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How should I describe Digital Synths?
MUFF WIGGLER Forum Index -> Modular Synth General Discussion Goto page Previous  1, 2 [all]
Author How should I describe Digital Synths?
Exhale
I'm yet heard a AD/DA chain that sounds like a copper wire.
Always something imparts, some blurring occurs.
Even on PRISM ones.

So yeah, in theory, after reconstruction filter, recorded waveform is similar to it's original analog form.
But some errors always present.

Try to take some snappy analog synth, no matter what.
I did it with Waldorf Pulse.

Feed left output to analog mixer, and right to ad/da chain and to the same mixer. A/B results.
Significantly less attack. No matter what you do, digital will smooth the sound.
But in theory, yeah, 96 khz is good enough to capture all details.
But....... human ears are very sensitive.
Espesially in midrange.

That's why I like to jam with analog synths pluged directly to analog mixer, little bit of fx, and then to speakers.
After digital stage lots of life mojo is lost.
Yes Powder
THIS ARGUMENT NEVER GETS OLD
Peter Grenader goatse.cx
Dave Peck
I agree with Cornutt. Asking 'how do I describe a digital synthesizer / digital synthesis'? is a bit like asking 'how do I describe weather'? There are a lot of different kinds.

There are several different types of synthesis that can all be done using digital synthesizer hardware & software (virtual analog subtractive, additive, FM/PM, sampling, wavetable, etc.). Some digital synths use only one of these types/methods of digital synthesis, some allow you to use more than one method.

So in general, it is any synthesizer that uses digital hardware and code to implement one or more of various methods of synthesis to create sound, as opposed to a synthesizer that uses analog electronic circuits to do this.
milkshake
Exhale wrote:

Feed left output to analog mixer, and right to ad/da chain and to the same mixer. A/B results.


That is not how you test these things, your results are meaningless.

Perceptual testing is very difficult, if you don't know what you are doing don't do it.


A simple test that everyone can do is to mix a song - 60dB below an other song. Can you hear that song?
flts
milkshake wrote:
nigel wrote:
milkshake wrote:
One other often misunderstood thing is that digital audio is a stream of numbers that represent a continuous signal. There are no steps in a digital signal. The confusion arises from sample editors. What you see in a sample editor is NOT the actual signal!

The digital "signal" is just a sequence of sample values, so really it's nothing but a series of steps. However it can be used to exactly reconstruct the original (bandwidth limited) signal.


A digital signal only has 2 values, that's what makes it so robust.
Iow digital audio is a row of binary numbers.

When you say steps, people confuse it with the picture in a sample editor.


This is getting into semantics territory, but as semantics is lot more fun to discuss than the endless gearslutz-level digital vs analog perceptual issues with flawed testing methods and the whole psychological shebang...

You're both right, and to understand the whole picture, both interpretations / meanings of the term are needed.

You are talking about the electronic, low level view of "digital signal", which is a binary (aka logic) signal of low voltage level and high voltage levels, 0 and 1.

Nigel is talking about "digital signal" in the context of digital signal processing (DSP), where that term in turn actually has the specific meaning of the discrete-time, discrete-amplitude sampled data - that is, the exact "waveform with steps" that one can zoom in on sample editor.

Ie. on a low level, your standard binary digital computer encodes sampled audio to binary data (0, 1). The signal processing algorithms, sample editors etc. process it on a level of abstraction where eg. a second of audio might be represented by 48000 samples, each of which can have an amplitude value from 0 to 65535 from an evenly-spaced quantized set of amplitude values (in the specific case of 16 bit unsigned, 48 KHz audio data).

Even though each of those example samples is, on a low level, represented by a row of sixteen "0"s and "1"s, on both abstraction levels a "digital signal" is the correct term (or one of the correct terms) to use. The context just needs to be mentioned.

And, looking from another angle, whether you want to say that the signal is a sequence of bits (from information-theoretic / electronic / cpu architecture / low level software dev point of view) or a sequence of numbers with given precision corresponding to quantized amplitude levels (from a DSP perspective), the Nyquist-Shannon sampling theorem says that there is no reason why, given high enough sampling rate to match the source signal, you couldn't reconstruct the original analog signal perfectly from that digital signal.
matthewjuran
A consideration when forming an opinion of an electronic instrument’s sound is that distortion doesn’t come just from the signal parameters.
cycad73
Digital synths reached their peak with the PPG 360 or maybe Alles synth/Crumar DGS and was all downhill from there... listen to early Laurie Spiegel or the first Rolf Trostel album you will be amazed at how these early digital synths can sound... in other words tone matters... musicality matters... the analog parts and specifics of the HW implementation above all matter....

samplers/romplers too, the best are like the Emulator I/II or the Kurzweil 250, which had separate voice cards and mixed all voices in an analog stage, the Emu also did NO DSP/sampling rate conversion, there was a separate variable-rate clock for each voice so you never had aliasing, if you needed a different pitch you simply clock the sample at a different rate (an analog operation). An engineer would look at that today and say there's a horrible overkill in components, we can simply do all this in software. but in the early 1980's all these tools were sold for > $10K because the technology had to prove itself musically against analog synths and acoustic instruments. so there was little incentive to make the process cheaper, so you had whatever was needed to get the best tone.

I am especially curmudgeonly these days after poring over 1 hr+ of google Nsynth videos and not hearing at any moment a single usable or inspiring sound. Just because everyone is an expert with software these days does not mean they can design something for use in a musical context. I have most respect for people like Olivier/Mutable or the mungo people who take ideas from the algorithms world but are overall agnostic as to the exact mix of technologies, because the bottom line is musicality.
commodorejohn
cycad73 wrote:
the Emu also did NO DSP/sampling rate conversion, there was a separate variable-rate clock for each voice so you never had aliasing, if you needed a different pitch you simply clock the sample at a different rate (an analog operation).

This is definitely an interesting facet of the design; effectively, this means that any noise introduced by the sampling and reconstruction process becomes additional harmonic (or enharmonic-but-tangibly-related) content in the output. I think this is also part of what makes the Amiga sound so distinctive.
cycad73
commodorejohn wrote:
cycad73 wrote:
the Emu also did NO DSP/sampling rate conversion, there was a separate variable-rate clock for each voice so you never had aliasing, if you needed a different pitch you simply clock the sample at a different rate (an analog operation).

This is definitely an interesting facet of the design; effectively, this means that any noise introduced by the sampling and reconstruction process becomes additional harmonic (or enharmonic-but-tangibly-related) content in the output. I think this is also part of what makes the Amiga sound so distinctive.


True, thanks! I should have corrected to say you don't have *inharmonic* aliasing (the objectionable kind). in a wavetable situation where you have a periodic and non-bandlimited waveform like a sawtooth you of course generate aliases by sampling, but if the sampling rate is an integer multiple of the fundamental these aliases fall exactly upon harmonics, meaning you have just a slight (and in most cases imperceptible) adjustment to the harmonic spectrum.

In fact the best kind of sound is when you forego the reconstruction filter and do a direct zero-order hold reconstruction, then you have a nice collection of spectral *images* at higher frequencies which are still integer multiples of the fundamental. Adds nice texture and grit without the bad sound of inharmonic aliasing wink

The Oberheim drum machines (DMX/DX) also used this technology, and I assume also the Linn machines and many others of the period although I've never owned the latter (always have been too much $$). If you pitch-shift a sample upwards on the DX it sounds in fact quite good, it's just played back at a higher rate.

There's also an overlap here with divide-down architectures, but that's another story...

Bottom line is it's historically been the *analog* parts of the architecture that lend digital synths their character, as much if not more than the algorithms employed. It annoys me to no end how engineers blindly apply the "principle of sufficient software", they devise more and more sophisticated DSP techniques to better approximate solving a problem that would just disappear if they focused on the analog part of the design or in general were more flexible about the paradigm.
commodorejohn
cycad73 wrote:
It annoys me to no end how engineers blindly apply the "principle of sufficient software", they devise more and more sophisticated DSP techniques to better approximate solving a problem that would just disappear if they focused on the analog part of the design or in general were more flexible about the paradigm.

Hear, hear.

Of course, that's because hardware solutions affect the cost per unit, while software solutions are amortized over every unit sold.
ludotex
applause we're not worthy
Drakhe wrote:
artieTwelve
Analogue synth = Continuous semi controlled chaos using as it's medium a subset of the electromagnetic spectrum.
Digital synth = Algorithmic semi controlled chaos that needs digital-to-analogue conversion to work in the same spectrum as analogue.
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