diff --git a/pd/doc/3.audio.examples/C07.envelope.follower.pd b/pd/doc/3.audio.examples/C07.envelope.follower.pd index b9ab26653e9bb7a46d5106ec5bc8da77c77256e1..1250915bfe41ab3451458d4c70ed5d0f048317ce 100644 --- a/pd/doc/3.audio.examples/C07.envelope.follower.pd +++ b/pd/doc/3.audio.examples/C07.envelope.follower.pd @@ -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 ths RMS signal level over the +#X text 19 37 The [env~] object reports the 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 diff --git a/pd/doc/3.audio.examples/D10.sampler.notes.pd b/pd/doc/3.audio.examples/D10.sampler.notes.pd index 19713c361cd8d5e850fb8ae398b80c21e575f991..e400aab760e5ca73aad60f839dcfc73887f0c516 100644 --- a/pd/doc/3.audio.examples/D10.sampler.notes.pd +++ b/pd/doc/3.audio.examples/D10.sampler.notes.pd @@ -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 parameterize 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 diff --git a/pd/doc/3.audio.examples/E05.chebychev.pd b/pd/doc/3.audio.examples/E05.chebychev.pd index f112295349106d3527214cc0c9adcfda177a6d1d..4e6d97ac750b56745d7296f985029aa5397233f4 100644 --- a/pd/doc/3.audio.examples/E05.chebychev.pd +++ b/pd/doc/3.audio.examples/E05.chebychev.pd @@ -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 Chebychef polynomials you can +#X text 20 457 When you get tired of Chebychev 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; diff --git a/pd/doc/3.audio.examples/G06.octave.doubler.pd b/pd/doc/3.audio.examples/G06.octave.doubler.pd index 8cd9def60a5abc34d4436e700cc7d9aac407291f..47f8cd1ed26c8ff17c86d84367511756842de8d4 100644 --- a/pd/doc/3.audio.examples/G06.octave.doubler.pd +++ b/pd/doc/3.audio.examples/G06.octave.doubler.pd @@ -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. THese two are added. Odd harmonics are +period of the incoming sound. These 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.; diff --git a/pd/doc/3.audio.examples/H02.high-pass.pd b/pd/doc/3.audio.examples/H02.high-pass.pd index 441a8bf20d9358ad7ea5ba93c66d06347753d7c8..f3cc9fc227550f34d93fec333e12a04e2f7cc72a 100644 --- a/pd/doc/3.audio.examples/H02.high-pass.pd +++ b/pd/doc/3.audio.examples/H02.high-pass.pd @@ -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-frequency 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. diff --git a/pd/doc/3.audio.examples/H06.envelope.follower.pd b/pd/doc/3.audio.examples/H06.envelope.follower.pd index 658f6a13fb0e9366bb4e0bcdd43e8a80a5c852ac..a03e465cf4cd8aaa858bd338baecfc735ff89605 100644 --- a/pd/doc/3.audio.examples/H06.envelope.follower.pd +++ b/pd/doc/3.audio.examples/H06.envelope.follower.pd @@ -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 amplitude 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~].; +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 diff --git a/pd/doc/3.audio.examples/H13.butterworth.pd b/pd/doc/3.audio.examples/H13.butterworth.pd index 82f4915ab78f36de7708f130708ce1c341c75b1a..4a6e7b677085132b74315c22276c6eb460251120 100644 --- a/pd/doc/3.audio.examples/H13.butterworth.pd +++ b/pd/doc/3.audio.examples/H13.butterworth.pd @@ -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 coefficients 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 diff --git a/pd/doc/3.audio.examples/H16.adsr.filter.qlist.pd b/pd/doc/3.audio.examples/H16.adsr.filter.qlist.pd index f7024b1f0859bf626d470e1d8577618ec0f55256..43321cf72ceb8272895a32e35b2af81cc276e0dc 100644 --- a/pd/doc/3.audio.examples/H16.adsr.filter.qlist.pd +++ b/pd/doc/3.audio.examples/H16.adsr.filter.qlist.pd @@ -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. Unkike the "floyd" example earlier +oscillator and a "vcf~' object. Unlike 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 diff --git a/pd/doc/3.audio.examples/I03.resynthesis.pd b/pd/doc/3.audio.examples/I03.resynthesis.pd index 9d0ce6212b89764536b1bda655b873bf77620e43..a0502dcca87f51de6b760edc4b2c6a49cb43fb50 100644 --- a/pd/doc/3.audio.examples/I03.resynthesis.pd +++ b/pd/doc/3.audio.examples/I03.resynthesis.pd @@ -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~ obly puts out those N/2 points.) There's only +(so it's OK that rfft~ only 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 diff --git a/pd/doc/3.audio.examples/I10.phase.bash.pd b/pd/doc/3.audio.examples/I10.phase.bash.pd index b67a00f66ab855c2152436be555363a04aeb52bb..519f9f04f7d3ef7e8875ea67f76fdbf0e2be4692 100644 --- a/pd/doc/3.audio.examples/I10.phase.bash.pd +++ b/pd/doc/3.audio.examples/I10.phase.bash.pd @@ -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 cvt to samples; +#X text 310 290 convert 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; diff --git a/pd/doc/3.audio.examples/J08.classicsynth.pd b/pd/doc/3.audio.examples/J08.classicsynth.pd index e6dadbbcadcc33c78425873797bfaaadc9759644..45db63f79ccbc5959abb34e9a951fec86ab93114 100644 --- a/pd/doc/3.audio.examples/J08.classicsynth.pd +++ b/pd/doc/3.audio.examples/J08.classicsynth.pd @@ -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. THe oscillators' waveforms and tuning relationship +the vcf~ "Q" parameter. The 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; diff --git a/pd/doc/3.audio.examples/filter-graph2.pd b/pd/doc/3.audio.examples/filter-graph2.pd index a800957d99f75c279dcd58ed250cda0570d26d27..b8d78789a2f683036ede6c19c7e424e0569a9ac2 100644 --- a/pd/doc/3.audio.examples/filter-graph2.pd +++ b/pd/doc/3.audio.examples/filter-graph2.pd @@ -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 cbeck if any table; +#X text 578 452 check 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.); diff --git a/pd/doc/4.data.structures/05.array.pd b/pd/doc/4.data.structures/05.array.pd index 675d151d4f10fed3591e506f50891d70eddb213c..2f0381b43b4280b4afb5f2bf6a00febd8527c883 100644 --- a/pd/doc/4.data.structures/05.array.pd +++ b/pd/doc/4.data.structures/05.array.pd @@ -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" amd "get"; +#X text 12 345 normal "set" and "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 diff --git a/pd/doc/4.data.structures/14.partialtracer.pd b/pd/doc/4.data.structures/14.partialtracer.pd index e62697efa5a1029ab80daab91447ecfd6fdcf364..af737caeffa16d2b00970f442f98816eed5511b0 100644 --- a/pd/doc/4.data.structures/14.partialtracer.pd +++ b/pd/doc/4.data.structures/14.partialtracer.pd @@ -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 resynthezier is limited to 50-voice polyphony.); +that the resynthesizer 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 diff --git a/pd/doc/5.reference/gatom-help.pd b/pd/doc/5.reference/gatom-help.pd index 75f1883de585962a629e336935ca1385fef4ceb7..d9e2d226d7e400d7757b2b7f81cc7e33a402a012 100644 --- a/pd/doc/5.reference/gatom-help.pd +++ b/pd/doc/5.reference/gatom-help.pd @@ -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; diff --git a/pd/doc/5.reference/oscparse-help.pd b/pd/doc/5.reference/oscparse-help.pd index d0c7843635cd28c92aa518e54ca795b4da5d890a..3b84ec586cc128136e9872cf4243a948eab830bb 100644 --- a/pd/doc/5.reference/oscparse-help.pd +++ b/pd/doc/5.reference/oscparse-help.pd @@ -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 slisten 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 diff --git a/pd/doc/7.stuff/soundfile-tools/6.vocoder.pd b/pd/doc/7.stuff/soundfile-tools/6.vocoder.pd index ce822ef4b223fd5131dc97ed507a72babe0079fd..61e57fc01688c8676ac88fc6b8f0b8d5296865b6 100644 --- a/pd/doc/7.stuff/soundfile-tools/6.vocoder.pd +++ b/pd/doc/7.stuff/soundfile-tools/6.vocoder.pd @@ -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. THe resulting sound is as long as the shorter of the two +the second. The 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; diff --git a/pd/doc/7.stuff/tools/testtone16.pd b/pd/doc/7.stuff/tools/testtone16.pd index f3874c60e945a0ce68cf7db4a7064fa344863beb..66a7297d345ea2c16cc5dfcabbcbdcd8ab1bbf45 100644 --- a/pd/doc/7.stuff/tools/testtone16.pd +++ b/pd/doc/7.stuff/tools/testtone16.pd @@ -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 AUTIO INPUT (RMS dB); +#X text 316 77 AUDIO INPUT (RMS dB); #X text 337 175 AUDIO OUTPUT ON/OFF; #X text 40 120 OFF; #X text 101 88 noise;