;
-#X text 210 37 You can delay a signal using the delwrite~ and delread~
-objects. In this example \, the pitch of the oscillator is varying
-slightly so that the delayed signal is different from the straight
-signal.;
-#X text 212 99 delread always delays the signal an integer number of
-samples and does no interpolation.;
-#X text 211 137 The delwrite~ object creates the delay line \; you
-give it a name and a size in milliseconds. Each delwrite~ should have
-a different name.;
-#X text 209 184 Delread~'s arguments are the name of a delwrite (of
-which there should be exactly one) and a delay time in milliseconds
-between 0 and the length of the delay line. Each delwrite~ may have
-as many delread~s as you wish \, which function as multiple delay taps.
-;
-#X obj 30 215 +~ 60;
-#X text 115 216 <-- pitch;
-#X text 83 401 <-- delay time;
-#X text 65 342 asymmetric triangle wave;
-#X text 236 372 write to delay line;
-#X text 232 425 read from delay line;
-#X text 64 454 add the original and the delayed signal;
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-#X text 356 50 bang to recalculate the mtof table;
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