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Is OPL3 the same synth as Creative Music Synth [220]


Green Xenon

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Because the OPL3 is a digital synthesizer. It creates the waveforms using digital technology and then the DAC chip (YAC512) converts it into analogue so that you may hear it.

This effect can also be achieved w/out recording it into a wave file or recording it at all -- I.E. in real-time, OPL3's FM-signals are played through the voice-canceller and causes the effects I described.

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This effect is achieved w/out recording it into a wave file or recording it at all. In real-time, OPL3's FM-signals are played through the voice-canceller and causes the effects I described.

The OPL3 is a digital synthesizer. It doesn't matter if you are playing it real-time, recording it, or running it through a karaoke machine - the OPL3 is a digital synthesizer.

All digital audio is based upon sample rates and bit rates. 44.1kHz, 16-bit is CD standard. 96kHz, 24-bit is DVD standard. There are ranges below, above, and in between. If the audio was ever at any point in the digital domain (on your hard drive, on a CD, generated by a digital synthesizer such as the OPL3), then it was at one point digital audio with an accompanying sample and bit rate.

In order for you to hear digital audio, it must be converted to analogue. This does not mean that it was never digital - if audio is generated by a digital synthesizer (like the OPL3), then it is a digital signal that must be converted to analogue in order for you to physically hear it. In the case of the OPL3, the digital OPL3 chip generates a digital waveform, which is then converted into an analogue sound by the YAC512 digital/audio converter chip. Therefore, if you hear any sound from an OPL3 synthesizer, regardless of what you ran it through, then it was at one point a digital signal with a sample and bit rate.

After an audio signal has existed as a digital signal, whether it originated as a digital signal or if it was converted even if for a split second, any artifacts from that digital existence (a low sample rate, a low bit rate, an error in the sampling process) will remain with the audio even once it has been converted into analogue. These sonic artifacts (such as clipping, noise, or slight digital errors) are permanently a part of that audio unless you can find a way to carefully remove them. It may not be digital anymore, but the audible effects of its digital past are still there.

When you created the samples you have shown us, there was a very slight difference between the left and right channels of the stereo file, which can be seen by viewing the 44,100 individual samples of the file itself. This could have been created by many things, one of which could very well have been the OPL3 chip itself. Regardless, this audio discrepancy became part of the signal and remains with the audio even once it has been converted into analogue. It's there when you record it, it's there when you play it in real time, and it's apparently there every time you have used the OPL3 digital synthesizer.

This discrepancy appears to manifests itself in the same way every time, and that is the way that Stryd demonstrated in his earlier post. This discrepancy is the reason you hear the effects you are hearing when you invert the phase of the sound file you included. Please, try out his experiment for yourself and see what happens. It may make more sense to you after witnessing it first hand.

I think that your difficulty in understanding what we are trying to say results from a confusion about how digital audio works. That's fine - digital audio is a dense topic, and it functions in a manner completely different from analogue audio. You can very easily understand one and not the other. However, it is a topic that can sometimes be difficult to fully grasp without diagrams and elaboration upon miniscule details. I sincerely hope that Stryd is able to piece together some sort of pictoral demonstration as I am unsure what else to say short of an extended discussion on the fundamentals of digital audio, which I believe may not be the best solution and would be better achieved through a book rather than an internet forum regardless.

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This is a bit OT but...

OPL3 is a digital FM synth...

Which means it generates digital FM signals...

Initially [before passing through an ADSRs, VCAs or other amplitude processors], an OPL3's FM signal will have a constant peak-to-peak amplitude...

Since these are digital FM signals with constant peak-to-peak amplitude with more than two frequencies, would these digital FM signals qualify as MFSK [multiple-frequency-shift-keying] signals?

Binary FSK:

http://en.wikipedia.org/wiki/Frequency-shift_keying

Multiple FSK:

http://en.wikipedia.org/wiki/Multiple_frequency-shift_keying

Seems like an FM signal from OPL3 would initially resemble an MFSK signal, because it is a digital signal that changes only by its frequency and its peak-to-peak amplitude at first remains constant.

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Wrong kind of FM

But I guess if you were a total freak you could synthesise a data stream, in which case the correct answer to your question is: How many operators are you using? What waveform? It depends on the patch.

I will revert to my old question: Why?

Just build the damn thing already! Enough talk... Build one and try it! Or... buy my LCDs and I'll tell you all about how it sounds :D

And if you're talking about digital data transmission then I don't think I need to draw those pix after all ;)

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Easier said than done.

I can understand your hesitation here, but I think the amount of time you have spent preparing samples and debating digital topics which you do not seem to fully understand is perhaps similar to the amount of time it would take to research and prepare the means to construct a streamlined MBFM synthesizer. DIY projects can be expensive of course, but one appeal of the Midibox projects is that they are quite modular so you can easily mold them to your needs. It doesn't actually take much time or money to construct a working FM synth - the price and time constraints don't really shoot up until you start building elaborate control surfaces and customizing them to your own desires. Since you have mentioned that you have spent time synthesizing within your computer using the OPL3 Soundblaster cards, I don't imagine you having any difficulty adapting to the JSynthLib SysEx editor. Using that you won't need a control surface and could fit the entire MBFM synth into a small box or corner. There are many people here who are willing to help out as long as you demonstrate that you are both willing to listen and willing to work.

No I am not talking about digital data transmission.

Perhaps you aren't intending to, but the topic of frequency-shift keying certainly concerns the matter. "Frequency-shift keying (FSK) is a frequency modulation scheme in which digital information is transmitted through discrete frequency changes of a carrier wave." Taken straight from the Wikipedia link you posted.

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The OPL3 (and other "frequency modulation" synths) actually use phase modulation instead of frequency modulation.  The reason is that a DC offset (eg non-zero amplitude at 0Hz) will give a frequency shift in FM but only a phase shift in PM.  See  http://www.nic.funet.fi/pub/sci/audio/misc/pm-intro for more information.

That maybe why I was getting the effect I described when phase-inverting one stereo channel and combine it with the other.

I would guess that in stereo PM synthesis, one channel would naturally have a phase that's different from the other channel.

Each stereo channel has its own tone generator.

Again, this is just a guess. I could be wrong.

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