Pd-list March 2004

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158 participants
381 discussions

ah ha.. OSXPM
by Christopher Mitchell 08 Mar '04

08 Mar '04

well, I did find one reference in the archive to OSXPM and got it to work to remove these packages. will try to get things working on my

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Re: [PD] Recording to Disk
by Johannes Taelman 08 Mar '04

08 Mar '04

Yes, but I think the writesf~ help patch ("pd/doc/5.reference/help-writesf~.pd") is the authorative example. But hey, in that example there is a [del 1000] object (del is shorthand for delay) connected that can stop the recording automatically one second after starting the recording. Did you copy that part too? j#|@ Owen Cartwright wrote: > Do you think you could send me a pd example of writesf~ so I could compare > it with my own? > > many thanks > > > ----- Original Message ----- > From: "Johannes Taelman" <j0(a)advalvas.be> > To: "Owen Cartwright" <ocartwri(a)bennington.edu> > Cc: <pd-list(a)iem.at> > Sent: Monday, March 08, 2004 11:50 AM > Subject: Re: [PD] Recording to Disk > > > >>A small off-topic story: >> >>Last week I was doing a 8-channel live multitrack location-recording, >>but I forgot my cubase dongle, so I couldn't use cubase :o >> >>So I made a quick pd patch with writesf~ :) That worked out fine, no >>single glitch in the recordings. >> >>The only minor problem was the writesf~ appeared to write files bigger >>than 2GB, but readsf~ couldn't open them afterwards. So I made sure >>everything was recorded in chunks smaller than 2GB. >> >>So writesf~ is definately able to record more than a second [on WinXP >>with hammerfall dsp soundcard]. Is sound I/O working stable on your >>system? Is there free space on your hard disk? Does the folder you're >>recording to, exist? Is there any error message in the pd command >>("dos") window? >> >> j#|@ >> >>Owen Cartwright wrote: >> >>>Hello, >>> I am a new user to the wonderful world of PD and am having some >>>trouble recording to my hard disk. Using the 'writesf~' command will >>>only allow me to record around one second of audio. I have also tried >>>using sfrecord of the zexy library but with no success. >>>I am running the latest verson of PD on WindowsXP and using a Realtek >>>AC'97 sound card... any suggestions? thanks, >>> Owen C. >> >> >>_______________________________________________ >>PD-list mailing list >>PD-list(a)iem.at >>http://iem.at/cgi-bin/mailman/listinfo/pd-list > >

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uninstallation on OSX?
by Christopher Mitchell 08 Mar '04

08 Mar '04

Howdy, list, I've just run the Package installer for PD on my powerbook but I want to remove it from here altogether and put it on my Cube. Is there any means of simple uninstallation? The package certainly had no trouble throwing files all over the place. This is one of those examples where using the "package" installer is really lame when compared to an application package (single file/folder)... All help here is appreciated! Looks nice a nice alternative to Max but I need it off this machine :/ Yours, Chris

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Re: [PD] band-limited square wave
by matthew jones 08 Mar '04

08 Mar '04

interesting arguments larry. But you're definitely wrong....! ;) how amusing. > For the next experiment, imagine that you are sampling a non-filtered > square wave through an ADC. In other words, there is no anti-aliasing > filter in place. What do you expect the digitized samples to look like in > this case? Would they not be exactly the same as a square wave constructed > entirely in the digital domain? well, this is where aliasing will occur. aliasing MEANING more than one analogue signal will give rise to the observed digital signal. aliasing is MEANINGLESS if you're already in the digital domain. it refers to going from a high bandwidth to a lower bandwidth. > The reconstruction filters in an DAC are there to remove all analogue > frequency content above nyquist, without affecting the frequency content > below nyquist. > So, yes it would ripple, but not in the right way. no, this would be in the right way! the ripples are due to filtering out all above Nyquist as you say, which is what we want...!!! that IS naturally bandlimited. > A bandwith-limited square digital osc sould give the same output as > ADC'ing a squarly analog square wave. So antialiasing filters are required. ?? what ?? sorry, don't get u > And it's not the only way to get aliasing: nonlinearities (waveshaping), > all kinds of modulation (fm, am,...), and much more. >Digital sine oscillators and digital filters are bandwith-limited by nature.(<-AGREED) > First, set your sampling rate to 44100Hz. Now take a digital sine > oscillator, say [osc~] in PD and set its frequency to 30000Hz. > What do you expect to hear from the DAC? A sine wave at 30000Hz, even > though that frequency is higher than fs/2? No, you will in fact here > an aliased sine at 14100Hz - do you agree. But wait, we constructed this > signal entirely in the digital domain, and yet we have aliasing! well, in DSP terminology, this isn't aliasing. of course in the digital domain you can't synthesise ANYTHING higher than nyquist, it's impossible. if your own code runs functions that attempt to do that, but end up doing something else (ie return numbers that represent tones that have the same reflected frequency as the equivalent aliased signals) then that function is not producing aliased outputs, but correctly generating frequencies that are lower than YOU intended. again, aliasing only applies to going from high bandwidth to low bandwidth, so if you're sample rate hasn't changed, you're not getting aliasing. this is getting pedantic, I can see what you're getting at but at the end of the day:- constructing a bunch of numbers in the digital domain cannot produce 'aliased' outputs because the bandwidth has not changed. reconstruction will 'perfectly' bandlimit the output (although perfect sinc functions are impossible) everytime. Matt -=-=-=-=-=-=-=-=-=-=-=-=- http://www.loopit.org/ -=-=-=-=-=-=-=-=-=-=-=-=-

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one other thing, bout sending info to a message
by Adrian deCleir 08 Mar '04

08 Mar '04

Ok I dont know if this is possible its sounds a bit weird, but In a project I have where people play music together over the internet or a network there is several netsend connections, each one for something different, like |connect 10.42.50.10 3001| for volume, then |connect 10.42.50.10 3002| for the actual note, and | connect 10.42.50.10 3003| for pitch bend or something different like that, this means I have about 10 netsends overall , all for different features. This means when Im connecting with someone different I have to write in the IP address 10 times for everything to work, this project is for college and a problem like this will seriously make the project look bad, any ideas? as in is there a way of just writing the IP once and send it to all the different |connect| messages so it'd be like this you write in the IP here |10.42.50.10 | and it sends it out to all the other ones like it shows there | | |____________________________________ | |_________________________ | | | | |connect 10.42.50.10 5002| |connect 10.42.50.10 5000| | |connect 10.42.50.10 5001| cheers if any one can help

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Niagara question
by Jorge Cardoso 08 Mar '04

08 Mar '04

Hi, I have a question concerning the [niagara] object (please see attach). The question is why doesn't the second [route] match the 'r'? And what should I do to make it match it? It seems strange because if I change the number '2' in the message to anything that is not a number it works as I would expect it... Thanks, jc --- Jorge Cardoso <-> jorgecardoso(a)ieee.org Centro de Investigação em Ciência e Tecnologia das Artes (CITAR) Univ. Católica Portuguesa Centro Regional do Porto Rua Diogo Botelho 1327 4169 - 005 Porto Portugal Tel: 226196200 Ext: 133 "Whether you believe you can, or whether you believe you can't, you're absolutely right." - Henry Ford

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Recording to Disk
by Owen Cartwright 08 Mar '04

08 Mar '04

Hello, I am a new user to the wonderful world of PD and am having some trouble recording to my hard disk. Using the 'writesf~' command will only allow me to record around one second of audio. I have also tried using sfrecord of the zexy library but with no success. I am running the latest verson of PD on WindowsXP and using a Realtek AC'97 sound card... any suggestions? thanks, Owen C.

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RE: [PD] swapping from netsend to OSC
by Adrian deCleir 08 Mar '04

08 Mar '04

Hmmm, I think you'd want to be a super musician to be able to play constantly out of time like that : (, I tried both TCP and UDP and, and although TCP was slower it was steadily slower, whereas UDP might sometimes might have been perfect other times it was awful, e.g. 3 notes might come in at the same time after about 5 seconds. Ah well cheers for the help Adrian deCleir wrote: > At the moment I have several number boxes attached to different netsend > objects, everything goes through a different port number, like for volume I > use port 3001, for sending pitch bend I use port 3002, etc and as I change > these numbers the other person receives them instantly, would this be > possible with OSC??? Since OSC uses exactly the same network that netsend uses and has a bit more overhead it can only be slower. The real problem is that the realworld internet is not instantaneous and packets can be delayed by unpredictable amounts up to many seconds - try traceroute to some site to see how long it can take. Using UDP for the connection instead of TCP improves latency by dispensing with some handshaking and error-detection. OSC theoretically supports timestamps, so you could introduce a known delay for everyone...if the beat were set to a multiple of this delay, you could play out of time but still be in sync...probably only a certain kind of techno music would be acceptable to play like this though.#-) Martin

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swapping from netsend to OSC
by Adrian deCleir 08 Mar '04

08 Mar '04

Hi, Ive been doing a project that allows people to play music in real time over a network or the internet , etc. I have this working by sending midi messages over the net using netsend, although this works perfectly with a LAN or a really high speed internet connection , there is alot of latency with normal internet connections. I had a look at OSC before I started this project and found it difficult to understand and use, there was little tutorials for it too, but my question is, would it be a big job to change how my program works if I used OSC instead of netsend/netreceive? At the moment I have several number boxes attached to different netsend objects, everything goes through a different port number, like for volume I use port 3001, for sending pitch bend I use port 3002, etc and as I change these numbers the other person receives them instantly, would this be possible with OSC??? I'm not sure how to get started with it Thanks for any help if you can cheers Adrian

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Re: PD-list digest, Vol 1 #1017 - 14 msgs [OT]
by Trevor Agus 08 Mar '04

08 Mar '04

Yes, I've had problems with aliasing too, and would be grateful for any ways to get around it. Just to check my understanding of it: If I try to synthesize a 66150Hz sine wave at 44100Hz sampling-rate (after doodline on a bit of paper) the samples end up looking like a 22050Hz sine wave. As Matt points out, the obvious way to get around this is not to try and create such high frequencies. But unfortunately, square-waves and saw-tooth waves (for example) have ridiculously high harmonics, whether you want them or not. When you represent them at a certain bit-rate, all the harmonics above half the sampling rate gets aliased somewhere where you don't want them. So how does this link up with Matt's suggestion that aliasing doesn't happen in the purely digital domain? Well I guess in a sense the square and saw-tooth waves in my head/formulae are analogue (i.e. infinite bandwidth.) and realising them in the digital domain involves quantising and sampling from that. So it is certainly aliasing. The only way I can think to get around it is to build all complex tones from sine-waves, which is a little calculation intensive, and doesn't solve the quantisation problem. Trevor PS I tried to pose that as a question, but now suspect I've just joined an argument... Rather than delete it, I've marked it 'off-topic' as an acknowledgement of that. Please don't flame... ----- Message: 5 From: "matthew jones" <m.jones(a)signal.qinetiq.com> To: "Larry Troxler" <lt(a)westnet.com>, "PD-List" <pd-list(a)iem.kug.ac.at>, "Johannes Taelman" <j0(a)advalvas.be> Subject: Re: [PD] band-limited square wave Date: Mon, 8 Mar 2004 13:40:55 -0000 interesting arguments larry. But you're definitely wrong....! ;) how amusing. > For the next experiment, imagine that you are sampling a non-filtered > square wave through an ADC. In other words, there is no anti-aliasing > filter in place. What do you expect the digitized samples to look like in > this case? Would they not be exactly the same as a square wave constructed > entirely in the digital domain? well, this is where aliasing will occur. aliasing MEANING more than one analogue signal will give rise to the observed digital signal. aliasing is MEANINGLESS if you're already in the digital domain. it refers to going from a high bandwidth to a lower bandwidth. > The reconstruction filters in an DAC are there to remove all analogue > frequency content above nyquist, without affecting the frequency content > below nyquist. > So, yes it would ripple, but not in the right way. no, this would be in the right way! the ripples are due to filtering out all above Nyquist as you say, which is what we want...!!! that IS naturally bandlimited. > A bandwith-limited square digital osc sould give the same output as > ADC'ing a squarly analog square wave. So antialiasing filters are required. ?? what ?? sorry, don't get u > And it's not the only way to get aliasing: nonlinearities (waveshaping), > all kinds of modulation (fm, am,...), and much more. >Digital sine oscillators and digital filters are bandwith-limited by nature.(<-AGREED) > First, set your sampling rate to 44100Hz. Now take a digital sine > oscillator, say [osc~] in PD and set its frequency to 30000Hz. > What do you expect to hear from the DAC? A sine wave at 30000Hz, even > though that frequency is higher than fs/2? No, you will in fact here > an aliased sine at 14100Hz - do you agree. But wait, we constructed this > signal entirely in the digital domain, and yet we have aliasing! well, in DSP terminology, this isn't aliasing. of course in the digital domain you can't synthesise ANYTHING higher than nyquist, it's impossible. if your own code runs functions that attempt to do that, but end up doing something else (ie return numbers that represent tones that have the same reflected frequency as the equivalent aliased signals) then that function is not producing aliased outputs, but correctly generating frequencies that are lower than YOU intended. again, aliasing only applies to going from high bandwidth to low bandwidth, so if you're sample rate hasn't changed, you're not getting aliasing. this is getting pedantic, I can see what you're getting at but at the end of the day:- constructing a bunch of numbers in the digital domain cannot produce 'aliased' outputs because the bandwidth has not changed. reconstruction will 'perfectly' bandlimit the output (although perfect sinc functions are impossible) everytime. Matt -=-=-=-=-=-=-=-=-=-=-=-=- http://www.loopit.org/ -=-=-=-=-=-=-=-=-=-=-=-=-

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Pd-list March 2004