[LAD] [PD-announce] smeck (6ch guitar processing patch) released]
Hi, something for the guitarists around. This is the patch, that Miller Puckette explained during the workshop at LAC2008. Ciao -- Frank - Forwarded message from Miller Puckette - Date: Mon, 24 Aug 2009 20:52:33 -0700 From: Miller Puckette To: pd-annou...@iem.at Subject: [PD] [PD-announce] smeck (6ch guitar processing patch) released Hi all, I've finally done a good enough job of cleaning up the 'smeck' patch (seen at Pd conventions 2007 and 2009) to consider it releasable. This patch takes a 6-channel signal from a separated guitar pickup (such as the Roland GK series) and does a variety of cool things to it. Smeck is available from: http://crca.ucsd.edu/~msp/ cheers Miller ___ Pd-announce mailing list pd-annou...@iem.at http://lists.puredata.info/listinfo/pd-announce ___ pd-l...@iem.at mailing list UNSUBSCRIBE and account-management -> http://lists.puredata.info/listinfo/pd-list - End forwarded message - ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
On Mon, Aug 24, 2009 at 6:13 PM, Ralf Mardorf wrote: > I don't know actual Peter Gabriel recordings, but I bet he still uses > old synth, e.g. the Fairlight, especially for pad sounds. he doesn't. ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
On Mon, 2009-08-24 at 21:10 +0200, Ralf Mardorf wrote: > Virtual synth often tend to make the mix muddy, when > playing pad sounds, because the polyphony isn't limited, every released > note is able to end the complete release decay. For the Oberheim some > notes are cut, while the wanted notes still can play the release decay. > Where do the newly assigned voices start their envelopes from when they are stolen from decaying voices? Say "the glass is half empty", will they then: a) "Empty the glass" and start over with a fresh attack from scratch. Synced so to say. b) Contnue from the level they were at when re-assigned, only that now "the glass is half full" instead, and the attack will reach max almost immediately. I think the latter could be more expressive when there are no more voices than you can easily direct with a single two-handed chord, to get in control of the stage again. Six voices would pretty good for that, but I remember five like the Prophet had was annoying. Or that at least I got lost fighting my own clumpsyness. Hmmm ... Split 2+4 comes to mind as well. > A selectable limit for polyphony might be a feature, that should become > more common again, not only for virtual analog synthesizers. ++ ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
Nick Copeland wrote: > will not sound like an Oberheim. PS: And there's no need to sound identically. Even if a programme should program something absolutely new, not an emulation, the 'Prophet *5*'-'Matrix *6*'-Effect of having less voices, could add some charm. Btw. I never used the Alsa modular synth, but even an intelligent way like Fons described, might be fine for my needs, if it's possible to reduce the polyphony to 6, 5 or less voices. ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
Nick Copeland wrote: > Voice allocation really depends on what you want the virtual synth to do. > If you want it to sound like the original then it should use a similar > algorithm, > if you want something that sounds better than or like the original > then for > something like an Oberheim, it will probably not be the voice > allocation that > is lacking on the emulator, more likely the overall quality of the > sound that will > prevent the emulator from being as good: whatever you do with voice > allocation, > if the emulator does not have Oberheim filters and oscillators it will > not sound > like an Oberheim. I can sample some sounds of my Oberheim and make soundfonts and just need a soundfont player, that is able to emulate the voice allocation. A lot of sounds can't be emulated by algorithm¹ or layered samples, but this isn't what I need, not all sounds are using features that can't be sampled. Btw. there are very good emulations for the ARP step sequencer sounds available as proprietary VST or modern external, stand alone synth. Tonight I'll record some sounds of my Oberheim and then try to work with swami to make a soundfont. Theoretically I don't need any virtual synth, but because on my computer there's too much MIDI jitter for external equipment, but no MIDI jitter for virtual synth, band-aid should be to sample some of my synth. Might become funny for long vector synthesis sounds, that I'll also sample. I guess I have to spend 20,- € to pimp my RAM from 2 GB to 4 GB. Ralf ¹ resp. not today or just by some proprietary plugins ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
Voice allocation really depends on what you want the virtual synth to do. If you want it to sound like the original then it should use a similar algorithm, if you want something that sounds better than or like the original then for something like an Oberheim, it will probably not be the voice allocation that is lacking on the emulator, more likely the overall quality of the sound that will prevent the emulator from being as good: whatever you do with voice allocation, if the emulator does not have Oberheim filters and oscillators it will not sound like an Oberheim. The algorithm was also different between synths, many used a FIFO buffer: if the voices were all allocated then the oldest voice was stolen. Some would steal the lowest note pressed, some would steal the highest note. You will eventually notice which algorithm it uses based on what you are playing - each algorithm has its defects. Siel did have an allocation algorithm that was supposed to be superior in that stealing voices was not that noticible. Never seen the code but it apparantly stole a note from somewhere in the middle so the player never lost either extremes of the scale. I've tried this as well and the results are a noticable improvement on FIFO. Pure FIFO stealing is a cheap algorithm however it perhaps does have the most noticable flaws when being played. VIrtual synths can easily improve on this, emulators can't, really, and still be something that approaches the original. If you want an virtual synth to sound like an OB-X the it will have to steal notes in the same way. Regards, nick. "we have to make sure the old choice [Windows] doesn't disappear”. Jim Wong, president of IT products, Acer > Date: Tue, 25 Aug 2009 00:13:36 +0200 > From: ralf.mard...@alice-dsl.net > To: ha...@opensound.com > CC: linux-audio-dev@lists.linuxaudio.org > Subject: Re: [LAD] Selectable limit for polyphony of virtual synth > > Hannu Savolainen wrote: > > Albert Graef wrote: > > > >> hollun...@gmx.at wrote: > >> > >> > >>> One obvious question there is: > >>> what should the synth do when it reaches the limit? > >>> There are several things that are possible and afaik implemented in > >>> synths. It could drop the first note played, or the highest, or ... > >>> > >>> > >> Well, that's called voice stealing. Most synths do it, if they don't > >> have dynamic voice allocation. Usually, you assign voices in a round > >> robin manner, and the oldest note has to go when you're running out of > >> voices. > >> > >> > >> > > Ideally the synth should use some kind of priority mechanism when > > stealing voices. Killing the oldest one is not the best way. For example > > some kind of psychoacoustic algorithm could be used to find voices that > > are masked out by the other voices playing at louder levels. Some voices > > may have decayed to inaudible levels or their pitch may be close enough > > to the new note to be played. > > > > Best regards, > > > > Hannu > > Hi Hannu :) > > this is a good idea to cover unwanted cutting. "My problem" is, that > most virtual synth have enough voices ;). I would like to have the > effect of old synth, e.g. listen to Peter Gabrial's pad sounds. It's > wanted to hear the cutting, because it has a musical function, it > produces ambience. My fault was, that I only referred to the elimination > of muddy sound by notes with long release times, I've forgotten to bring > up the musical function. > > I like to have the effect that chords will be cut by new cords, similar > to the effect for monophonic sounds, with one difference, sometimes one > or two notes shouldn't be cut. There are some very good synth with a > polyphony of 5 or 6 voices, e.g. the Prophet 5 or the Matrix 6, btw. I'm > using a Matrix-1000 (the 1000 is for 1000 sounds in the memory, resp. I > didn't check if the battery is still fine, it might be possible that the > 200 RAM sounds of my Matrix are lost ;)). > > For most of my external synth and 'virtual' synth I'm missing this > effect, they never run out of voices. It's bad not to have enough > voices, but if you have enough voices than some charm gets lost. > > I don't know actual Peter Gabriel recordings, but I bet he still uses > old synth, e.g. the Fairlight, especially for pad sounds. > > Cheers, > Ralf > ___ > Linux-audio-dev mailing list > Linux-audio-dev@lists.linuxaudio.org > http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev _ With Windows Live, you can organize, edit, and share your photos. http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gallery-edit.aspx___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
Fons Adriaensen wrote: > Or some _explicit_ feedback from somewhere downstream the patch > telling the voice allocator that a particular voice has decayed > far enough to be a candidate for re-use. My exploratory designs > for AMS II (gathering dust since four years) did exactly that. This "was" a very good mechanism at the times when the first synth were able to play different sounds for different MIDI channels, but tone generators were to expensive, because of technical limits, in the 80ies there were no DSP chips, 4MB RAM were very expensive :D etc., I still have got a 40 MB hard disk, that was more expensive than the complete dual core PC I've today. It's easy to understand that your AMS II is gathering dust :). I had to dust my Oberheim ;). It would be suspect if somebody would write to the list, that e.g his Linux synth don't have enough voices. I often wonder why a lot of synth emulation don't sound as good as the original synth, even if comparing a held note sounds identical to an original synth. They differ in their behaviour, when playing a song, but not by comparing single held notes. To limit the polyphony today seems to be more important for some usage, than thinking about not having enough voices :). Cheers, Ralf ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
Hannu Savolainen wrote: > Albert Graef wrote: > >> hollun...@gmx.at wrote: >> >> >>> One obvious question there is: >>> what should the synth do when it reaches the limit? >>> There are several things that are possible and afaik implemented in >>> synths. It could drop the first note played, or the highest, or ... >>> >>> >> Well, that's called voice stealing. Most synths do it, if they don't >> have dynamic voice allocation. Usually, you assign voices in a round >> robin manner, and the oldest note has to go when you're running out of >> voices. >> >> >> > Ideally the synth should use some kind of priority mechanism when > stealing voices. Killing the oldest one is not the best way. For example > some kind of psychoacoustic algorithm could be used to find voices that > are masked out by the other voices playing at louder levels. Some voices > may have decayed to inaudible levels or their pitch may be close enough > to the new note to be played. > > Best regards, > > Hannu Hi Hannu :) this is a good idea to cover unwanted cutting. "My problem" is, that most virtual synth have enough voices ;). I would like to have the effect of old synth, e.g. listen to Peter Gabrial's pad sounds. It's wanted to hear the cutting, because it has a musical function, it produces ambience. My fault was, that I only referred to the elimination of muddy sound by notes with long release times, I've forgotten to bring up the musical function. I like to have the effect that chords will be cut by new cords, similar to the effect for monophonic sounds, with one difference, sometimes one or two notes shouldn't be cut. There are some very good synth with a polyphony of 5 or 6 voices, e.g. the Prophet 5 or the Matrix 6, btw. I'm using a Matrix-1000 (the 1000 is for 1000 sounds in the memory, resp. I didn't check if the battery is still fine, it might be possible that the 200 RAM sounds of my Matrix are lost ;)). For most of my external synth and 'virtual' synth I'm missing this effect, they never run out of voices. It's bad not to have enough voices, but if you have enough voices than some charm gets lost. I don't know actual Peter Gabriel recordings, but I bet he still uses old synth, e.g. the Fairlight, especially for pad sounds. Cheers, Ralf ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
On Mon, Aug 24, 2009 at 11:25:17PM +0300, Hannu Savolainen wrote: > Ideally the synth should use some kind of priority mechanism when > stealing voices. Killing the oldest one is not the best way. For example > some kind of psychoacoustic algorithm could be used to find voices that > are masked out by the other voices playing at louder levels. Some voices > may have decayed to inaudible levels or their pitch may be close enough > to the new note to be played. Or some _explicit_ feedback from somewhere downstream the patch telling the voice allocator that a particular voice has decayed far enough to be a candidate for re-use. My exploratory designs for AMS II (gathering dust since four years) did exactly that. Ciao, -- FA Io lo dico sempre: l'Italia è troppo stretta e lunga. ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
Albert Graef wrote: > hollun...@gmx.at wrote: > >> One obvious question there is: >> what should the synth do when it reaches the limit? >> There are several things that are possible and afaik implemented in >> synths. It could drop the first note played, or the highest, or ... >> > > Well, that's called voice stealing. Most synths do it, if they don't > have dynamic voice allocation. Usually, you assign voices in a round > robin manner, and the oldest note has to go when you're running out of > voices. > > Ideally the synth should use some kind of priority mechanism when stealing voices. Killing the oldest one is not the best way. For example some kind of psychoacoustic algorithm could be used to find voices that are masked out by the other voices playing at louder levels. Some voices may have decayed to inaudible levels or their pitch may be close enough to the new note to be played. Best regards, Hannu ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
hollun...@gmx.at wrote: > One obvious question there is: > what should the synth do when it reaches the limit? > There are several things that are possible and afaik implemented in > synths. It could drop the first note played, or the highest, or ... > > Just as a hint. I clear the stage for the synth freaks now. > Philipp That's right Philip :) I funk to answer it ;), because my Oberheim has got 6 voices and I played 5 voices all the time, I couldn't discern it, but I guess for this case it doesn't make a big difference, release only is for 'all at last played notes', it can be called 'monophonic polyphony' ;). What you are talking about seems to be interesting for synth that needs to manage e.g. a limit of 64 voices, but for 16 sounds played by 16 channels. I do have such synth, they don't sound like my Oberheim. The selection needs to be able to enable what I call 'monophonic polyphony'. Ralf ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
hollun...@gmx.at wrote: > One obvious question there is: > what should the synth do when it reaches the limit? > There are several things that are possible and afaik implemented in > synths. It could drop the first note played, or the highest, or ... Well, that's called voice stealing. Most synths do it, if they don't have dynamic voice allocation. Usually, you assign voices in a round robin manner, and the oldest note has to go when you're running out of voices. -- Dr. Albert Gr"af Dept. of Music-Informatics, University of Mainz, Germany Email: dr.gr...@t-online.de, a...@muwiinfa.geschichte.uni-mainz.de WWW:http://www.musikinformatik.uni-mainz.de/ag ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] Selectable limit for polyphony of virtual synth
On Mon, 24 Aug 2009 21:10:55 +0200 Ralf Mardorf wrote: > Hi programmers of virtual synthesizers :) > > more than an hour ago I wiped of the dust of my old Oberheim to make > soundfonts of some basses, but some pad sounds captivated me. The > Oberheim is limited by 6-voice polyphony, resp. this isn't a > limitation for those sounds. Virtual synth often tend to make the mix > muddy, when playing pad sounds, because the polyphony isn't limited, > every released note is able to end the complete release decay. For > the Oberheim some notes are cut, while the wanted notes still can > play the release decay. > > A selectable limit for polyphony might be a feature, that should > become more common again, not only for virtual analog synthesizers. > > I know that e.g. fluidsynth-dssi has a global setting for polyphony, > but the problem with this is, that it has impact to all used > fluidsynth-dssi plugins. > > Cheers, > Ralf One obvious question there is: what should the synth do when it reaches the limit? There are several things that are possible and afaik implemented in synths. It could drop the first note played, or the highest, or ... Just as a hint. I clear the stage for the synth freaks now. Philipp ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
[LAD] Selectable limit for polyphony of virtual synth
Hi programmers of virtual synthesizers :) more than an hour ago I wiped of the dust of my old Oberheim to make soundfonts of some basses, but some pad sounds captivated me. The Oberheim is limited by 6-voice polyphony, resp. this isn't a limitation for those sounds. Virtual synth often tend to make the mix muddy, when playing pad sounds, because the polyphony isn't limited, every released note is able to end the complete release decay. For the Oberheim some notes are cut, while the wanted notes still can play the release decay. A selectable limit for polyphony might be a feature, that should become more common again, not only for virtual analog synthesizers. I know that e.g. fluidsynth-dssi has a global setting for polyphony, but the problem with this is, that it has impact to all used fluidsynth-dssi plugins. Cheers, Ralf ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
Re: [LAD] [ANN] Invada Studio LV2 Plugins 1.2.0
Fons Adriaensen wrote: > On Sun, Aug 23, 2009 at 07:27:18PM +0100, Dan Mills wrote: > > >> Lets say your card is aligned so that 0dbFS = +18dbu (EBU standard), >> then 0Vu = +4dbu = - 14dbFS, so a software VU calibrated for 0Vu = >> -14dbFs should read the same as an external Vu calibrated for +4dbu = >> 0Vu. If it does not then either a calibration setting is off somewhere >> or one of the meters is faulty. >> > > True. > > But even a definition such as 'dB FS' is ambiguous, and > it's easy to make mistakes as a result of that. > > Consider a sine wave that is just below digital clipping. > This would be called '0 dB FS', but the actual RMS level > is 3 db lower. I've seen at least one context where this > same signal would be called -3dB FS. Which somehow makes > sense as well. > > AFAIK the first interpretation is the more common one. > 'Peak level' vs 'intrinsic level': * 'dbFS' refers to 'peak level', 'full-scale square wave' * 'dbFS RMS' refers to 'intrinsic level, 'full-scale sine wave'. This is how is in Germany distinguished and because of the words on English, it might be an international way to distinguish it. >> actually fairly common with professional cards. >> > > And it avoids a lot of problems. Semi-pro cards will not > have the correct levels, but some can be quite consistent > between channels. For example my Terratec EWS88MT has less > than +/- 0.1 dB variation between its 8 channels, both > for input and output. But the actual level is just +3.5dBu > for a FS sine wave. > > The real misery starts when (as reported in a recent post > on LAU or LAD), a user finds four volume controls between > his audio file and the physical output (plus of course > a fifth one on his amplifier), and then of course gets > confused on how to set all of them. I guess this is the leading point. Even professional IO amps at times need calibration. We should imagine a setup like 'sound card --> tube pre-amps --> mastering recorder'. For this scenario a VU meter needs to be an external one. Ralf ___ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo/linux-audio-dev
