i just got off work so i should be able to test tonight
cant post audio till i fix my pc tho
i’ll describe my findings and ask questions if i run into problems
again, massive massive thanks for continued work on this and making it available for others to check out!
figured out internal seq…its awesome:
set at least one of the 8 tracks to CLK
select another track and set it to TRIG
choose track assigned to CLK as input for TRIG
arrange sequencer for TRIG track
choose bpm and play the sequencer at desired pace
yes, but 3 is not necessary. also muting the CLK track output will halt the clock. two at same time will “double” the speed.
is there a way to control sample speed and direction in the sequencer?
or is that only possible in tape mode?
I’m working on a new mode that (if I get it to work) will make it possible to sequence pitch and direction, so later on hopefully.
(but there is a bug… if you program a trig mode pattern and change to tape mode, the tape mode will play that pattern as pitch, but it´s really not suppose to work like that and the pattern is locked for editing, so well…)
that glitch could actually be useful, i’ll check it out
i have to reiterate, the sound quality of this module is superb…having dynamic control of pitch along with the volume (which is already in this version) would make PRGM even more playable
if you have any other feature ideas feel free to bounce them off me here or via pm
now i’m getting greedy
is there any way to divide the clock instead of doubling and multiples?
bloating the seq track length works but its much harder to slow down than speed up
the internal clock is only ment as a tool for quick checks, thus the coarse resolution of this mode. however it would be easily to add a couple of slower bpm values.
here’s an example of the weirdness this app spits out
I really love it
hm… so i’ve tried these steps but i’m still totally in the dark about how to get started using this app!!! @Test2
do you have to have samples loaded or is there a way for it to sample incoming audio?
i spent hours with this today and still can’t even get a peep out of it!
Is there any step by step tutorial anywhere about how one might sample and playback any kind of sound? or even just process incoming sound in real time?
sounds like some fast bpm rates there!
a short run-down on how to get sound, some basic scenarios
using sequenced samples
- have a look in this thread on how to convert wav files, place files in …/samples folder
- sw1+enc4, set a track to ‘trig’ mode
- enc2, raise the fader of this track to MIX
- mode switch, flip page and edit the sequencer
- on this page, enc1 sets sample per step, enc2 sets volume per step, enc3 selects step
a clock is also needed to drive the sequencer
- sw1+enc4, set a track to ‘CLK’ mode
- connect a pulse, saw+ or trig type clock signal to cv-iniput 1
on the MIX page, pressing sw3 will restart all tracks on step 1
using looped playback with tape mode
- sw1+enc4, set a track to tape mode
- enc2, raise track level to MIX
- sw1+enc3, select a sample
- sw3 start/stop playback
- enc3, change playback speed, negative values plays in reverse
process external signal with delay
sw1+enc4, set a track to [input] mode
sw2+enc4, set track input to In1 or In2 to input external audio
sw3, raise the input level (this should be max per default but I think I forgot to set that correctly in the last version…)
sw1+enc4, set another track to [dly] mode
sw2+enc4, set input to be the input-track above
sw2, raise track level to MIX
sw3, set delay time
enc1+sw3, set delay feedback,
several [dly] tracks can be set up like this. it should be possible to set [dly] input to external directly, but I think this got removed temporarily because I’m reworking some stuff,
tip! by using the trk inputs it is possible to connect delays in series
using aux send
- enc1, raise the aux send level (sw1+enc1 sets pre or post fader)
- setup a [dly] track and it´s input set to aux
tip! the INV-inputs will reverse the phase of the incoming signal
tip! more aggressive feedback can be achieved by using aux send to send signal output back into the track itself
using mute groups
- sw4, prepare mute of a MIX send
- on the MIX page, sw4 will mute all the prepared track sends, pressing sw4 again will swap muted/unmuted sends, holding mode switch and pressing sw4 will reset/remove all mute sends
using track output mutes, can be useful when you want to mute the incoming signal to a delay without muting the delay trail
on the sequencer page
- sw4 mute
- sw3 solo, the mode led will light whenever solo is active on any track
mixing with groups
- sw1+enc2 to set send destination, MIX, group 1, group 2 or combinations
- on MIX page, enc1 sets group 1 level to MIX, enc 3 sets group 2 level to MIX, enc 2 is the master fader setting the MIX level to external ouputs 1&2
yep until i get slower clock from cv i decided to play around with it as is
@dspk …maybe i will make a crude video sometime this week
you might figure things out before i do
I am trying this tomorrow. Looks really promising @Test2, thanks for your hard work!
people those tips posted above are gold!
i’ve only scratched the surface but the new group feature allows for very creative signal mixing
stacking delays is discussed earlier in this thread (or was it on the old forum?)
you can breach into reverb territory if you tweak things a certain way
also something that might be confusing
with a few exceptions (that i cant recall off head) this is the mapping of the encoders
@test2 can correct me if i’m wrong or if this is a glitch
I´m not sure but I number the encoders like this:
also, there is a typo above on the [dly] mode, delay time is enc3 and feedback is sw1+enc3. I double-checked and it is possible to “patch” an external input directly to a track in [dly] mode, testing 0.1.2 today and running 5x [dly] tracks at the same time is possible with this kind of setup.
mute/solo track output can still get weird in some scenarios, but this is an easy fix later on.
also, i think i mentioned this before but dont know if it’s possible…would adding more tracks push the hardware/software beyond stability? or would it mean sacrificing other features?
8 is fantastic but mid patch I sometimes wish i had 3-4 more slots for delays and feedback
when the dsp is “tipping over” the possible number of calculations per sample the pitch will drop, sometimes this can happen shortly, like at a certain step or it will drop to half immediately. before the new mix features we’re added I recall running 6 delays, but at the moment I´m more happy with mix groups etc and 5 delays max. the max number of tracks for this app is also constrained by the size of dsp on-chip memory.
So I am trying to make it work but as a starter does it really have to be big endian files? All editors convert to little endian, which seems to be the norm… then when I use a little endian file your app will start, whereas with a big endian file it gets stuck in the uploading process.
Anyway, I tried with the little endian file but have seriously not found how to load it, how to start the sequencer and so on. Somehow it seems close but I just don’t get it and I did follow your procedure, probably in a wrong way though. Thanks for any help!
Edit: I now managed to get some white noise in there, although unintentionally (I suppose it is due to the format). I’ll let go for now, @glia if you can somehow explain a bit more how you made working files for this it’ll be greatly appreciated!
here is a quick tutorial on loading your own samples, based on how I have got it to work.
only two formats have been tested to work, 16bit/96k and 32bit/48k, I have so far only tested mono files, not sure what will happen with a stereo interleaved .wav
folder structure on the card
the whole folder structure that is included in the .zip-files is necessary, sample files are to be placed in this folder:
(there should be a short test sample called trig.prgm already included in /samples folder, as part of the zip-file)
samples needs to be in raw pcm file format (also called headerless), so a .wav file must be converted. One way of doing this is using an app called ffmpeg (it´s free and seems to work on a lot of platforms). run ffmpeg from terminal/command prompt. here are two examples where I placed the ffmpeg exe in the same folder as my .wav files,
a 16/96k mono .wav:
./ffmpeg -i 909oh.wav -f s16be -acodec pcm_s16be 909oh.prgm
a 32/48k mono .wav:
./ffmpeg -i loop3248k.wav -f s32be -acodec pcm_s32be loop.anything
the .wav files I have tested with have been created with the bounce tool in ProTools 10.
hope this can be to some help.
EDIT, one more thing… loading samples takes a looooong time, a 300k file takes approx 90 seconds to load in this version.