Re: [music-dsp] A little frivolous diversion on the effect of using a delay
On 2015-07-20, Tom Duffy wrote: Using separate reverbs on each instrument in a DAW recording gives a richer mix that just a single reverb on the master channel. What it gives you is higher decorrelation across channels. And our ears are used to that, because as soon as you move a sound source even one metre in an enclosed, reverberant space, the precise reverberation pattern changes drastically. We perceptually expect a lot of decorrelation from the decaying part of a reverberant sound...though at the same time less from the early, distinguisable slap echoes. (Or, let's say, we expect a different kind of decorrelation; in the short time frame interaural decorrelation because of delay, and in the longer frame essential whiteness overall.) Tom, look at how DirAC processes its arrivals. Starting from Ville Pulkki's research at then TKK Acoustics Lab, and now continuing at Aalto. It's entirely predicated on this sort of thing in its reverb leg. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 ___ music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
On 7/20/15 7:49 PM, Nigel Redmon wrote: To add to Robert’s comment on discrete-time analog… The only thing special about digital sampling is that it’s stable (those digital numbers can be pretty durable—the analog samples don’t hold up so well) and convenient for computation. But the digital audio samples are just a representation of the analog audio that’s been pulse amplitude modulated. (I never worked with BBDs or CCDs, and suspect there’s some zero-order hold involved in practical implementations, there's gotta be *some* voltage at the output at all times. doubt that it's return to zero, so ZOH makes the most sense. but it doesn’t matter as long as that’s compensated for—digital to analog converters have also dealt with the same sort of issue. Still, the basis is that those samples in the BBD/CCD represent impulses, momentary snapshots.) Just as with the digital versions, in the analog versions you have a lowpass filter to ensure the input spectrum remains below half the sample rate, and on the output you have a filter to get rid of the aliased images, created by the modulation process. In the early days of digital delays, the analog delays had some advantages that are probably not obvious to someone coming from today’s knowledge. For instance, today we’d make a delay with a constant sample rate, and use a software LFO and an interpolated delay line to make a flanger. But back then, computation was difficult and costly, so it was done the same way that the analog delays did it: variable sample rate and vary the clock frequency with a hardware LFO. The advantage of digital was better fidelity, but the analog delays could sweep over a much wider range. Digital memory wasn’t so fast back then, and super-fast A/Ds were huge bucks (I worked for a group in a company in the late ‘70s that made a 40 MHz 8-bit A/D chip that was $800 in low quantities, and they sold ‘em as fast as they could make ‘em). geepers. that's fast. around 1979-80, i did a DSP project with a MC6809 and a 12-bit DAC that i double-up and used with a comparator to be a successive approximation ADC. in those days the DAC was $40 and we didn't wanna spend money getting an ADC. the sampling rate was something like 10 Hz (it was a medical application and the signal was very slow.) -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
On Jul 20, 2015, at 11:12 PM, robert bristow-johnson r...@audioimagination.com wrote: On 7/20/15 7:49 PM, Nigel Redmon wrote: To add to Robert’s comment on discrete-time analog… The only thing special about digital sampling is that it’s stable (those digital numbers can be pretty durable—the analog samples don’t hold up so well) and convenient for computation. But the digital audio samples are just a representation of the analog audio that’s been pulse amplitude modulated. (I never worked with BBDs or CCDs, and suspect there’s some zero-order hold involved in practical implementations, there's gotta be *some* voltage at the output at all times. doubt that it's return to zero, so ZOH makes the most sense. RIght. What I meant to imply is that the (mathematical) ideal is an impulse (return to zero), but for practical reasons it’s basically ZOH and you make adjustments. but it doesn’t matter as long as that’s compensated for—digital to analog converters have also dealt with the same sort of issue. Still, the basis is that those samples in the BBD/CCD represent impulses, momentary snapshots.) Just as with the digital versions, in the analog versions you have a lowpass filter to ensure the input spectrum remains below half the sample rate, and on the output you have a filter to get rid of the aliased images, created by the modulation process. In the early days of digital delays, the analog delays had some advantages that are probably not obvious to someone coming from today’s knowledge. For instance, today we’d make a delay with a constant sample rate, and use a software LFO and an interpolated delay line to make a flanger. But back then, computation was difficult and costly, so it was done the same way that the analog delays did it: variable sample rate and vary the clock frequency with a hardware LFO. The advantage of digital was better fidelity, but the analog delays could sweep over a much wider range. Digital memory wasn’t so fast back then, and super-fast A/Ds were huge bucks (I worked for a group in a company in the late ‘70s that made a 40 MHz 8-bit A/D chip that was $800 in low quantities, and they sold ‘em as fast as they could make ‘em). geepers. that's fast. around 1979-80, i did a DSP project with a MC6809 and a 12-bit DAC that i double-up and used with a comparator to be a successive approximation ADC. in those days the DAC was $40 and we didn't wanna spend money getting an ADC. the sampling rate was something like 10 Hz (it was a medical application and the signal was very slow.) These 8-bit ADCs were “flash converters (a string of resistors with comparators feeding a MUX), usually used in video applications. They dropped to $500 in quantities…or, you could buy ones with a missing code or two cheaper, and correct for it in software, as some people on a budget did. They also made those popular 16x16 multipliers and MACs (MPY-16 and MAC-16) that people would make hardware FFT butterflies with. It runs in my mind that the Bell Labs (Alles) synth used a bunch of the multipliers. Now imagine a board full of these things, dissipating 5W each (the Mac-16s anyway—the MPY-16s were a bit less as I recall)…LOL. One cool thing about the 6809 (because a multiply) was that they did all memory access on a half-cycle, so you could put two of them on the same memory out of phase to do more crunching. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
On 20 July 2015 at 22:24 Nigel Redmon earle...@earlevel.com wrote: Here’s an interesting interview: http://www.studioelectronics.biz/Documents/SSC.DEVICE.pdf http://www.studioelectronics.biz/Documents/SSC.DEVICE.pdf Thanks for sharing that delightfully inspiring read to find in my inbox this morning. So many timeless observations and patterns in the intro. The bit about choosing whether to go to market with half-bakery that would sell heaps but ultimately damage the company reputation (people really cared about long term reputation in those days), or perfect a product is notable. As is the transfer of ideas to a music technology domain from work he was doing in medical electronics. And gricky definitely needs to be an official delay control parameter in honour of St Croix. cheers Andy -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
robert bristow-johnson wrote: ... just for the record, none of them content words were written by me. ... And it's back, in the Prophet-6. I build one of those dual BBD effects with filters and electronics, with interesting options, with a sine wave LFO modulated clock to act as a Leslie effect, which was fun, though without noise suppression very hissy. so it's delay modulation, which is supposed to be the Leslie? you really should have a synchronous, but outa-phase, amplitude modulation along with the delay modulation, to emulate a Leslie. and multiple reflection paths to an auditory scene (a wall or two with reflections) and a stereo output derived from that. I was talking about the new P6 having BBDs (/simulation). Not directly connected with that, I used BBDs in the early 80s for simulating among other things the hard to do phase shifting of a imitated organ signal, with an added compander I designed. Nowhere near the sonic riches of good digital simulations from later time, but it sounded not eeky, or those to me dreadful serene listen to this messing with sampling errors way. I don't know how much error was in the balanced BBD I used, probably there was leaking between parts of charge passing stages, and forms of unspecified filtering. It was fun to just modulate the clock analog, like there were also digital delays in that time that would let you smoothly modulate the sampling clock. Doing the same proper with a digital simulation *including correction for sampling errors* isn't necessarily easy. That sure is better even with certain synthesizers (in this case the Yamaha Motif line) have nice oscillators, but it isn't possible to get certain sampled waves to not overlay more than 1 sample, ... uhm, what do you mean? do you mean that the samples for each voice are being played out at different sample rates ... What I mean is that for sound reasons and possibly for preserving intellectual property reasons (I don't know), the machines in many cases output more than one sample at the same time, even if you take one oscillator and one note is played, it outputs a combined waveform consisting at least (this has been a while since I looked at it) of two time shifted versions of the same sample. So the assignment would be to take a source which outputs a layer of the same sample, possibly (so let's presume) at the same frequency, but the layers shifted in time. SO you put all modulations, envelopes and filters of a Motif synth off, output a string wave form from only one oscillator, and you'd get two waves, in the simplest case I would like to get the sample out of some un-add delay effect which was layered and time shifted at the output of the synthesizer, so that out the delay-remover effect, I'd get the sample used in the synth. So essentially, you'd have to estimate the delay time used, and undo the adding of the delayed signal. Going frequency domain is fine, but some work and might not give sample accurate delay removal! I realize that's a bit tough and might involve inversions with linear algebra and some iterations even, but it's a fun subject. I mean so much going on, but simply extracting a signal in the presence of a delayed added version of the same signal isn't generally available! you mean inverting the filter: H(s) = 1 + alpha*e^(-s*T) where T is the delay and alpha is the relative gain of the delayed added version? that can be generally done if you know T and alpha. T. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
Er, minor correction, the effect I was talking about on the tune (where the echo is more part of the sound than perceived as a repeat) is the bass and the textural chordal change thing most easily heard in the sparse section starting at 1:37; my buddy added all the mallet things with echo (still cool, just differentiating what in my mind are two completely different uses of echo). On Jul 20, 2015, at 11:29 AM, Nigel Redmon earle...@earlevel.com wrote: Being a long-time fan of delays (and author of Echo farm PT plug-in and DL4 delay modeler stompbox), starting with tape delay (first a Univox tape delay, graduated to Roland Space Echo (the space echo emulation in Echo Farm is based on my aged RE-101)…when the digital first came in, it was neat at first, but exact (near exact) delay is so boring after a bit, and your realize that the rapid drop-off of frequencies in analog delays is a feature, not a fault, and certainly the pitch consistency of tape echoes. My old prog band recorded an album in 1979, and the engineer/producer wanted to use his shiny new MXR rack delay. I completely regret not demand that we use the space echo—my modular synth sounded so tiny. Anyway, I was having a conversation with my old bandmate some time back, over the phone; he’s a recording engineer producer theses days, and he mentioned something about delays, saying that he never quite latched onto their use (the way I had). I mentioned a fun way to use them that I had always liked (I guess similar to the Alan Parson’s I Robot), then after getting off the call whipped up some simple changes to show him what I meant. Being the guy he is, he couldn’t help but add drums and finish it out. I made a little video for it (he added the echoey sparse vibraphone/marimba melodic part, not really what I’m talking about; I’m referring to the baseline and the textural chordal change parts, also a mallet-ish sound by constant, where the echo is integral to the sound): https://youtu.be/BsNchxCglVk On Jul 20, 2015, at 9:43 AM, Theo Verelst theo...@theover.org wrote: Hi all, No theoretical dumbfounding or deep searching incantations from me this Monday, but just something I've through about and that somehow has since long been a part of music and analog and digital productions. I recall when I was doing some computer audio experiments say in the early 80s that there was this tantalizing effect that outside of special tape based machines hadn't really existed as an effect for using with random audio sources: the digital delay. I recall I was happy when I'd used (low fidelity) AD and DA converters and a early home computer with 64 kilobytes of memory to achieve an echo effect. It was fun. For musical purposes, a bit later I used various digital effect units that optionally could act as a delay line, and with a feedback control, as an echo unit. It seems however that with time, the charm of the effect wore off. Just like nowadays some people occupy themselves with (arguably desirable) reverb reduction, it seems that using a delay isn't very cool anymore, doesn't necessarily make your audio workstation output prettier waves when playing a nice solo, and even it makes samples sound uglier when a digital delay effect is used on them, now that everybody with a computer and a sound card can do some audio processing, in a way that's a shame. Some of the early charm must have been that the effect was featured in popular music, and wasn't easy enough to get for a hobbyist in the 70s, and possibly that the grungy and loose feel of the low bit depth and the jittery or modulated AD/DA converter clock signals was only fun while it lasted. Maybe instruments aren't designed to sound good with a delay effect either, or there's a conflict with audio system's internal processing, and as last suggestion, the studio delay effect does a little bit more than just delaying that makes it so addictive... T. — -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
Being a long-time fan of delays (and author of Echo farm PT plug-in and DL4 delay modeler stompbox), starting with tape delay (first a Univox tape delay, graduated to Roland Space Echo (the space echo emulation in Echo Farm is based on my aged RE-101)…when the digital first came in, it was neat at first, but exact (near exact) delay is so boring after a bit, and your realize that the rapid drop-off of frequencies in analog delays is a feature, not a fault, and certainly the pitch consistency of tape echoes. My old prog band recorded an album in 1979, and the engineer/producer wanted to use his shiny new MXR rack delay. I completely regret not demand that we use the space echo—my modular synth sounded so tiny. Anyway, I was having a conversation with my old bandmate some time back, over the phone; he’s a recording engineer producer theses days, and he mentioned something about delays, saying that he never quite latched onto their use (the way I had). I mentioned a fun way to use them that I had always liked (I guess similar to the Alan Parson’s I Robot), then after getting off the call whipped up some simple changes to show him what I meant. Being the guy he is, he couldn’t help but add drums and finish it out. I made a little video for it (he added the echoey sparse vibraphone/marimba melodic part, not really what I’m talking about; I’m referring to the baseline and the textural chordal change parts, also a mallet-ish sound by constant, where the echo is integral to the sound): https://youtu.be/BsNchxCglVk On Jul 20, 2015, at 9:43 AM, Theo Verelst theo...@theover.org wrote: Hi all, No theoretical dumbfounding or deep searching incantations from me this Monday, but just something I've through about and that somehow has since long been a part of music and analog and digital productions. I recall when I was doing some computer audio experiments say in the early 80s that there was this tantalizing effect that outside of special tape based machines hadn't really existed as an effect for using with random audio sources: the digital delay. I recall I was happy when I'd used (low fidelity) AD and DA converters and a early home computer with 64 kilobytes of memory to achieve an echo effect. It was fun. For musical purposes, a bit later I used various digital effect units that optionally could act as a delay line, and with a feedback control, as an echo unit. It seems however that with time, the charm of the effect wore off. Just like nowadays some people occupy themselves with (arguably desirable) reverb reduction, it seems that using a delay isn't very cool anymore, doesn't necessarily make your audio workstation output prettier waves when playing a nice solo, and even it makes samples sound uglier when a digital delay effect is used on them, now that everybody with a computer and a sound card can do some audio processing, in a way that's a shame. Some of the early charm must have been that the effect was featured in popular music, and wasn't easy enough to get for a hobbyist in the 70s, and possibly that the grungy and loose feel of the low bit depth and the jittery or modulated AD/DA converter clock signals was only fun while it lasted. Maybe instruments aren't designed to sound good with a delay effect either, or there's a conflict with audio system's internal processing, and as last suggestion, the studio delay effect does a little bit more than just delaying that makes it so addictive... T. — -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
Whenever vintage delays come to my mind, I hear the sound of the bucket brigade delay lines that were made from a chain of capacitors and switches. In the early 80s there were many electronic magazine articles and kits to build them. The SAD chips had a maximum delay time of about 200ms. Were they digital? Kind of. Were they analogue? Kind of too. A lost technology from gap between analogue and digital, you can hear them on a surprising number of records, especially early electronic. That odd dub effect where a sound converges on a single low frequency is often BBD set to maximum feedback I think, but is sometimes mistaken for tape echo or early DDL. best to all Andy Farnell -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
The first delay of which I was aware was in the piece Echo III played on the viola by Tim Souster in Cambridge in the early 1970s. Not an echo or reverb but a cannon. Delay was via two reel-to-reel tape machines, with a carefully measured distance between them. I cannot remember if it was the band Intermodulation or 0db, but I loved the piece. Not heard it for decades ==John ff -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
Related: Using separate reverbs on each instrument in a DAW recording gives a richer mix that just a single reverb on the master channel. Back in the analog days, you'd use the multitrack tape and mixer to do multiple passes through the best reverb in the studio. In the early DAW days, you'd have to do the same (because of limited CPU power and the overhead of a good reverb plug-in). Replacing some of the reverbs with delays gave the same result, adding a little bit of space around each instrument that didn't build up into a mess. A well programmed delay would be 2nd on my list of desert island plug-ins after a good reverb. I think the delays are still used on music you hear on the radio, but it's dialed back in subtlety. --- Tom. On 7/20/2015 9:43 AM, Theo Verelst wrote: Hi all, No theoretical dumbfounding or deep searching incantations from me this Monday, but just something I've through about and that somehow has since long been a part of music and analog and digital productions. I recall when I was doing some computer audio experiments say in the early 80s that there was this tantalizing effect that outside of special tape based machines hadn't really existed as an effect for using with random audio sources: the digital delay. I recall I was happy when I'd used (low fidelity) AD and DA converters and a early home computer with 64 kilobytes of memory to achieve an echo effect. It was fun. For musical purposes, a bit later I used various digital effect units that optionally could act as a delay line, and with a feedback control, as an echo unit. It seems however that with time, the charm of the effect wore off. Just like nowadays some people occupy themselves with (arguably desirable) reverb reduction, it seems that using a delay isn't very cool anymore, doesn't necessarily make your audio workstation output prettier waves when playing a nice solo, and even it makes samples sound uglier when a digital delay effect is used on them, now that everybody with a computer and a sound card can do some audio processing, in a way that's a shame. Some of the early charm must have been that the effect was featured in popular music, and wasn't easy enough to get for a hobbyist in the 70s, and possibly that the grungy and loose feel of the low bit depth and the jittery or modulated AD/DA converter clock signals was only fun while it lasted. Maybe instruments aren't designed to sound good with a delay effect either, or there's a conflict with audio system's internal processing, and as last suggestion, the studio delay effect does a little bit more than just delaying that makes it so addictive... T. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp NOTICE: This electronic mail message and its contents, including any attachments hereto (collectively, this e-mail), is hereby designated as confidential and proprietary. This e-mail may be viewed and used only by the person to whom it has been sent and his/her employer solely for the express purpose for which it has been disclosed and only in accordance with any confidentiality or non-disclosure (or similar) agreement between TEAC Corporation or its affiliates and said employer, and may not be disclosed to any other person or entity. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
robert bristow-johnson wrote: On 7/20/15 2:44 PM, padawa...@obiwannabe.co.uk wrote: Whenever vintage delays come to my mind, I hear the sound of the bucket brigade delay lines And it's back, in the Prophet-6. I build one of those dual BBD effects with filters and electronics, with interesting options, with a sine wave LFO modulated clock to act as a Leslie effect, which was fun, though without noise suppression very hissy. That sure is better even with a simple software delay and cheap built-in sound card now, even at 16 bit, a delay can work fine at CD quality. My interest at some point, which got me thinking, is that certain synthesizers (in this case the Yamaha Motif line) have nice oscillators, but it isn't possible to get certain sampled waves to not overlay more than 1 sample, in certain cases probably the same waveform playing over two sample replay partial engines, with a delay in between. So it would be a nice idea to be able to record the signal of a single note, and somehow extract the one sample from the two or three that play at the same time, presuming they're just time shifted. I realize that's a bit tough and might involve inversions with linear algebra and some iterations even, but it's a fun subject. I mean so much going on, but simply extracting a signal in the presence of a delayed added version of the same signal isn't generally available! T. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
Here’s an interesting interview: http://www.studioelectronics.biz/Documents/SSC.DEVICE.pdf http://www.studioelectronics.biz/Documents/SSC.DEVICE.pdf I first heard about it at AES (’75 in LA), from Stephen St. Croix himself. It was a brand new product, and Steve was trying to convince anyone who would listen. He was giving away cool t-shirts too, and my buddy and I wanted one. He was a little ticked, I think, because he could tell we were more interested in the t-shirts and were just waiting for him to finish and get the shirts, but he gave his passionate speech and I was listening more than he probably thought. He was basically selling against the new delta-encoded digital competition, telling us why it sucked, and why the wimpy clocking range (compared to his analog device) meant their flanging sucked, etc. He handed us our shirts and we were gone to see what other cool stuff was at the show. But not too long after, the electronic music lab at USC got one, and I made good use of it. At the end of summer, it was stolen. I was a lab rat and was the last booking before then shut down for a couple of weeks ahead of the fall semester—and when they opened the lab next, it was gone. They got a new one, and identical circumstances—again, I was the last guy to book the lab int he summer session, and when they re-opened the new one was gone as well. It’s not like they cleaned out the lab—someone really like those Marshall Time Modulators. So, interesting history with them. St. Croix was plagued by problems obtaining parts (the dvx modules, the CCDs), so I don’t think a large number were built, and they cost too much for me at the time. I sure loved the sound, though. On Jul 20, 2015, at 1:45 PM, pdowling hello.pdowl...@gmail.com wrote: Marshall Time Modulator got some good links? On 20 Jul 2015, at 21:40, Nigel Redmon earle...@earlevel.com wrote: Most of the old delays were BBD, but the king of discrete-time analog was the Marshall Time Modulator, which used CCDs. Between the dbx companding for increased s/n and the wide clock-sweeping range, it had awesome flanging (-80dB notches claimed)—great double/triple tracking too. On Jul 20, 2015, at 12:16 PM, robert bristow-johnson r...@audioimagination.com wrote: On 7/20/15 2:44 PM, padawa...@obiwannabe.co.uk wrote: Whenever vintage delays come to my mind, I hear the sound of the bucket brigade delay lines that were made from a chain of capacitors and switches. In the early 80s there were many electronic magazine articles and kits to build them. The SAD chips had a maximum delay time of about 200ms. Were they digital? Kind of. no, they weren't. not really. discrete-time is not the same as digital. Were they analogue? Kind of too. they were fully analog[ue]. A lost technology from gap between analogue and digital, you can hear them on a surprising number of records, especially early electronic. That odd dub effect where a sound converges on a single low frequency is often BBD set to maximum feedback I think, but is sometimes mistaken for tape echo or early DDL. to the precision of the A/D and D/A converters (which is considerable), there is no reason that a modern digital delay ling can't be made to sound like the old CCD (or BBD or whatever you wanna call it) delay products. like an analog[ue] amplifier, you might have to model in analog non-linearities, noise, buzz, hum, and interference to make it sound the same. with the exception of the non-linearities, i normally think that modeling the noise and buzz leaking through is not desirable. who knows? one thing i think might be cool is to use different delay/echo effects on each string of a hex-pickup gitfiddle. just like you might have different pitch shifting done on each string. -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
On 7/20/15 3:00 PM, jpff wrote: The first delay of which I was aware was in the piece Echo III played on the viola by Tim Souster in Cambridge in the early 1970s. Not an echo or reverb but a cannon. Delay was via two reel-to-reel tape machines, with a carefully measured distance between them. I cannot remember if it was the band Intermodulation or 0db, but I loved the piece. Not heard it for decades the first i remember was the Echoplex. single tape loop, but one of the heads (i think the playback head) was on a mechanical slider. i think there was a feedback gain knob. i dunno what may have preceded that. did Echo III precede the Echoplex? -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
Most of the old delays were BBD, but the king of discrete-time analog was the Marshall Time Modulator, which used CCDs. Between the dbx companding for increased s/n and the wide clock-sweeping range, it had awesome flanging (-80dB notches claimed)—great double/triple tracking too. On Jul 20, 2015, at 12:16 PM, robert bristow-johnson r...@audioimagination.com wrote: On 7/20/15 2:44 PM, padawa...@obiwannabe.co.uk wrote: Whenever vintage delays come to my mind, I hear the sound of the bucket brigade delay lines that were made from a chain of capacitors and switches. In the early 80s there were many electronic magazine articles and kits to build them. The SAD chips had a maximum delay time of about 200ms. Were they digital? Kind of. no, they weren't. not really. discrete-time is not the same as digital. Were they analogue? Kind of too. they were fully analog[ue]. A lost technology from gap between analogue and digital, you can hear them on a surprising number of records, especially early electronic. That odd dub effect where a sound converges on a single low frequency is often BBD set to maximum feedback I think, but is sometimes mistaken for tape echo or early DDL. to the precision of the A/D and D/A converters (which is considerable), there is no reason that a modern digital delay ling can't be made to sound like the old CCD (or BBD or whatever you wanna call it) delay products. like an analog[ue] amplifier, you might have to model in analog non-linearities, noise, buzz, hum, and interference to make it sound the same. with the exception of the non-linearities, i normally think that modeling the noise and buzz leaking through is not desirable. who knows? one thing i think might be cool is to use different delay/echo effects on each string of a hex-pickup gitfiddle. just like you might have different pitch shifting done on each string. -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
On 7/20/15 2:44 PM, padawa...@obiwannabe.co.uk wrote: Whenever vintage delays come to my mind, I hear the sound of the bucket brigade delay lines that were made from a chain of capacitors and switches. In the early 80s there were many electronic magazine articles and kits to build them. The SAD chips had a maximum delay time of about 200ms. Were they digital? Kind of. no, they weren't. not really. discrete-time is not the same as digital. Were they analogue? Kind of too. they were fully analog[ue]. A lost technology from gap between analogue and digital, you can hear them on a surprising number of records, especially early electronic. That odd dub effect where a sound converges on a single low frequency is often BBD set to maximum feedback I think, but is sometimes mistaken for tape echo or early DDL. to the precision of the A/D and D/A converters (which is considerable), there is no reason that a modern digital delay ling can't be made to sound like the old CCD (or BBD or whatever you wanna call it) delay products. like an analog[ue] amplifier, you might have to model in analog non-linearities, noise, buzz, hum, and interference to make it sound the same. with the exception of the non-linearities, i normally think that modeling the noise and buzz leaking through is not desirable. who knows? one thing i think might be cool is to use different delay/echo effects on each string of a hex-pickup gitfiddle. just like you might have different pitch shifting done on each string. -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
To add to Robert’s comment on discrete-time analog… The only thing special about digital sampling is that it’s stable (those digital numbers can be pretty durable—the analog samples don’t hold up so well) and convenient for computation. But the digital audio samples are just a representation of the analog audio that’s been pulse amplitude modulated. (I never worked with BBDs or CCDs, and suspect there’s some zero-order hold involved in practical implementations, but it doesn’t matter as long as that’s compensated for—digital to analog converters have also dealt with the same sort of issue. Still, the basis is that those samples in the BBD/CCD represent impulses, momentary snapshots.) Just as with the digital versions, in the analog versions you have a lowpass filter to ensure the input spectrum remains below half the sample rate, and on the output you have a filter to get rid of the aliased images, created by the modulation process. In the early days of digital delays, the analog delays had some advantages that are probably not obvious to someone coming from today’s knowledge. For instance, today we’d make a delay with a constant sample rate, and use a software LFO and an interpolated delay line to make a flanger. But back then, computation was difficult and costly, so it was done the same way that the analog delays did it: variable sample rate and vary the clock frequency with a hardware LFO. The advantage of digital was better fidelity, but the analog delays could sweep over a much wider range. Digital memory wasn’t so fast back then, and super-fast A/Ds were huge bucks (I worked for a group in a company in the late ‘70s that made a 40 MHz 8-bit A/D chip that was $800 in low quantities, and they sold ‘em as fast as they could make ‘em). But you probably sweep those CCD clocks from something like 20 Khz to over 1 MHz (kinda of guessing here, but the point is that you could get nowhere remotely close to that with a DDL). On Jul 20, 2015, at 12:16 PM, robert bristow-johnson r...@audioimagination.com wrote: On 7/20/15 2:44 PM, padawa...@obiwannabe.co.uk wrote: Whenever vintage delays come to my mind, I hear the sound of the bucket brigade delay lines that were made from a chain of capacitors and switches. In the early 80s there were many electronic magazine articles and kits to build them. The SAD chips had a maximum delay time of about 200ms. Were they digital? Kind of. no, they weren't. not really. discrete-time is not the same as digital. Were they analogue? Kind of too. they were fully analog[ue]. A lost technology from gap between analogue and digital, you can hear them on a surprising number of records, especially early electronic. That odd dub effect where a sound converges on a single low frequency is often BBD set to maximum feedback I think, but is sometimes mistaken for tape echo or early DDL. to the precision of the A/D and D/A converters (which is considerable), there is no reason that a modern digital delay ling can't be made to sound like the old CCD (or BBD or whatever you wanna call it) delay products. like an analog[ue] amplifier, you might have to model in analog non-linearities, noise, buzz, hum, and interference to make it sound the same. with the exception of the non-linearities, i normally think that modeling the noise and buzz leaking through is not desirable. who knows? one thing i think might be cool is to use different delay/echo effects on each string of a hex-pickup gitfiddle. just like you might have different pitch shifting done on each string. -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] A little frivolous diversion on the effect of using a delay
On 7/20/15 4:52 PM, Theo Verelst wrote: robert bristow-johnson wrote: On 7/20/15 2:44 PM, padawa...@obiwannabe.co.uk wrote: Whenever vintage delays come to my mind, I hear the sound of the bucket brigade delay lines just for the record, none of them content words were written by me. And it's back, in the Prophet-6. I build one of those dual BBD effects with filters and electronics, with interesting options, with a sine wave LFO modulated clock to act as a Leslie effect, which was fun, though without noise suppression very hissy. so it's delay modulation, which is supposed to be the Leslie? you really should have a synchronous, but outa-phase, amplitude modulation along with the delay modulation, to emulate a Leslie. and multiple reflection paths to an auditory scene (a wall or two with reflections) and a stereo output derived from that. That sure is better even with a simple software delay and cheap built-in sound card now, even at 16 bit, a delay can work fine at CD quality. My interest at some point, which got me thinking, is that certain synthesizers (in this case the Yamaha Motif line) have nice oscillators, but it isn't possible to get certain sampled waves to not overlay more than 1 sample, in certain cases probably the same waveform playing over two sample replay partial engines, with a delay in between. So it would be a nice idea to be able to record the signal of a single note, and somehow extract the one sample from the two or three that play at the same time, presuming they're just time shifted. uhm, what do you mean? do you mean that the samples for each voice are being played out at different sample rates and zero-order held and then the different voices overlay their samples coming out at different rates? i might think that if you analog LPF each voice separately before adding them, the overlay more than 1 sample wouldn't be an issue. I realize that's a bit tough and might involve inversions with linear algebra and some iterations even, but it's a fun subject. I mean so much going on, but simply extracting a signal in the presence of a delayed added version of the same signal isn't generally available! you mean inverting the filter: H(s) = 1 + alpha*e^(-s*T) where T is the delay and alpha is the relative gain of the delayed added version? that can be generally done if you know T and alpha. -- r b-j r...@audioimagination.com Imagination is more important than knowledge. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
