Hi,

Three questions

1. Does anyone know what command(s) I use to control a synth patch in Pure Data using the keyboard of my laptop?

2. Another thing I am wondering about is how I might control a synth with a piano. I have a midi to usb cable that I can connect from my laptop to the keyboard.

3. Are there any great synth patches available for download to experiment with?

thanks

or is it [ctlin] ? i never remember which. anyway..either [ctrlin] or [ctlin] to get controller data.

hey, thanks alot, that ARPodyssey looks great.

It has a very graphic interface, so I'm unsure of how I might get the midi keyboard to work on it. I have a midi cable that connects from my keyboard to my laptops usb port.

PD picks it up, and I set it as a MIDI input in the Media/MIDI Settings, but there's obviously something else that needs to happen because it's not picking up the keyboard when i play on it.

I tested the MIDI device, and I can get my piano to run through with the monitor box checked. It plays through the first 2 channels...do I have to specify that within the ARPodyssey synth patch?

thanks

ah, nevermind that, i got it working. are there any piano simulator sort of synths that you can play a lot of notes at the same time? It doesn't seem like I can do chords with this synth.

search the pd documentation for polysynth

Great, thanks.

Do you know how to extend the length of audio that is captured in a sample in pd?

In the sampler.loop example, they have a sampled voice that is 1 second. I can change the audio file, but I would also like to change the length of audio playback from 1 second to 7 seconds..is that possible?

thanks for the help

maybe this is what you want?

http://puredata.hurleur.com/sujet-643-sample-player

yeah, that's sort of it. that will get me the loops i want, but I need to get a way to control the frequency and speed using the changing x/y coordinates based on a camera's tracking. Is there a way to link changing x/y coordinates to the sliders on this patch so that the numbers will adjust the frequency and speed up and down?

right click on a slider to bring up its properties. the send and recieve settings will show where the data is coming from / going to.

awesome, is there a command to round off numbers? I am multiplying the coordinates to get realistic slider values, and I would like to clean it up and get rid of all of the decimals.

ok, at this point i'm going to have to tell you to go read the documentation - that's what it's for.

if you can't find it within a couple of hours, report back.

edit..actually, no, i'm not that busy that i can't help you. the object you need is [int].. but you should definately read as much of the documentation as you can.

yea, it's rude of me to keep throwing up trivial questions. i do apologize, and i really appreciate the help as I have until the 11th of december to get this figured out for my computer art project.

I have got through the manuals and examples quite a bit today, and i am running into another problem:

I'm trying to read sound files and manipulate their frequencies and speeds via the coordinate readouts from my camera. My goal is to be able to trigger each sound with a conditional statement involving my x/y coordinate readouts (i.e. if x <= 4 then read this file1.aif, if x >= 4 then read file2.aif). I have been able to accomplish this to an extent, but I only want the sound files to be reloaded every 30 seconds or so. As of now, my program is constantly reloading the sounds so there is no time to experiment with the sound before it's either reloaded or loaded the different sound.

my computer's camera is sending x/y coordinates to my pure data program. I have multiplied and rounded the numbers using the int that you suggested..this way, the values in the x and y category will range from 0-9. from there, I am connecting those numbers to the [moses 4] command (this sends data that is less than the numeric value to one side, and data that is greater than to the other side). each of my [read *audio file path*] are connected to a different output of the [moses] command, so, depending on my x coordinat location (0-9), that will determine what sound is played. I believe i need to use a combination of a delay and a timer with my moses in order to control the number of times it is telling the program to read the sound files...i can't wrap my head around how to do it though...the example counter that counts from 1-10 seems like it might be the way to go, but i need a way to implement the delay with the moses so that the moses command only sends out data every so many seconds rather than constantly..

does that make any sense?

i can attach the pd file..that might offer more info...thank you again for the help

instead of [moses 4], have a look at [> 4]

and then, to stop gettign multiple readouts, maybe the [change] object will come in handy.

I got it working using the spigot command along with a [metro 5000] attatched to a [delay 100] which checks the spigot on and off, so the data is only allowed to pass through the spigot during the time between the metro firging a bang every 10 seconds and the delay unchecks the spigot after just after every bang.

One of my last problems is how to define the length of the audio samples that I'm using. Right now, every sound I load up only plays in a one second loop regardless of how long the actual file is. I would like to be able to run the entire audio file on a loop. Most of them aren't too long..probably 7 seconds on average. I'm using the sampler.loop example to load my audio files, and I can't figure out what is telling it to only play one second of every file. In the easysample.pd file that you linked to earlier, I can load up an audio file, and it will loop the entire sound..

do you know what makes the easysampler dependent on the audio file for length as opposed to the sampler.loop which forces the audio file into an already determined length?

thanks

Could I use a -resize flag in combination with the readtab4~ command in order to automatically resize the array table that the sound file is read to?

I haven't really seen the actual examples you're talking about, but i've included an abstraction called simpleSampler. You could easily modify it so that it has another outlet that fires a bang once the sample has played totally (inside the sampler you'll find a number labeled "PLAY TIME", you can use this with a [del] object to trigger the bang). With a [spigot] you could then link the new outlet back to the play inlet.

Hope this helps,

Domien

http://www.pdpatchrepo.info/hurleur/simpleSampler.zip

I would love to take a look at your example, but I can't get it to open. when I unzip it and save to desktop, it appears to have no extension. I tried renaming it to a .pd file, but it wouldn't load in pd.

Mmm... posting something seems tricky... Maybe this works.

http://www.jittered.com/pd/simplesampler.zip

D.

@lapras53 said:

Could I use a -resize flag in combination with the readtab4~ command in order to automatically resize the array table that the sound file is read to?

That's exactly right.

You haven't posted any code yet have you? You might need to do that if you you're still struggling to meet a Monday 11th deadline. I should be around this weekend mostly.

Possibly study the use of a message like

[1, 44100000 1000000]
|
[vline~]

(that's digit ONE, a comma, and 44100000 and 1000000, no commas)
don't worry that Pd immediately converts these to SI notation.

which is a catch all ramp generator. It says to vline~ start ramping up at a rate of
44100 every 1000 milliseconds.. for a long long time.

This solves most sample playback scenarios where you know only the sample rate but not the length of each file beforehand.

thanks for the help. would I place a command like the vline~ in the upper input of the [tabwrite] command?

I will attatch my file. It's very messy as I have been trying out a lot of things. I just need to figure out how to define the array in terms of the sound that is being looped.

http://www.pdpatchrepo.info/hurleur/FinalProject_Practice2.pd

Yes, [vline~] is used to index the sample table instead of the [phasor~].
I think the command you are searching for is the -resize argument for soundfiler, if you put that in your load messages things should get better. Stop sending things to the chunk size (scaling), unless you also want to modulate the playback rate. If you use the size of the file as the scale then a sweep from 0 tp 1 will play the entire file. Multiply that phasor by a fraction to get slower playback speeds. Look at the example Hardoff told you to, it's in there.