Attaching a work in progress, would love some feedback. Especially on the interface - I feel like there's a lot going on, with the delay lines, with the playback speed ratio... So if there's anything that doesn't feel intuitive, anything that could be simpler, anything that doesn't need to be there, let me know! Looking for "user feedback" !
The gist is this: you open two samples, and they loop in the background - it's "played" by periodically allowing the signal through, into delay lines. there's a fader to move between the two channels, and playback speed is determined by ratios, so that the layers are musically/harmonically related.
One way to use it: set up a bed of low-frequency sounds, and then double the speed (2 to 1 ratio), add some moments of a-perfect-fifth-above (3 to 2) and create melodies.
Here is an example, using clips from two piano pieces recorded to tape.
I've also used field recordings and sound effects, it's been fun to experiment
Here's the situation:
I've got 22 toggles that are each running their own array. But because they're all headed to the same input, I need a way to switch between them easily. The trick is, the patch really only works if I send just one array at a time...
My first thought was to use vradio and some conditional logic, but I've come across this issue:
If I click value 2 for example - even if it turns on the toggle assigned to value 2 AND doesn't turn on the toggles assigned to the other 21 values, it still WILL send bangs through to all the rest of the arrays, causing many of them to play out of turn.
Is there a clean fix, maybe an object I don't know about?
What are my options?
I need a 5 second sine wave that oscillates between 500 and -500 and lasts for 20 seconds. any advice? not sure how to change the parameters of the osc~ object!
using pd vanilla.
Thanks in advance!
by the way, what are my options beside osc? are there other easily accessible waveform options?
hey thanks a ton, David!
I have a couple questions... the array seems to write for only the first couple minutes (I'm seeing 4e+06 in the number box under the soundfiler tied to the read -resize pretty consistently for longer files. shooting for ~20 minutes)
also for louder clips I'm getting clicks during transition. I know there's a way to write in quick crossfades, just not sure where to put it in the patch. is that what the delay 50 is for?
by the way, the "loop" feature is SO handy. what a great thing to add.
oh, plus the "grain" slider ! it rocks, what exactly is it doing? seems like its corralling the picks either closer or father from the last pick. what’s that called?
Cheers and thanks again!
Hello! looking for a little guidance.
I have an audio file, a field recording of nature sounds with lovely blips of birds and other intermittent animal calls.
Imagine a timed sequence cycling through the clip, so that every 1 second (or however long) it hops to another part of the file. At every switch, a new short clip from the one file. Might be a kind of cool and rhythmic way to shuffle through the recording, almost like a beat but not quite.
Here’s where I’m at:
probably can random-number-generate a start point (I guess I’ll have to pre-load the amount of samples the audio file has first though?)
should be able to trigger the switch using metronome (to get that nicely moveable tempo, too)
What are your thoughts? Is it doable? The file is pretty long, is readsf the way to go?
By the way, using the basic Pd-0.48-0
Thanks a ton!
So here's where I'm at!
I've seen pure data work with great success rearranging midi notes as a sequencer, but its always timed to a metronome.
I'm hoping to load in ~40 .wav files, each anywhere from 2-4 seconds long (they are clips of short phrases from a speech) so that the patch rearranges them, maybe randomly, and creates a new version of the speech - the hope is that its basically unintelligible, but retains the melody and rhythm of the speaker... but I’m running in to a couple roadblocks.
First - how do I load that many audio files into pd? I’m assuming its going to be with tabread, not readsf…
Second - how do I line them up so that the end of playback of the first file triggers the start of playback of the second, then the third, etc?
It doesn’t matter really the qualities of the sequence, whether it randomizes the files themselves and ends when its over, or just picks randomly and plays back infinitely… I’m not picky!
PS - all the audio will be cut from the same source, so they’ll all be structurally the same (same sample rate etc), just different lengths. I’m using the basic Pd-0.48-0 on a Mac.