Yamaha DX7 Technical Analysis
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Chipsounds is great if you're into 8-bit / chiptunes and the like (they also have a "chipsynths" series of plugins emulating SNES/MEGADRIVE, and other classic game consoles, but I haven't tested them...)
And Bidule is quite a thing. Is some sort of modular VST synth with tremendous options for routing that seems very well suited to prepare live performances. The Philip Glass Ensemble, through Michael Resman has used it at least once to make complex live setups: https://www.plogue.com/michael-riesman.html
I have TX7 for and the Arturia DX-7V. Typically FM or Phase Distortion(Casio CZ synths) arent really the type of sound I go for, but they definitely shine in clanky metallic dissonance.
If you take a DX7 patch and implement it on something like the PreenFM2 or PreenFM3 (which uses actual FM), any algorithm with more than a few operators is going to sound quite different.
I still love the old Yamaha FM synths though, the lo-fi-ness of the DACs are so musical/magical.
Here's a IDM track I made with just the Yamaha TX81z 4op FM synth. https://soundcloud.com/cassilda_and_carcosa/ontologies-tape-...
It’s true that it’s phase modulation, but phase is just the derivative of frequency, and the derivative is a linear operator (essentially a 6dB/oct EQ). So I don’t think it’s an especially important piece of terminology to get correct.
The statement about feedback is actually backwards. You cannot directly incorporate feedback into a phase modulation system. In the Yamaha system, the feedback is delayed by a sample to make it work. In an FM synth, this would not be necessary, and you could just directly feed the output back into the input.
The guy behind the preenfm2 says, that feedback is not that important, when you have a variety of waveforms for modulation. Linear FM also has the advantage of being a bit "friendlier/warmer" when using complex waveforms as modulators (Higher harmonics of the modulator will have a smaller effect than in phase modulation).
One thing you loose when not having feedback: The normal feedback happens after the amplification/envelope if I remember correctly Therefore the amount of feedback scales with the amplitude. With feedback you therefore get less harmonics and therefore a "softer" sound at lower amplitudes. This is quite useful, because it mirrors the behavior of most natural sound sources. It is the same idea as using a lowpass gate or using the same envelope for the amp and the filter in subtractive synthesis. Of course FM synths give you enough options to get this effect in another way. I still like the yamaha style feedback though....
This is just my personal experience. I often dig through DX7 patches in order to reimplement them for demoscene projects.
Sounds like you’re getting good use out of the TX81Z. I have the TX802, which is on paper a better synth, but controlling it from a DAW is much more of a pain because the program change messages control the “performance”, rather than the patch. So it’s been exiled from my rack, at least for now.
Slightly related: I really love the build quality of Yamah's 80s/90s rack synths. The physical interfaces were very well-designed, and look amazing.
I wish i had a TX802 or a fs1r yamaha. Those are so nice!
FM (if strong enough to create negative frequencies) and PM are equivalent when you're modulating a signal with a sine wave. As soon as you chain modulators, they're no longer mathematically equivalent or capable of producing the same sounds, since frequency-modulating a signal by a complex wave is different from phase-modulating a signal by the same complex wave (which can produce sharp corners given a bounded input with sharp corners).
The one-sample delay basically turns the feedback into an iterated function system. With certain parameters, you end up with aperiodic results—cool in theory, but in practice, often just used as a noise source.
For certain values of x and c! You will end up with branch cuts for |c| > 1.
And in John Chowning's original article, he bops back and forth between the two without mentioning it.
In practice, phase modulation is more useful and makes more sense. And yes, the DX-7 uses phase modulation throughout.
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With Frequency Modulation, you can have an operator feed back on itself. However... with such an arrangement it is entirely possible for it to get stuck at 0 Hz. Not very musical.
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And phase modulation makes a lot more sense. Watch as I demonstrate.
Student: "How does FM synthesis work?"
Teacher: "Well, do you know what happens if we have one oscillator modulating the frequency of another oscillator?"
Student: "You mean like vibrato?"
Teacher: "Yes, exactly. Now imagine a vibrato at the same frequency as the second oscillator."
Student: "That's doesn't even make sense."
Teacher: "Not to worry, the Bessel Functions still apply."
Student: "What's a Bessel Function. Can I calculate it?"
Teacher: "Not directly. It's kind of complicated."
Student: "Mutherfucker!"
Okay, now let's try phase modulation.
Student: "How does phase modulation work?"
Teacher: "Okay... Do you know how distortion works?"
Student: "I'm a guitar player. I'm intimately familiar with distortion."
Teacher: "Great! So distortion is a nonlinearity on the Y axis of a waveform."
Student: "Yep!"
Teacher: "Well, phase modulation is a nonlinearity on the X axis of a waveform. We stretch the waveform here [hand gesture], and compress the waveform over here [hand gesture]."
Student: "Hmm, warping the waveform along the X axis. That sounds kind of like PWM."
Teacher: "It's kind of similar, yeah."
Student: "Cool; thanks. I understand."
Thanks!
The customer kept adding features, and when I was all done, I had all the code in 511 bytes.
All a phase accumulator is is a long register you add to, and only take the top N bits off of. It's quite easy to have 64 bits of frequency resolution, regardless of how many bits your SINE table lookup is going to the DAC.
What a waste ;)
It's funny how this works though: customer makes yet another request, at first your response is 'forget it'. Then, late at night when the house is quiet you really start to think about it and bit by bit the solution starts to present itself. Then a highly annoying bout of code golf later you manage to squeeze the feature in, and end up with more free space than you had before...
Asking for a friend :-)
I recall that HC11 version too! I bought the PLCC version and made my own dev board with a through hole PLCC socket :-)
[edit] memory just caught up to me. The HC05K1 had 504 bytes of program memory, not 511.
https://gearspace.com/board/electronic-music-instruments-and...
check post #34 from Charlie Clouser who toured with them in that era and is also a pretty well known producer/engineer
Nope, just destroy it because you or your audience are knuckledraggers.
Why this was allowed to be patented? This is a pretty much obvious thing to do when you have this kind of limitations.
It seems like people reviewing patents don't really understand what they are about. There is plenty of stuff like this in audio space.
E.g. this one https://patents.google.com/patent/US20170060527A1/en
"If you have a lot of patents and somebody comes after you, chances are one of your patents will overlap enough with one of their patents that you can negotiate a deal so nobody gets hurt. Whereas if you don't have anything to offer and you have nothing in your stable of patents, then you're stuck"
The article I sourced this quote from can be found here: http://summit.sfu.ca/item/7720
I just chalked this kind of weird specificity up to the legal minutiae required to patent musical hardware. Maybe someone with legal experience can weigh in on this. I'm guessing that if you go into this level of detail regarding the technical implementation, it probably makes it much easier to litigate patent infringement later, and narrows the possibility that someone else can get away with a similar implementation. The titles of some of Yamaha's other musical patents are pretty obscure: "Electronic musical instrument with user programmable tone generator modules (5,040,448)", "Electronic musical instrument capable of varying a tone synthesis operation algorithm (4,558,857)", etc.
https://120years.net/tag/ccrma/
When Chowning met Dave Bristow, who was Yamaha's product specialist and designed many of the DX7 presets, he apparently asked if Bristow could explain FM to him because he didn't really understand how to use it.
This was partly a joke and... partly not. It was also before they wrote this together:
http://www.burnkit2600.com/manuals/fm_theory_and_application...
https://120years.net/wordpress/the-synclavier-ii-new-england...
> Somewhere in this lull, a tiny Vermont-based synthesizer company aptly named the New England Digital Corporation beat Yamaha to the punch by producing the world’s first digital synthesizer, the “Synclavier.” Though only 20 units were sold at $41,685 each, and they were all reserved for top-notch musicians, Stanford took no chances, and swiftly sued the company for infringing on its FM synthesis patent. From that point forward, the university received a sum of $43 every time a Synclavier was sold.
Yamaha had been working on FM synthesis for a few years already at that point.
Also, technology for digital synthesis was already there, the big game was to avoid patent infringement from other companies.
For example, Casio invented the "Vowel–consonant synthesis" used in their 80s Casiotone keyboards mostly to prevent conflicts with an existing patent from Allen Organs[0]. And later, they came out with the "Phase Distortion Synthesis" as an *alternative* FM synthesis[1].
[0]: See the bottom of this page: http://weltenschule.de/TableHooters/Casio_CT-410V.html
Most PC keyboards leverage this so they can skip diodes, and just route the matrix so that common simultaneous combinations (modifier keys, WASD for games etc.) don't interfere with each other.
so:
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and | | | | | |
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Would look identical to the CPU without diodes.So in essence because the diode solution requires rows times columns diodes and this one only has a couple of parts per row and per column it scales better and for larger keyboards will end up with a substantially reduced parts count and a simpler circuit.
That patent is quite a bit younger than the synth though, so it's kind of logical that that technique would not be used, and my hunch on why they used those diodes back then is likely correct, even if we have a smarter way of doing this now.
Turns out the rubber contact strip was installed inverted. I think this one didn't have notches to make sure it was oriented correctly. After flipping it, notes sounded as expected.
> Yamaha CEO Pleased With Current Production Of Jet Skis, Alto Saxophones, Snowmobiles, Power Generators, Scooters, Golf Carts
An all time fav
By the time it was ready for release, Apple released their own driver, so I just open-sourced my work.
I gave the DX7 to someone. Can’t remember who.
They are incredibly durable, one that I got had been in a fire but it still worked, even though more than half of the keyboard had been burned off. The mainboard went on to repair one that was otherwise good but had a fried CPU. Another one I got had been owned by a touring musician, who apparently didn't know about flight cases :) There wasn't a lick of paint on the outer case. I got it for free because it wouldn't turn on any more. Turned out that the voltage selector had somehow been moved in transport and got stuck between the two settings, leaving it disconnected. That was the fastest fix ever, there simply wasn't anything wrong with it. (And good thing that voltage selector had not been pushed a tiny little bit further because then it would have been end-of-story for sure).
So, I have a pretty large collection of spares, looms, main boards, display boards, keys (they do break, especially as the plastic ages it can get more brittle, more so in synths that have been in the sun), switches, PSUs and so on. Every now and then I put an ad out to get people to sell me their broken ones or to help them out and then a new batch of working DX-7's goes back into the world. It's never going to make me a dime, especially not when counting the time but it feels pretty good to keep these oldies alive, and to see how happy musicians are when their baby works again.
The best part is being able to map the sliders to /almost/ any parameter within a patch, so it's possible to have expressive control over the tonality of your instrument while playing live. Sure it involves menu diving to figure out ahead of time, but I'm a programmer. It's really more about the joy of being like "this sound needs more wonk, lets tie this parameter to the output of this other function and ooOOOH not what I was expecting but let's go"
If you ever can't find a part let me know.
Mostly though I've just got some weird dirty hiss (sounds like a rainstorm) on the output all the time that makes it unsuitable for recording. I use DEXED + midi in that case, but it's still fun on it's own (also weirdly I find menu-diving to edit patches easier than being presented with that big DEXED UI)
Thanks for keeping these things alive
They were dirt cheap and as mentioned, insanely solid.
Same thing happened to to the Casio CZ line. I bought one at a time they were very undesirable for $75 and they now go for $500+
There's a financial lesson in here somewhere
That and the DX7 was used to death and more so - 99% just used the pre-sets and got to stage that you could name the sound they used on a sound and all got cliché.
Opening up the DX7ii (and other synths from the same time) is a pain compared to the nice hinge construction you get on the original DX7.
About 200K DX-7's were made, which is enough that you can find them and not so many that there is no value in repairing them, on top of that the degree to which people are attached to their synths is something that still amazes me. The level of emotion when a dead synth comes back to life with the owners is always quite gratifying, it's some of the best time spent for me.
Next time I have one of these die-hards I'll be sure to take pictures.
> As convoluted as this sounds, it makes more sense when visualised: [1]
...but does it??
[1] https://ajxs.me/static/img/articles/dx7_technical_analysis/f...
Notice how, when the modulator wave is at its top the FM wave is very short, and when the modulator wave is at its bottom the FM wave is very long.
FM = adjust the frequency (width) of the carrier based on the absolute value of the modulator.
[1] https://web.eecs.umich.edu/~fessler/course/100/misc/chowning...
I think in this case the clearest explanation is the equations + graphs:
1. modulator(t) = sin(t)
2. carrier(t) = sin(t*4)
3. result(t) = sin(t*4 + d modulator(t)/dt) = sin(t*4 + cos(t))
http://www.fooplot.com/#W3sidHlwZSI6MCwiZXEiOiI0K3NpbigxMCp4...EDIT: it's wrong, see below
If you step through the lookup table with constant size steps, you would get a sine wave. If the steps are not constant size then you get a distorted sine wave. The "rate of change of the phase angle" would then be the step size.
What is usually called frequency, and has the same value in boring cases like an unvarying sinusoid, is the inverse of the period of the resulting sound as someone would hear it, which is not instantaneous and needs on the order of one period's worth of samples (usually more) to estimate, for a certain definition of period (e.g. interval between zero crossings) or a certain mathematically reasonable calculation (e.g. estimating the frequency of the strongest component from a fixed-length window of past samples).
Everyone is absolutely right that this is the same as frequency. I think I was used that particular poetic flourish to emphasise how complicated the academic, mathematic explanation is, in contrast with the diagram. Quite likely I picked up that exact phrasing of "phase angle" from some academic literature in particular.
That's a lie, because:
> I think I was used that particular poetic flourish to emphasise how complicated the academic, mathematic explanation is, in contrast with the diagram
is a totally different to what the person you are replying to said
In an AM radio signal, the frequency of the carrier is constant; the amplitude of the audio modulates the amplitude of the carrier.
In an FM radio signal, the amplitude of the audio modulates the frequency of the carrier. That's the difference!
In musical terms, I very much like some analog, and some FM, synthesizers. But certainly, back in the day, getting some rich sounds out of FM synths was a lot cheaper than analog. (Just in terms of the cost of patch cords alone ;-) Without Chowning there wouldn't have been a synthpop era!
This is not correct, unfortunately. Aliasing is a time domain artifact, affected by the sample rate. The limited table size instead causes distortion, which affects the values.
If you ever run a bitcrusher plugin, you’ll see two sliders—one for bit depth, one for sample rate. Aliasing is affected by the sample rate. This is kind of like the other slider—the distortion slider in a bitcrusher plugin. Although it is like having not one bitcrusher, but many bitcrushers, reducing every signal inside the synthesizer to about 12 bits. Kind of. The distorted signal is evaluated at some sample rate, which also introduces aliasing (in addition to the distortion).
These effects are musically quite distinct. Aliasing is not generally harmonic, distortion generally can be (if you arrange it to be so).
The inverting input of IC56 might be a good place.
I originally got very interested in this when discovering the YM-2151 chip that made 80's arcade game sounds was also found in pro-audio keyboards. They used a very similar YM-2164. This got me to explore the whole DX line including DX-7, DX-9, DX-27S and that keytar that's still popular for bass and talkbox, the DX-100.
By the way, if anyone is interested in a modern, affordable, nostalgic incarnation of the DX7, check the “reface” edition.
Oh, and it's even more unbelievably massive and heavy. I will never part with it - they are the most criminally underrated synths, and it's crazy they are still often priced lower than the DX series.
It would be great if someone would built a better live editor for them though.