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Reference [AH]:

 

This is an interesting point. Being an engineer and the developer of TimewARP 2600, I understand very well how emulations (at least the TimewARP 2600) are implemented.

 

It is a matter of fact that if you want more accuracy in your digital emulation, then more CPU power is required. One simple example of this is by looking at a pulse wave that is generating a 10khz signal on a computer setup with a 44.1khz sample rate. If you set the pulse width of the signal to 10% there is no real accurate way to represent it at the 44.1khz sample rate since there are only approximatly 4.1 samples per cycle of the waveform at that rate(for simplicity, let's not look at oversampling which can defiantly improve the resultant waveform). The easiest way to improve the accuracy is to increase the sample rate which also has the effect of multiplying the CPU usage by what ever the difference is. If you choose a 96khz sample rate instead of the 44.1khz sample rate, you now have 9.6 samples per cycle, which will give you a much better resultant waveform. This will cost you however by using quite a bit more CPU.

 

As far as modeling components rather than outputs, the TimewARP 2600 does this to a great extent now. There are certain modules in a synth that cannot be accurately modeled by components however. Oscillators are the biggest culprit here. Because of the fact that you are in a sample based environment, you MUST keep all generated harmonics below the sample frequency otherwise you will hear audible aliasing in the signal. This phenomenon applies to any digital signal, sampled or generated. As such, if you create an oscillator in software that exactly models the circuit (saw, pulse, triangle) you will defiantly get aliasing in the output signal. To produce signals that sound and behave like real analog signals in a digital environment, you must band-limit everything to be below the Nyquist limit (which is the sample frequency divided by 2). On the other hand, many components can be simulated using the components of the circuit. The best candidate for this type of emulation are filters, however, even in filters, you must be concerned with band-limiting. This is usually done with oversampling in filters.

 

One of the biggest areas of difference between most emulations an real analog has to do with the rate at which parameters are updated. Many emulations only update parameters and control cv sources at the digital frame rate, which is usually between 50 and 500hz. This of course effects the quality of the output signal. This is most apparent when you have patches that use higher frequency control sources. Audio frequency modulation is out of the question for emulations that use this scheme to update parameters and control cv sources. As a side note, TimewARP 2600 updates ALL parameters and ALL sources at the full sample frequency.

 

To the point that no two vintage analog devices sound the same, this is quite true. Alan R. Pearlman told me that when they were building synths they spent a great deal of there resources matching an qualifying components so the circuits would behave as consistently as possibly. There was always variations that they could not control. When you implement an emulation, there is inconsistencies go away and the resultant output is always the same for a given patch. In order to introduce "life" into the emulation, we do add stochastic behavior in certain places, otherwise the emulation would not feel right. It would not be out of the question to add features to an emulation to allow the user to adjust certain component values and thereby hear the difference, but we (at least not Way Out Ware) have not produced that product yet. If there is enough demand for it, then I would love to take a project like that on.

 

Best regards,

 

Jim Heintz

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:eek:
:eek:

That's mind-boggling.


My tiny brain can barely comprehend that.


Every parameter (so many of them too), is being updated 96,000 times per second.



That's probably not the way to look at it. I Am Not A Software Engineer, but it seems more efficient and likely it's scanning a snapshot of the synth's total state at audio rate, and not all of them simultaneously. I expect that there would be minor synchronization and major performance issues, otherwise.

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I just listened to a messe interview with John Bowen regarding the Solaris..

A few interesting things he mentioned:
USB port for uploading samples or wavetables
The 'rotor' which accepts 4 inputs and cross-fades between them, allowing wavesequencing type sounds. The fun bit is that the 'rotor' can run at audio rates, and act as an oscillator itself. Sounds interesting!

I'm kinda intrigued about this machine- it could be the digital machine to have

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:D
Found it!
:p
http://www.delamar.de/wp-content/uploads/2007/03/podcast_musikmesse2007_solaris.mp3




I just found this too! finally some juicy details from John himself! it's got wavetable [sounds like both waldorf and prophet VS tables are available?] and sample osillators in addition to V/A!- this is the Waldorf Wave killer :eek: - the "rotor" gives you a weird granular harmonic generator or simple wavesequencing-

HOLY {censored}! it is TOTALLY MODULAR! with the outputs of all the ocillators and filters patchable as audio-rate modulation sources :eek:

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Both VS and Waldorf wavetables are available on the Scope version of Solaris, so I bet he'll include them on the hardware version as well. What I'm wondering is if any of the Adern Flexor modules will eventually find their way onto the Solaris. :eek:

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Hmmm....an early Stromberg slayer.


^^^ AGAIN BOINGGG ^^^

 

 

I wouldn't be suprised. While I'd like W to succeed, the extra price you pay for this goes to lots of juicy controller goodness, and the uberclean oscillators of the Q will meet their match.

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I wouldn't be suprised. While I'd like W to succeed, the extra price you pay for this goes to lots of juicy controller goodness, and the uberclean oscillators of the Q will meet their match.

 

Yes, but the Strom will have those analog filters on its side. I am quite enthralled with the idea of the Solaris, but I wouldn't count Waldorf as beaten by any means. Let's see who can be first to market and what the real feature set and bug list looks like at that time.

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Yes, but the Strom will have those analog filters on its side. I am quite enthralled with the idea of the Solaris, but I wouldn't count Waldorf as beaten by any means. Let's see who can be first to market and what the real feature set and bug list looks like at that time.

 

 

 

we have heard the Stromberg's filters- they are the same as in the Q+/ASB16/ etc- pretty basic curtis filters- the Solaris has digital filters- but they are running at 96kHz from SharC DSPs- so they won't sound thin or flat like your run-of-the mill VA

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we have heard the Stromberg's filters- they are the same as in the Q+/ASB16/ etc- pretty basic curtis filters- the Solaris has digital filters- but they are running at 96kHz from SharC DSPs- so they won't sound thin or flat like your run-of-the mill VA

 

The Solaris could run the calculations at a bajillion QHz, but that still does not give it the marketing prowess in these circles that "analog filters" would.

 

Bah. Ima wait and see.

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