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Originally Posted by
monomer
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I was actually refering to timing glitches between the DAC ic and the resistor network DAC.
That's not going to affect the output. Most DAC designs produce some manner of glitch when the code changes, because the bits don't all change at the same time, but the sample and hold eliminates that.
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I would expect slew rate problems, if any, to be present mostly in the op amps. I did check the datasheet for the first amp following the DAC ic and it turns out to be a specifically high bandwidth device that is pretty fast and has a pretty low overshoot. Didn't bother to look into the actual op amps in the filters and buffers but at least for the DAC ic part they specced it pretty well.
Right, it would be the DAC buffers or the sample and holds that slew. But probably that's not happening to a significant degree.
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Well, software is inherently more loopable and branchable think. Usually one programs for a single core. So it's natural to re-use parts of code.
But with hardware you can more easily design a parallel system. I think this is what is surprising about these yamaha designs, that it's more economical to implement the algorithm in hardware with a single envelope and operator.
Of course it cost a lot more to populate silicon in those days, but it's a nice insight nevertheless i think.
Replicating everything a number of times on the same IC would take up too much space. There are only a few things that do that. Most things use a single phase accumulator, a single multiplier, etc. and time multiplex to achieve polyphony. Then the polyphony is limited only by the clock frequency, desired sample rate, and the number of registers. The only exceptions are cases where the hardware can't be clocked fast enough to do this, or when it's using clock divider ICs or other things that can't be time multiplexed.
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Originally Posted by
madtheory
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It is an amazing design, everything they did to make it work is totally not what we'd do these days with super powerful general purpose CPUs (Yamaha were unique among synth makers because their own chip fabrication since about 1970, so "populating silicon" was not an issue for them).
The remarkable part is that it's pipelined well enough to calculate one operator per clock, doesn't use a multiplier, and greatly outperforms everything else that was available at the time. All the algorithms are implemented with just a couple registers and multiplexers. Modern romplers, etc. are still based on custom ICs that work in much the same way. So it's not all software even now.
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Originally Posted by
living sounds
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I can only speak for myself. It's my ears telling me they don't sound the same. The software part is easy for the most part, but the sample rate conversion and converters are obvious starting points.
Accurately emulating the internal workings of mostly undocumented custom ICs isn't "software". There are a number of things in there that are quite difficult to correctly reverse engineer.
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But we're not even sure the samplerate has much to do with the specific sound of the DX7. I doubt it.
The specific sample rate mainly just affects the aliasing. No one seems to notice particularly that the Yamaha V50 has a different sample rate from the TX81z (50 kHz vs. 55.930 kHz).
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Originally Posted by
DrewE
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Has anyone done any critical comparisons to see if all DX7's sound identically the same? I would not be the slightest bit surprised if there is as much variation between "identical" DX7's as there is between good software equivalents and the real thing.
The DAC distortion will vary from unit to unit.