Hello,

i am wrapping my head around how to control PD-patches sample accurately from a DAW?

With Camomile?

As this is a bit confusing, could someone help with some light please?

For example:

** How to build a sample-accurate envelope, - triggered and modulated by a DAW? **

I am using Linux, so cross-routing with Jack is an option, too.

Probably somehow with [vline~] or [vphasor~]? This is sample accurate in the next block, isn't it? i.e. with latency-compensation in the DAW it is on point?

Open Sound Control is not sample accurate, is it? Seems like the PD-sheduler does not handle OSC synchronous!?

( How to trigger a set of control-rate objects by an audio-signal sample accurate? What would be the port / the transition between the two?
This isn't possible, is it?
I guess it is, with [pd]{ [inlet~] [block~1] [unsig~] [outlet] } and [vline~]? )

Do I need to patch the whole envelope with audio-rate objects?
i.e. [vphasor~] instead of [vline~]?

What is the way to go
to merge PD with a DAW sample-accurately?
DAW automation <--> PD control- and audio-objects.
And audio-intermodulation of DAW & PD?

@lacuna Data on every DSP tick.
A sort of DSP timecode.
I think only 32-bit for Pd under Windows but 64-bit for OSX and Linux.
https://cdm.link/2016/11/free-jazz-now-ableton-link-sync-works-pure-data/
https://www.ableton.com/en/manual/link-synchronization-and-rewire/
[abl_link~] through Deken.
It could be a starting point?
David.

I... had... no... idea... Pd had Ableton Link support

I oughta read this forum more often

hjh

@whale-av thank you, good idea, I will try.
(Not using Ableton live at all)

as far as I see ths is not "data on every DSP tick". What did you want to say with this?
It works differently:
there has been this presentation at Linux Audio Conference 18:
https://media.ccc.de/v/lac2018-42-ableton_link_a_technology_to_synchronize_music_software

Not sure how accurate this gets. as there seems to be a lot of "average" "time filtering", "guessing" and it is about beats, not samples.

The ease of use seems really cool!

.
I am still confused about the scheduler of PD:

sample accuracy within Pure Data and the ports to the environment...
Message domain, audio-rate, timestamps, block-boundaries, scheduler, ect.
But I think that I will post another thread about this.
As this is essential to understand programming in PD and there seems to be a lot of confusion about it even for advanced PD users, because the perception is most of the times "allright it works", not being picky in exact timing.

@lacuna I haven't tried it, but thought it could be a starting point.
The help for the Pd object says that.......
"abl_link~ receives...... the current phase and beat time on each DSP tick....."
and as it's purpose is to keep all rewire programs in perfect sync that makes sense.
Pd can only execute control chains between dsp ticks, so everything that Pd needs to do with data from the object has to be executed (right down the chain) before the next tick starts..... so it should be accurate. All done after sample 64 and before sample no.65 . Of course if you try to do too much you could get dropouts.
Pd needs to run at the same samplerate and blocksize as the other programs of course, but you can go down to single samples in a subpatch if you need to.
Going sub sample can't be done. It is just a number at a time. If you want more accuracy you need to increase the sample rate, and if your sorce is not at that rate then you need to interpolate and will have more time accuracy but not sample accuracy.
I am hopeful it will get you what you are looking for.
David.