3 ms·
For a circuit meant to be used with a guitar or similar instrument, it is practical for many purposes to use a much lower sampling rate, e.g. 22 KHz. The frequ
by vibrolax 11y ago
For a circuit meant to be used with a guitar or similar instrument, it is practical for many purposes to use a much lower sampling rate, e.g. 22 KHz. The frequency content beyond 10 kHz is quite limited. I have used conventional LTspice to simulate several vacuum tube guitar/bass/microphone preamps I have built, for example [1]. I am tempted to try a real-time A-B comparison of the actual device with a livespice simulation.
[1] http://www.frontiernet.net/~jff/SonOfSVPCL/DIYSVTBassPreamplifier.html http://www.frontiernet.net/~jff/SonOfSVPCL/DIYSVTBassPreampl...
- TheOtherHobbes 11y agoAs soon as you create non-linearities in DSP you have to oversample to avoid aliasing. In fact it's usual in DSP distortion circuits to oversample by a factor of at least 4 or 8. (I"ve seen some devs use 256X oversampling...) I can't think of a reason why this wouldn't also be needed in an LTspice model. You can do distortion without it, but if you sample at 22k you're guaranteed to get that harsh halo around the midrange and high end that digital is famous for. Analog circuits don't have the same problem, because anything out-of-band simply gets rolled off. Unlike a digital system analog doesn't have a Nyquist limit beyond which frequency components reflect back, so a physical circuit should always sound smoother.
- vibrolax 11y agoGood point, that nonlinear processes generate frequencies not present in the input. One of the biggest challenges in making analog circuits with good-sounding clipping distortion is designing proper lowpass and bandpass characteristics into the coupling between stages to roll off undesired artifacts. The advantage in being able to actually listen to a real-time simulation modeled at the level of resistors, capacitors, inductors, and tubes or transistors is that one would be able to immediately hear the effect of component value changes, rather than just see them on the transient or frequency response plots. Modeling the system at the level of DSP blocks would be more computationally efficient, but then one is left to derive the equivalent analog circuit. It's not rocket science, but it does add a level of indirection. In any case, being able to simulate 5 triode sections and the linear filters between them in real time at a 48 kHz sampling on a modern CPU very well might be good enough for rock and roll. My use case is a design tool, not to obtain a substitute for a physical device.