I am working with a Windows 7 64bit System on a Core i7, 16 GB RAM and an RME Fireface UC. I am using pd to play back samples triggeded by commands from the serial port (using a COMPORT object) from an Arduino board.

As of now, 40 ms is the lowest possible latency I can set in pd without experiencing glitches. Buffer Size in the Fireface control software is set to minimum (48 samples).

It appears as if ASIO and non-ASIO output is available in the pd-audio-settings. However. All output channels of the fireface are available as non-ASIO, but only a generic "ASIO Fireface USB" shows up as ASIO device. Using this device results in silence. Nothing shows up on the output channels of the TotalMix software.

Does anyone have ideas for getting this card to work in ASIO mode with pd? The measured latency of my system is just too high at the moment.

Hi,

I am trying to use soundfiler to read raw data from multichannel recordings. I have recordings with 1 to 4 channels and 8, 16 or 24 bits per channel.

read -resize -raw 65536 3 3 l somefilename table1 table2

works fine.

read -resize -raw 65536 3 3 3 3 l somefilename table1 table2 table3 talbe4

or

read -resize -raw 65536 3 3 1 l somefilename table1 table2 table3

however fails and I get the error
error: usage: read [flags] filename tablename...
flags: -skip <n> -nframes <n> -resize -maxsize <n> ...
-raw <headerbytes> <channels> <bytespersamp> <endian (b, l, or n)>.

I am absolutely sure that the files in my example contain 4x24 bit channels and 2x24bit+1x8bit respectively since I have written myself a conversion tool to extract my required channels before the use in pd. But reading them directly would be much more convenient.

Another question: is there a way to use spaces in the names of files/directorys?

Thanks for any advice!

TreibAir

P.S.:Just found out: trying to read a file with just one channel leads to the same problem

Hi,

I have and Ardiuno ethernet shield and try to send data from my Arduino to PD.

First I tried the Arduino ChatServer demo patch and modified it to print any incoming data to the serial console in the Arduino IDE which worked out of the box.

Unfortunately I can't transmit anything back to PD. I tried to send strings i.e. "client.write("String");" and "client.print("String");" but no results showed up in PD. The only thing that works is the connection counter on my netreceive object.

Has anyone ever successfully tried this?

Alex

Hi,

I am trying to use a RS232 DMX interface in PD. The serial port settings are 19200 bps, no parity, two stop bits. The comport object only allows me to send either no or one stopbit. Is there a possible work-around (i.e. manually sending the stopbit whenever needed?).

Thanks in advance,
Alex

Hi There,

I have patched a signal generator for an active noise cancellation experiment. Basically it consits of an osc~ which feeds a delay line. I use two delread~ to drive my outputs. One is not delayed at all, the second one can be set to 0-5 ms delay via a slider.

While checking my outputs with an oscilloscope I figured out that my delread~ does not delay the signal at all until I move the slider over about 1.5 ms. Above this level the delay increases proportionally. I have compensated it by adding 1.5 to the value set by the slider but this is obviously a bad work-around.

Any Ideas where this effect comes from?

Thanks to the help of hardoff and pulsed my patch is finally working quite well when I run pd from the command line using

sudo pd -rt -jack -channels 2 -send ";pd dsp 1" open /some/file.pd

However, I do not get how the -send option is acutally used. The documentation only states.

-send "msg..." -- send a message at startup (after patches are loaded)

How can I specify where the message is received. I have tried a receive node [r something] and the comand line option -send "something; 1", hoping that the "1" will be received.

What I want to to is to send a number to a specific node in my patch on startup.

Hi everybody,

I use PD as part of my PhD project. Since audio/haptic asynchronity is an issue I have to keep latency down. My test subjects have to press buttons and virutally instantly hear the sound they triggered. The sounds are stored in arrays and played back using tabplay~
I have just checked the delay between the press of a button and the resulting sound. The buttons are connected to my computer using a custom microcontroller board which is quite common at my university and send the trigger command to a PD comport object. The baud rate is set to 230400 bps and only one byte is needed as command. Latency of the entire button-microcontroller-serial output should be way below 1 ms.
For sound output I use a Lexicon Lambda USB interface at 96000 kHz and 5 ms latency.
When I monitor the delay between physical trigger and output I get latency times ranging from just under 10 ms to over 25 ms. More annoyingly, this latency is not stable but but changes every time I trigger.
Since my "hardware" latency should be just over 5 ms and PD adds up to 20 ms to this I assume PD interally processes things at a certain framerate which should be arround 50 Hz or 20 ms. Is it possible to increase this framerate in any way?
The system runs on Ubuntu Studio 8.04, it would be possible to switch another audio interface (RME Hammerfall).

Any ideas are greatly appreciated.

Alex

Hi there,

the project I am currently working on involves synthesis of short samples and their playback. I intend to use PD as frontend for my synthesizer as well as for the playback of sounds. The synthesizer itself however is implemented in Matlab. I therefore should transmit my parameters to Matlab somehow. It was quite easy to generate a TCP object in Matlab and to receive the data in PD via netreceive. However, the other way round won't work. Both the PD netsend object and Matlab appear to be clients which require a host. While netreceive seems to provide this host for Matlab, netsend somehow doesn't. Has anyone every had this problem before or is there an alternative to a TCP/UDP connection?

Thanks in advance,
Alex

Hi there,

I'm pretty new to PD and try to use it for a research project. I have built a very simple sample player which reads and plays back a WAV File according to a number which is received via OSC.
The problem is that as soon as a new file is triggerd the playback of the old sound stops. I have 16 sounds in total which might be triggered in quick succession so that I would actually need a patch that allows the playback of more than one sound at a time.
My idea is to build a parallel structure of 16 sample players and depending on the number on the input one of this 16 players is triggered. How can I do that?

Thanks,
Alex

UPDATE: upgrading to Version 0.43.4-extended solved the issue. ASIO is corretly supported and I am running pd with 5 ms latency.