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 Forum index » How-tos » Production - engineering/mixing
the Digital/Analogue time bog
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Digital and Analogue together?
Totally
66%
 66%  [ 10 ]
Mostly
6%
 6%  [ 1 ]
A little bit
26%
 26%  [ 4 ]
No way
0%
 0%  [ 0 ]
Total Votes : 15

Author Message
Johan Zwart



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PostPosted: Sat May 27, 2006 1:27 am    Post subject: Re: POLL : Digital and Analogue together?
Subject description: Q: do you think they mix good?
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teradacto wrote:
this is the poll. let me know, your side of this epic battle.


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PostPosted: Sat May 27, 2006 8:23 am    Post subject: Reply with quote  Mark this post and the followings unread

Kassen wrote:
Right. Well, I don't know anybody who can make a digital system sound like "analogue saturation". Rob comes close but it's not quite the same and I never heard anything that comes closer. I don't see any real point to it either; to me digital imitations of analogue systems are like physical models of existing instruments. It's interesting as a purely accedemic persuit in order to facilitate a deeper understanding of what's going on but as a instrument it makes no sense to me.


My point is the digital and analog are just circuit techniques - no not mutually exclusive. Virtually every modern piece of equipment uses both techniques these days, even the analog gear.

When an engineer designs something, they have to compromise by picking components and techniques available at the time the design is made. Rob used a synthesizer to model analogue saturation. I don't think he'd argue that the G2 patch this was a quick and dirty digital breadboard to demonstrate some principles while providing a useful patch. If he was motivated to come up with a better solution, he could have either using more G2 resources, or perhaps using another platform. What would be better - more wow and flutter, 50/60 Hz hum, hiss, analog pops?

BTW, it is not "analog saturation" (you know that because you put it in quotes but I'm making a point). Many of the finest analog circuits will clip when over driven. Rob didn't call it analog saturation but rather an emulation of the Revox A77. Tape echo on a Teac 3300 would be a bit different, I guess.

Academic persuits are sometimes futile and meaningless, but in this case they are very valuable. The Nyquist therory establishes that you can represent any band limited signal if you sample it at at least twice the highest frequency of interest. So this theory establishes the basis that time quantized systems are theoretically equivalent continuous sytems - within the band limited constraints.

As you pointed out, digital systems are also level quantized. Another academic foundation is psychoacoustic research which can verify that humans can not perceive small differences.

Many of the analog/digital discussions mirror the tube/transitor discussions of the 50s and 60s.

OPG's comment about the issues recording an analog synth with a digital recorder. Assume you record it well with the levels set correctly, the most undesirable artifacts will result from the analog componets in the signal chain, including the D/A and A/D converters. In my experience, a decent Firewire audio interface of today is light years better than the finest tape recorders. At best, those beasts had 55 dB S/N ratios (who could have afforded one that good). That's why we used Dolby noise reduction units (talk about horrible mutilation of the sound).

If you want to record an electronic signal transparently and with fidelity using today's technology, I would suggest using digital. If you want some sort of special effect - then using tape is cool.

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PostPosted: Sat May 27, 2006 9:31 am    Post subject: Reply with quote  Mark this post and the followings unread

These are all great posts. And, if I haven't stated it enough in other discussions, I absolutely OBSESS over those faults in analog systems (hiss, flutter, clicks and pops, etc). So far, it's been hard to replicate these faults in the digital world without losing other desired qualities, but I'll experiment and learn.

Now if you'll excuse me, I'm going to boot up my Apple IIe, start running a bunch of POKE commands for the speaker output, and record them onto the computer to make a soundfont. Cool
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PostPosted: Sat May 27, 2006 12:17 pm    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:

My point is the digital and analog are just circuit techniques - no not mutually exclusive. Virtually every modern piece of equipment uses both techniques these days, even the analog gear.


No, of cource not, but nobody claimed they were mutually exclusive; I just said it matters which one you pick where.

Before this gets too confusing; You cliamed that analog and digital were the same at the core level (this is true) and that this means it doesn't matter which one you use (this isthe bit I have a problem with).

Quote:
If he was motivated to come up with a better solution, he could have either using more G2 resources, or perhaps using another platform.


True. Instead Rob switched to developing analogue boards. In nearly all cases implementing a technique prototyped in the G2 in analogue cerquitry led to a marked inrease in sound quality. for these speciffic techniques this can be traced back directly to the quantisation in the time dimention.

I know because I co-designed some of the things he's working on and I can point out exactly where and how it goes wrong.

Quote:
The Nyquist therory establishes that you can represent any band limited signal if you sample it at at least twice the highest frequency of interest. So this theory establishes the basis that time quantized systems are theoretically equivalent continuous sytems - within the band limited constraints.


Ok, yes, this is true, but this theorem talks exclusively about recording, storage and playback. It says nothing about the situation where something gets recorded, stored, treated, then played back. Some sample rate may be sufficient to record a speciffic signal but that doesn't mean it's also a good idea to *treat* the signal at the same rate. That's a entirely different matter. Some treatments will demand a sample rate that's well above what current digital systems can provide in real time. In those cases you can compromise, you can glitch or you can switch to a analogue system which will indeed lead to issues by itself. In the case of those designs I mentioned above these quantistion issues generate far, far more noise then the natural hiss of electrical components, this holds true even on the G2 with it's high sample rate and even when working with bass frequencies. The resolution is simply too coarce and everything will go upleasantly out of tune.

Quote:

As you pointed out, digital systems are also level quantized. Another academic foundation is psychoacoustic research which can verify that humans can not perceive small differences.


Very true, but do keep in mind that what we can and can't hear is highly dependant on the context. For example; most people can't hear a 24KHz sine but we can tell the difference between a 12KHz sine and a 12KHz saw.

Also; many techniques in electronic music involve some sort of feedback or recursion. Just because it's provable that we can't hear the effect of one pass does not mean we won't be able to preceive the effect of several thousand over the cource of a note. This too affects analogue systems as well as digital ones and it's effect is especially pronounced in hybrid systems but for some reason it's not taken into account when talking about digital systems to the same degree that it is in analogue ones. I think that leads to a scewed perception of the qualities of digital systems.

Quote:

Many of the analog/digital discussions mirror the tube/transitor discussions of the 50s and 60s.


True and I'm happy you bring that up. Early transistors weren't all that good at all. In the same way early digital synths didn't perform as they were advertised to. I have the manual of my s612 here and that manual cheerfully claims the output of the s612 is completely identical it the input because it's such a high quality *digital* device. Well, I'm terribly sorry but that is a flat out lie.

It's also a flat out lie to pretend that they have progressed to the degree that they are now perfect; they aren't and they won't be untill we have processors that facilitate a sample rate that's several orders of magnitude higher then the current ones.

Currently transistors are preferable to tubes in all but the most exotic cases; I would say that in the 50's and 60' the situation was rather different and you'd better pick the right one for your speciffic aplication. I'm sure there was a lot of debate and it was debate that would be silly to maintain now but there was actual cause for debate back then.

The same holds true for curent technologies where signal treatment is involved.

Quote:

OPG's comment about the issues recording an analog synth with a digital recorder. Assume you record it well with the levels set correctly, the most undesirable artifacts will result from the analog componets in the signal chain, including the D/A and A/D converters.


Absolutely.
Asuming a quality recorder the sound should come out again as it went in. This would be the right choice if you want to register your sound sources exactly as they are and no further treatment will be needed.

Quote:

In my experience, a decent Firewire audio interface of today is light years better than the finest tape recorders. At best, those beasts had 55 dB S/N ratios (who could have afforded one that good). That's why we used Dolby noise reduction units (talk about horrible mutilation of the sound).


Well, of cource! Decades of development paid off. Modern soundcards are very, very good for many things. I myself invested 500 euro in getting a very good soundcard, I'm a big fan of this development.

However, it needs to be seen in perspective. Perfect recording and perfect plaback aren't all there is to digital systems. They are relatively straightforward stages in the process and so are ahead in terms of development. If we are talking about using the whole system and particularly if we are planing to use the whole of the system in processing sound in order to make the sound pleasant then the situation becomes rather more complicated.

For example; I use a Philips tape recorder. Philips, unlike some other brands had their own compander chips instead of the Sony Dolby designs. These lead to very speciffic effects that I personally experience as rather pleasant in some cases. Also; for single takes I'm not losing any sleep over a StoN ratio of 55db because that's not that much over the 60 db ratio of records; I don't care that much.

Quote:

If you want to record an electronic signal transparently and with fidelity using today's technology, I would suggest using digital. If you want some sort of special effect - then using tape is cool.


Sure, that's why I have nice versions of both and make a educated choice based on the demands of the situation.

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PostPosted: Sat May 27, 2006 3:38 pm    Post subject: Reply with quote  Mark this post and the followings unread

Kassen wrote:
...I use a Philips tape recorder. Philips, unlike some other brands had their own compander chips instead of the Sony Dolby designs. These lead to very speciffic effects that I personally experience as rather pleasant in some cases.


Too bad, I'm not familiar with it's characteristic sound. Phillips tape decks are a rare breed in the US.

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PostPosted: Sat May 27, 2006 4:20 pm    Post subject: Reply with quote  Mark this post and the followings unread

Yeah, I figured that. Those chips themselves are also found elsewhere though. For example Boss used them in the feedback path of their analogue delays to supress noise and help in mixing new material with repeats of old stuff.

Needless to say there will be trouble if you set those loose on everything in sight but with carefull aplication they can be very nice.

I trust we are now in agreement on most of these points?

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PostPosted: Sat May 27, 2006 4:47 pm    Post subject: Reply with quote  Mark this post and the followings unread

Aha.. the socalled DNR? Very Happy
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PostPosted: Sat May 27, 2006 6:18 pm    Post subject: Reply with quote  Mark this post and the followings unread

Philips did use DNR (Dynamic Noise Reduction), it was not a compressor / expander system though but rather high off filtering depending on signal levels (of a filtered version of the signal IIRC). A playback only system, and pretty simple, just a few transistors per channel were used.

But I vaguely seem to remember they used some two way system as well, which was not Dolby and not Dbx.

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PostPosted: Sat May 27, 2006 6:30 pm    Post subject: Reply with quote  Mark this post and the followings unread

Yup, that sounds like their DNR. I think there are some schematics online somewhere.
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PostPosted: Sat May 27, 2006 7:15 pm    Post subject: Reply with quote  Mark this post and the followings unread

Schematics .. I once built one myself, so it should be somewhere in the cubic meters ...

Which reminds me that I'll have to move all off that stuff before next wednesday, they're coming to mount new windows (not for the pc, plasticize the woodwork rather & double glazing. This is progress, and it can't be stopped unfortunately).

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PostPosted: Sun May 28, 2006 8:24 am    Post subject: Reply with quote  Mark this post and the followings unread

Kassen wrote:

True. Instead Rob switched to developing analogue boards. In nearly all cases implementing a technique prototyped in the G2 in analogue cerquitry led to a marked inrease in sound quality. for these speciffic techniques this can be traced back directly to the quantisation in the time dimention.

One can find many examples where one version of a design is better than another version, at least from one persons perspective. Argument by analogy is one thing, but using a specific example to support a general statement doesn't convince me of anything. For example, I'm sure a Kyma implementation would sound different from the G2 - better or worse, who's to say. You personally don't like "the sound" of the G2 so you won't care for anything you hear from it.

Quote:
Some sample rate may be sufficient to record a speciffic signal but that doesn't mean it's also a good idea to *treat* the signal at the same rate. That's a entirely different matter.

Specifically, non-linear (recursion or feedback is generally linear, BTW) transformations can produce aliases, which is why these are treated by increasing the internal sampling rates. . That's why some people use 192 KHz on their DAWs.

Quote:
Some treatments will demand a sample rate that's well above what current digital systems can provide in real time.

I think current technology can provide a lot of processing power, you just have to pay for it. A Kyma with 27 DSPs can really cook. Still, "some treatments" is sufficiently vague that arguement is impossible. Similarly, one could say that some very complex treatments that can be accomplished digitally can not be implemented with current analog technology.

Quote:
Early transistors weren't all that good at all. In the same way early digital synths didn't perform as they were advertised to.

That goes back to the point that it is unproductive to generalize from specific examples. Someone might say, for example, "I have never heard a transistor amp that sounds as good as a tube amp, so transistors are not as good, or they at least sound different".

I think that early transistors were fine, but many of the early amps had problems when they overloaded. The most damning thing the audiophiles didn't like about solid-state was the clean undistorted sound. Robert Carver helped put an end to all that when he released a high-end transistor amp with a tube emulator circuit that won over those people who disliked the transistor. I remember some of the reviews - "Finally, Carver's solid-state amp sounds as good as tubes" - or somthing similar. The emulator added distortion that the tubes had. Some people liked this - and some still hate transistors not to mention digital.

Quote:
I trust we are now in agreement on most of these points?


Maybe so, but we don't need to be. No reason that between the two of us we shouldn't have three opinions.

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PostPosted: Sun May 28, 2006 9:02 am    Post subject: Reply with quote  Mark this post and the followings unread

ok. Please read what I actually wrote above, then we can talk if you still feel you disagree.
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PostPosted: Sun May 28, 2006 5:43 pm    Post subject: Reply with quote  Mark this post and the followings unread

I just finished that Apple IIe soundfont today. Mother of God, it has a big sound! I'm sure there are programs I could have found for it, but I had to create the chromatic notes myself using the POKE command. Anyway, if I had heard the sounds without knowing where they came from, I would have said an analog synth. I don't know if this is a good thing or not. It almost makes me second-guess the WSG. Perhaps I need to re-record the WSG.

On the Apple IIe, there is a tape I/O for storage. One unique thing about it (unlike the Tandy m100 I have), is that the 1/8" cassette output will put out the actual sound that the internal speaker was programmed by the user to do, not the code (the squealing modem-like sound that is normally what is saved onto tape).

By running:

10 POKE -16336,0
20 GOTO 10

you will put out a tone to the speaker. But if you enter the same thing but with "-16337,0" it will be sent out to the cassette. There is in fact a range of numbers that are associated with toggling the speaker. For creating sound to be sent out to tape, the range is -16337 to -16352. There has to be a comma and number after it for correct syntax. This is where the fun begins. Cool

Through simple trial and error, I came up with POKE numbers for about two octaves. I haven't really seen a pattern in the numbers, but what I do know is that the larger the number is after the comma, the lower the note is. And it is rare that you will come across a combination that is perfectly in tune. You can alo enter decimals, and to some extent, letters. For example:

-16337,4
-16337,04
-16337,.4
-16337,0.4
-16337,.04
-16337,004

These all create different tones. In fact, you could just enter a decimal point after the comma and get a new tone.

So what is going on here? How much is digital and how much is analog? How close is this to an analog IC in a synth? Probably not much, since it is an old computer. Then there's the cassette out jack that baffles me. It sends out the same tone to tape as you would hear from the internal speaker. When I tried this on the Tandy m100, only the audio version of the code that had been entered was sent to tape.

Sorry if that was a bit off-topic, but it really got me thinking about Mosc's talk on digital and alaong being fundamentally the same. Now if you'll excuse me, I'm going to return to the Apple and create short programs of different tones while varying the speed to create percussion. I've already created a bass drum! Smile
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PostPosted: Sun May 28, 2006 5:51 pm    Post subject: Reply with quote  Mark this post and the followings unread

Are you going to publish some of these soundfonts? Very Happy
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PostPosted: Sun May 28, 2006 6:16 pm    Post subject: Reply with quote  Mark this post and the followings unread

As for the analog vs digital being the same, this is a nice observation but I am not sure this is relevant when answering posts like this one:

http://electro-music.com/forum/topic-11713.html

We are talking about gear and not really about what is possible on a really sunny day.
It is quite correct that audio gear sound the way they do because of the design and the parts used. Because of this a cheap Yamaha digital mixer won´t be a substitution for an upmarket Allen and Heath mixer. A Behringer digital mixer won´t sound like a Neve board and a Neve board won´t sound like a Midas Venice. The UAD-1 plugins are quite excellent it truth be told in many circumstances it is easier to create some sort of signature sound using that one rather than using the real thing. -But then in some cases the UAD-1 concept won´t work at all because in order to get the needed signature sound you really need a full analog signal chain complete with upmarket ribbon mics and a very nice recording studio.
I think what this is really about is knowing how stuff works and then aquiring gear for the right reasons. I am not into nostalgia and I am definitively not into a lot of the current analog/digital hype. When new members post their "what gear do I need and by the way my friends say I need fat warm analog gear in order to make in the music industry" well.. how do we answer this? 25 years ago it was far easier. Then most of the newbie questions were about the music and how to put together a band. If you needed a certain beat all you needed to do was to learn it or .. you could sack the current rythm section and look for better musicians. Shocked Cool

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PostPosted: Sun May 28, 2006 8:44 pm    Post subject: Re: cool
Subject description: thanks
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teradacto wrote:
the other being when, I used an analogue sequencer and tried to synchronize it with a drum machine tempo, it seemed like it was imposoble.


That's should be easy with a MSY2 from doepfer. It converts MIDI clock to SYNC and clock signals.

http://www.doepfer.de/msy2.htm

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PostPosted: Sun May 28, 2006 9:45 pm    Post subject: Reply with quote  Mark this post and the followings unread

opg wrote:
Sorry if that was a bit off-topic, but it really got me thinking about Mosc's talk on digital and alaong being fundamentally the same.


Glad it got you thinking - that's all I was trying to do.

Peace

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PostPosted: Mon May 29, 2006 6:44 am    Post subject: Reply with quote  Mark this post and the followings unread

elektro80 wrote:
Are you going to publish some of these soundfonts? Very Happy


Sure, I'll put them up for download on my website (whenever I get that finished). It's kind of a mess right now. Embarassed
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PostPosted: Tue May 30, 2006 5:36 pm    Post subject: uh
Subject description: ah
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Embarassed I quit making electronic music. Shocked

i'm going to make Pop music, or Yodeling or something...just so i can find a forum where hopefully everybody isn't so god damn smart. Very Happy

ya frikkin' braniacs
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PostPosted: Tue May 30, 2006 5:47 pm    Post subject: Re: uh
Subject description: ah
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teradacto wrote:
Embarassed I quit making electronic music. Shocked

i'm going to make Pop music, or Yodeling or something...just so i can find a forum where hopefully everybody isn't so god damn smart. Very Happy

ya frikkin' braniacs


I'm just a big phoney. I talk big, but I still haven't learned to tie my shoes. Long live Velcro!

Seriously, sometimes I get so frustrated with musician's block and "sound quality" that I think I should just scale down to almost nothin'- like a Casio SK-1 (not circuit-bent) and a 4-track. Or a Gameboy and a minidisc recorder. Luckily, I have these things in case I have a freak-out. Shocked
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PostPosted: Tue May 30, 2006 6:53 pm    Post subject: Reply with quote  Mark this post and the followings unread

Kassen wrote:
Some treatments will demand a sample rate that's well above what current digital systems can provide in real time. In those cases you can compromise, you can glitch or you can switch to a analogue system which will indeed lead to issues by itself. In the case of those designs I mentioned above these quantistion issues generate far, far more noise then the natural hiss of electrical components, this holds true even on the G2 with it's high sample rate and even when working with bass frequencies. The resolution is simply too coarce and everything will go upleasantly out of tune.

...

Also; many techniques in electronic music involve some sort of feedback or recursion. Just because it's provable that we can't hear the effect of one pass does not mean we won't be able to preceive the effect of several thousand over the cource of a note. This too affects analogue systems as well as digital ones and it's effect is especially pronounced in hybrid systems but for some reason it's not taken into account when talking about digital systems to the same degree that it is in analogue ones. I think that leads to a scewed perception of the qualities of digital systems.


I think this is very interesting and is definitely something alot of electronic musicians aren't aware of... it's certainly not something we covered in digital theory at uni, but it makes sense. How bad is this? What sort of things should people avoid? Is there some kind of test I can do to hear the true results of this recursive quantisation?
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PostPosted: Tue May 30, 2006 6:59 pm    Post subject: Reply with quote  Mark this post and the followings unread

Yeah, that is a good point. Nobody wants to bash the "new technology," but they love pointing out the flaws of older technology, especially when they aren't as skilled at it and claim it is "limited."
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PostPosted: Sat Jun 03, 2006 6:41 pm    Post subject: Reply with quote  Mark this post and the followings unread

opg wrote:
Yeah, that is a good point. Nobody wants to bash the "new technology," but they love pointing out the flaws of older technology, especially when they aren't as skilled at it and claim it is "limited."


Indeed. I'd love to know for sure what kind of stuff accentuates these quantisation issues.... I assume Kassen's talking about delay based effects like chorus, flange etc... Any chance you could elaborate/specify Kassen?
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Kassen
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PostPosted: Thu Jun 08, 2006 10:39 am    Post subject: Reply with quote  Mark this post and the followings unread

Afro88 wrote:

I think this is very interesting and is definitely something alot of electronic musicians aren't aware of... it's certainly not something we covered in digital theory at uni, but it makes sense. How bad is this? What sort of things should people avoid? Is there some kind of test I can do to hear the true results of this recursive quantisation?


Lots of big questions. First of all "How bad is it?" Well, it's disasterous and completely unaceptable -in some rare cases- and it's also perfectly fine and not something to wory about at all -in soe other cases-. The tricky thing is that you need to sort them yourself, probably by ear.

You are not going to get around quantisation in digital systems. Digital systems are quantised in both time and amplitude and there is no way around it. Did I mention there isn't? Well, there isn't.

One very simple and clear cut case is digital IIR filters (that's filters that work with feedback, meaning most of the musical ones). If we have a IIR filter in a 16 bit system then at the end of the equation for one sample value we get a multiplication and a adition. Two 16 bit values will be multiplied, this will give a 32 bit value. However, we can't have 32 bit values bouncing around in 16 bit systems and so it's rounded to 16 again. The difference between the 32 bit value we should have at this point and the 16 bit one we get is basically noise. This noise becomes a part of the signal and is passed again and this adds up. This is a big deal. It's a non-linear part of the system and may actually make a filter unstable. Fortunately this is a well known problem and there are strategies against it and typically in modern instruments it's not something the end user needs to wory about but it's a nice and clear example of where it goes wrong. Even having to define a filter mathematically in a finite number of bits inherently means the practical filter's character that you get is different from the ideal one you wanted.

Don't lose sleep over this; very smart people are working on these problems but it's important to know that while digital systems don't have hiss like tape does they are not immune to noise AT ALL. For any practical system every operation will add some noise due to rounding. Anybody who tells you digital systems don't have noise doesn't understand how they work.

Much like analogue systems the question becomes how much noise is acceptable. For example; if you have a noise floor of 80db you may regret having some noise but in practice nobody may notice it. at 30db it may be very objectionable and may need measures to combat it. This is not news; this stuff is a standard question to ask when building or using some system. chances are the system could be improved but improvements come at costs. You could have a much better filter if you were willing to have your whole pc render at it all night for one phrase of a lead line but that cost is probably impractical to pay.

Let's have a look at the system I was dealing with and meantioned above, then move on to practical matters. What Rob & me had was a system where we needed to detect zero crossings. Since in digital systems with a high resolution it's exceedingly rare for any one sample to be exactly "zero" we can only detect the waveform went through zero in the last sample period AFTER that period has passed. This means that on average we have half a sample period of offset. If you now considder that how this is preceived depends on the ratio between the input wave's frequency and the sample frequency you'll understand that this leads to sweeps being preceived as terribly out of tune. This could be fought by a huge increase in the sample frequency but that would come at very high cpu costs (not to mention that fast ADC's that are also high resolution are expensive). Unlike what Howard sugests that's a fundemantal property of realtime digital systems and has nothing at all to do with the G2, this is simply one example of a technique where it's curently unpractical to choose a digital implementation if tonal character during realtime sweeps are important. There is no nostalgia, "G2 aversion" or whatever there; this is pure cold, hard math.

Ok, let's get back to practical DAW's. Say you are working on some project that is to be on a cd. Say you use only 16bit 44KHz wavefiles. (we have 90db or so of S/N ratio there and 22KHz of frequency resolution) If you set some plugins and filters lose on this these will internally upsample your material, process it, then downsample again and all the problems you would have had will be in the bits that get dropped at the end. The largest amount of noise introduced in this should be in the rounding to 16 bit at the end. Great; you can't hear that. Now another plugin, and another, and another.... And suddenly your material has that vague glass-like character that mid-90's "experimental techno" material had.

What's going on here? The errors added up. You can now see you could prevent this by using higher resolution files and only going down to cd quality at the end; in that case the rounding errors do get in the file but in the bits below the 16th (hopefully!) and still get discarded.

One thing you have to remember is that "noise" gets defined as material that's not part of the signal but that does not need to mean it's actually purely random. Noise in digital systems tends to come from a interaction between the program material (your music) and the sample frequency. Now considder how your ears work; your ears are looking for signals all the time and filtering out random noise (this is tigers v.s. the wind around your ears, your brain is very, very good at sorting those; your ancestors would have been dead if it weren't). they won't find a signal in the quantum noise of resistors but at the exact same low volume they may preceive this interaction in digital systems as a "signal". I'm willing to bet a good bottle of wine that if some method could be used to replace the aliassing and rounding errors in mid 90's experimental techno by the exact same amount of analogue noise (by RMS power) this would be preceived as less objectionable *if it would get conciously preceived at all*.

lots of gear lists the total harmonic distortion as a spec, only the good stuff lists the total inharmonic distortion. This is because you don't preceive harmonic distortion as a seperate signal but you WILL preceive inharmonic distortion of sufficient volume as a signal on it's own and it WILL casuse your brain to wonder wether this new signal is linked in any way to sex, violence or food (there is no way around the repitel part of your brain, terribly sorry). Digital systems typically have inharmonic distortion.

Ok, I'm tirered of typing now so we'll go into the trouble and joys of chorus later. Before the flames start; the second half of my two person bed is currently filled with "Microsound", "the Csound handbook" and "Digital Signal Processing; a practical aproach (second edition)" and I wrote over a thousand lines of programing code in the past few weeks, I like this stuff but if you are imagining it to be perfect you are setting yourself up for disapointments and hard to trace issues. I can't stress enough that there is no perfection anywhere, the best you can hope for is making good choices and trade-offs.

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PostPosted: Thu Jun 08, 2006 7:34 pm    Post subject: Reply with quote  Mark this post and the followings unread

Kassen wrote:


Ok, let's get back to practical DAW's. Say you are working on some project that is to be on a cd. Say you use only 16bit 44KHz wavefiles. (we have 90db or so of S/N ratio there and 22KHz of frequency resolution) If you set some plugins and filters lose on this these will internally upsample your material, process it, then downsample again and all the problems you would have had will be in the bits that get dropped at the end. The largest amount of noise introduced in this should be in the rounding to 16 bit at the end. Great; you can't hear that. Now another plugin, and another, and another.... And suddenly your material has that vague glass-like character that mid-90's "experimental techno" material had.

What's going on here? The errors added up. You can now see you could prevent this by using higher resolution files and only going down to cd quality at the end; in that case the rounding errors do get in the file but in the bits below the 16th (hopefully!) and still get discarded.


When I first started using plug-in effects, I was lucky enough to learn about this phenomenon soon afterward. I try to limit the use of plug-ins as much as possible. Because of my experiences as a kid using noisy tape recorders, I very quickly realized what happens when you tape a copy of a tape of a tape of a tape etc. Therefore, it was easy to apply this same logic to the plug-ins, and I didn't even let the digital/analog issue enter my mind. It was simply sound degradation.

Where I could get confused is why a particular mix doesn't sound right. If I were using analog tape and I wanted to know why a particular instrument was missing the low end and I isolated it and heard a horrible S/N ratio, that would be a quick and easy solution. With digital, if I can't hear these phenomenons which are obviously occuring, I would be left with so many options to try and fix it, including adding more/different plug-ins, but maybe those plug-ins weren't optimally designed, etc etc.
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