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Rishabh Gupta
purr-data
Commits
7476eae5
Commit
7476eae5
authored
Nov 08, 2017
by
Jonathan Wilkes
Browse files
Merge branch 'vanilla-port'
parents
f92c6c74
1e59369d
Changes
24
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pd/doc/3.audio.examples/C07.envelope.follower.pd
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7476eae5
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@@ -66,7 +66,7 @@ on and off. If on \, a test is run to determine whether to turn off
20 10 1 18 -261139 -33289 0;
#X obj 407 2 pddp/pddplink http://puredata.info/dev/pddp -text pddp
;
#X text 19 37 The [env~] object reports th
s
RMS signal level over the
#X text 19 37 The [env~] object reports th
e
RMS signal level over the
last 256 samples (by default) or any other power of 2 that's at least
twice the block size. The analysis is done in an overlapped fashion
so that results appear every N/2 points if N is the analysis window
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pd/doc/3.audio.examples/D10.sampler.notes.pd
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@@ -197,7 +197,7 @@ phase \, and sample number in one message;
the parameters one by one.;
#X obj 19 492 output~;
#X text 16 560 This patch take the same principle as the earlier "one-shot
sampler" \, but allows you to parametrize sample playback. Since we
sampler" \, but allows you to paramet
e
rize sample playback. Since we
must wait 5 msec before starting the playback \, we store all the parameters
in "f" objects \, and recall them to construct the new note. Transposition
is done by altering the amount to play back in the (artificial) ten
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pd/doc/3.audio.examples/E05.chebychev.pd
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@@ -235,7 +235,7 @@ rises from zero to one.;
the table (which would be clipped appropriately). Anyway \, the polynomials
increase rapidly in value outside the interval from -1 to 1 that we
are using here.;
#X text 20 457 When you get tired of Chebyche
f
polynomials you can
#X text 20 457 When you get tired of Chebyche
v
polynomials you can
draw your own functions by hand and/or try other formulas.;
#X obj 1 1 cnv 15 445 20 empty \$0-pddp.cnv.header waveshaping_with_Chebychev_polynomials
20 10 1 18 -261139 -33289 0;
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pd/doc/3.audio.examples/G06.octave.doubler.pd
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@@ -105,7 +105,7 @@ odd harmonics using a variable-delay comb filter tuned one octave above
the incoming sound. We use two taps into the delay line. The fixed
one (delread~) adjusts for the delayed output of [fiddle~]. The variable
one (vd~) adds to this an additional delay equal to 1/2 the measured
period of the incoming sound. T
H
ese two are added. Odd harmonics are
period of the incoming sound. T
h
ese two are added. Odd harmonics are
180 degrees out of phase at the two taps and cancel. Even harmonics
get through -- so the sound goes up an octave \, without denaturing
the timbre as a speed-up would.;
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pd/doc/3.audio.examples/H02.high-pass.pd
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@@ -127,7 +127,7 @@
#X restore 289 283 graph;
#X text 313 425 --- 0.02 sec ---;
#X text 19 36 Many synthesis algorithms and transformations can have
outputs with a zero-freqency component (commonly called DC for "direct
outputs with a zero-freq
u
ency 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.
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pd/doc/3.audio.examples/H06.envelope.follower.pd
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@@ -80,9 +80,9 @@ Wilkes revised the patch to conform to the PDDP template for Pd version
#X obj 6 567 pddp/pddplink ../5.reference/pddp/help.pd -text help;
#X text 19 37 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"
to RMS amp
l
itude or to decibels if you wish.) The term "mean square"
means simply that the signal should be squared \, and then averaged.
The averag
e
ing is done using a low-pass filter such as [lop~].;
The averaging is done using a low-pass filter such as [lop~].;
#X text 18 203 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
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pd/doc/3.audio.examples/H09.ssb.modulation.pd
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@@ -67,7 +67,7 @@
#X restore 125 505 pd startup;
#X obj 19 478 output~;
#X obj 186 330 phasor~;
#X text 121 421 <-- complex multipier;
#X text 121 421 <-- complex multip
l
ier;
#X text 120 438 (calculates real part);
#X text 289 354 to form the real and;
#X text 289 370 imaginary part of a;
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pd/doc/3.audio.examples/H13.butterworth.pd
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@@ -71,7 +71,7 @@ Wilkes revised the patch to conform to the PDDP template for Pd version
#X text 19 37 links:;
#X restore 103 461 pd References;
#X obj 6 461 pddp/pddplink ../5.reference/pddp/help.pd -text help;
#X text 19 143 The [butterworth3~] abstraction computes filter coeffients
#X text 19 143 The [butterworth3~] abstraction computes filter coeffi
ci
ents
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
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pd/doc/3.audio.examples/H16.adsr.filter.qlist.pd
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@@ -89,7 +89,7 @@ tempo 4;
#X text 441 483 click to reload qlist2.txt;
#X obj 22 519 output~;
#X text 201 39 This is an analog-synth sound made using a wavetable
oscillator and a "vcf~' object. Un
k
ike the "floyd" example earlier
oscillator and a "vcf~' object. Un
l
ike 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 199 105 The qlist reads the file \, "qlist2.txt" \, which contains
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pd/doc/3.audio.examples/I03.resynthesis.pd
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@@ -27,7 +27,7 @@ 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~ o
b
ly puts out those N/2 points.) There's only
(so it's OK that rfft~ o
n
ly 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
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pd/doc/3.audio.examples/I10.phase.bash.pd
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@@ -379,7 +379,7 @@ of recorded sounds.;
#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 c
v
t to samples;
#X text 310 290 c
onver
t 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;
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pd/doc/3.audio.examples/J08.classicsynth.pd
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@@ -98,7 +98,7 @@ 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. T
H
e oscillators' waveforms and tuning relationship
the vcf~ "Q" parameter. T
h
e oscillators' waveforms and tuning relationship
is controlled by other parameters set within the "pd 16x" window.;
#X obj 1 1 cnv 15 445 20 empty \$0-pddp.cnv.header the_classic_subtractive_synth_sound
20 10 1 18 -261139 -33289 0;
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pd/doc/3.audio.examples/filter-graph2.pd
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@@ -50,7 +50,7 @@
#X obj 620 215 t b;
#X text 583 184 clear filters;
#X text 582 198 to start;
#X text 578 452 c
b
eck if any table;
#X text 578 452 c
h
eck 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.);
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pd/doc/4.data.structures/05.array.pd
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@@ -67,7 +67,7 @@ the template subpatch.);
#X text 210 264 template5;
#X text 12 360 work as before \, but on;
#X text 12 375 array elements...;
#X text 12 345 normal "set" a
m
d "get";
#X text 12 345 normal "set" a
n
d "get";
#X obj 1 1 cnv 15 445 20 empty \$0-pddp.cnv.header arrays_in_data_structures
20 10 1 18 -261139 -33289 0;
#X obj 407 2 pddp/pddplink http://puredata.info/dev/pddp -text pddp
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pd/doc/4.data.structures/14.partialtracer.pd
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@@ -831,7 +831,7 @@ above.;
#X text 19 37 This patch derives sinusoidal "tracks" from a sampled
sound using sigmund~ and the data structure facilities. The number
of tracks may range from 1 to 50 You can edit the tracks (but note
that the resynthez
i
er is limited to 50-voice polyphony.);
that the resynthe
si
zer is limited to 50-voice polyphony.);
#X obj 1 1 cnv 15 445 20 empty \$0-pddp.cnv.header sinusoid_tracking
20 10 1 18 -261139 -33289 0;
#X obj 407 2 pddp/pddplink http://puredata.info/dev/pddp -text pddp
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pd/doc/5.reference/acoustics-help.pd
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@@ -10,7 +10,7 @@
#X text 12 125 WEBSITE http://crca.ucsd.edu/~msp/;
#X text 12 45 LICENSE SIBSD;
#X text 12 105 AUTHOR Miller Puckette;
#X text 12 165 HELP_PATCH_AUTHORS Updated for
p
d version 0.40. Jonathan
#X text 12 165 HELP_PATCH_AUTHORS Updated for
P
d version 0.40. Jonathan
Wilkes revised the patch to conform to the PDDP template for Pd version
0.42.;
#X text 12 25 KEYWORDS control conversion MIDI;
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pd/doc/5.reference/gatom-help.pd
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@@ -72,7 +72,7 @@ TIME. In a production patch \, you'll want to set a specific width.
;
#X floatatom 70 143 0 0 0 0 - - -;
#X text 148 143 width = 0 characters (read below);
#X text 69 240 A width of one gives a clickable toggle switch ala Max.
#X text 69 240 A width of one gives a clickable toggle switch a
la Max.
;
#X text 88 223 width = 1 character: toggle between 0 and 1;
#X text 19 260 limits;
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pd/doc/5.reference/oscparse-help.pd
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@@ -18,7 +18,7 @@
#X obj 57 576 print packet;
#X text 234 445 a blob;
#X text 416 442 packets from network;
#X text 521 466
s
listen on port 5000;
#X text 521 466 listen on port 5000;
#X text 558 521 UDP packets \, binary output, f 13;
#X text 45 41 oscparse take incoming lists of numbers \, interpreting
them as the bytes in an OSC message. The output is a list containing
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pd/doc/7.stuff/soundfile-tools/6.vocoder.pd
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@@ -290,7 +290,7 @@
#X msg 29 179 stop the transformation;
#X text 193 9 (old-fashioned) VOCODER;
#X text 28 31 This takes in two soundfiles and uses the first to "vocode"
the second. T
H
e resulting sound is as long as the shorter of the two
the second. T
h
e resulting sound is as long as the shorter of the two
inputs.;
#X msg 29 116 read the analysis sound from file;
#X msg 29 137 AND read the sound to be processed from file;
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pd/doc/7.stuff/tools/testtone16.pd
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@@ -730,7 +730,7 @@ set 1 \;;
#X obj 608 144 tgl 20 0 tone-ch116 tone-ch16 16 5 -8 0 12 -262144 -1
-1 1 1;
#X text 235 25 16 channel test tone patch;
#X text 316 77 AU
T
IO INPUT (RMS dB);
#X text 316 77 AU
D
IO INPUT (RMS dB);
#X text 337 175 AUDIO OUTPUT ON/OFF;
#X text 40 120 OFF;
#X text 101 88 noise;
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