Bass amp electrical properties and DSP

Hi there!

I’m an electrical engineer and recently picked up playing bass (a TRBX174!), so I thought it would be fun to try designing my own solid-state amp. However, I’m unfamiliar with the electrical properties on consumer basses and bass amps and would like to design mine to be roughly as good. I’m shooting for 100W RMS output.

I would like to know things like:
What sort of input voltage range (Vp-p) do amps need to tolerate? Do active and passive basses have massively different output ranges?
Why do many common amps have different attenuator dB values (I’ve seen -6 to -15)? What’s the criterion for choosing these?
I would guess that almost all amps contain several op-amps; what’re usually their rail voltages?
What does a typical frequency response of the onboard EQ look like? How much boost/cut is usually possible, and what frequencies does each band cover?
Big potentiometers are noisy; how is the preamp gain knob set up to avoid this?
What’s a good speaker frequency response range? I understand that the low-B (31 Hz) and low-E string (41 Hz) are heard more by their harmonics, so is a speaker with a corner frequency of 50 Hz good enough?

Most of these questions are about the preamp, as I have a unique DSP idea for the power stage that I’d like to try. All of them might be answered with a single schematic or picture of a PCB, so if anyone has one for a common bass amp (like an Orange Crush or Fender Rumble), that would be greatly appreciated.

Thanks!

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Hi @sevenskiesrain, welcome to the forums! I’m going to summon @DaveT, who I believe is a sound engineer of sorts and might be able to answer your questions.

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And welcome aboard!

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Think we shared some schematics from the Fender Rumble too somewhere on this forum. Best to do a search :eyes:

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To clear up any mystery, I have a degree in electrical engineering and theatre sound design. I’ve spent the last 32 years designing audio, video and control systems for new construction or fixed installations across a variety of applications including performance spaces, critical listening rooms, cruise ships, theme parks, education and lecture facilities, training and simulation facilities, operations command and control centers, one private yacht and two private residences.

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Sounds like a fun project!

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Fantastic, this is very helpful; much appreciated!

As an extension to this (and this is open for anyone to answer), I’ve noticed that higher-end amps tend to have a separate plug for the attenuated input while others have what appears to be an SPDT switch routing the signal to the attenuator (the former is shown in this schematic). Is one better from any standpoint?

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Here’s something EBS does that I think is just brilliant. Wide boost. Narrow cut.

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This person measured and graphed a bunch on tone controls . . .

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Thanks, this is really insightful!

If it helps, I can explain what I plan to put after the preamp. I’d like to sample the signal with an ADC, apply the EQ in the digital domain and feed the signal to a Class D amplifier. As you probably know, precision ADCs don’t have massive voltage ranges, so I figure the range I get out of my preamp gain knob should reflect that.

The issue then becomes: where should I clip my signal? The schematic above shows the signal being clipped at ±15V shortly after input, but I might actually need a second diode clipper between the preamp gain and the ADC to protect it, right?

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Analogue Devices say . . .

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Some Bergantino Tone Controls . . .

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I’m an EQ nut, so I like all the fancy controls. Many people do not. The SansAmp VT Bass DI is very popular and it only has . . .

image

I use parametric EQs, but if I had to have them fixed I’d want this set . . .

image

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Thanks for the information so far. A few questions about the digital domain:

I’ve found an audio ADC with an ENOB of around 20 bits and a microcontroller that’s barely powerful enough to do the transformations necessary. The system should work with and output 48 kHz. I can do FFTs on 16-bit signed integers (which the ADC outputs by default) or 32-bit floating point numbers. The former runs approximately 5 times as fast, so I could keep buffers smaller and reduce latency by using it.

The questions:
Do I really need to be working at 48 kHz (Nyquist frequency 24 kHz)? I feel like working with the entire audial range means I can accomodate basses and guitars’ overtones, but if there are some parts that I can reasonably lowpass off, that would make each FFT bin a lot more effective.
Will signed 16-bit resolution be no worse than a typical amp, or is there a dynamic range-related merit to using 32-bit floats?
How much latency do you think is noticeable/bad between plucked string and amp output? This linearly decides the number of FFT bins I can get.

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@terb has built multiple amps. He should be able to answer some questions.

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In my opinion those parameters really change the feel of the amp when you’re playing the instrument, if we’re speaking about a digital preamp. Also when the signal is recorded, it has to be modified (typically adapted to a specific dynamic range) so you’ll need some more information than what will be left after that.

44 or 48 kHz is not something really noticeable (the best thing being a switch to change from 44 to 48, which means that the signal can avoid being converted when recording), but the 32 bit range is mandatory in my opinion.

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Thanks for the answer on samplerate! With respect to the dynamic range, my guess is I will have to adjust the amplitude from the preamp using the gain knob to exactly fit the voltage bounds of the ADC without clipping, just like with analog equipment. The rest of the dynamic range problem comes down to the digital representation: the ADC outputs binary integers (only the top ~20 bits of which are not noise), and the Class D amplifier on the output also receives integers, so wouldn’t doing the parametric EQ in floating-point numbers just eventually force me to round to the nearest code before I can output?

It’s not that I can’t do floating point, it’s that the sound buffers would have to be longer to accommodate, so latency would go up. Let me know if you have any opinion on “acceptable” DSP audio equipment latency times as well.

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well yeah, in this case you will be in 20 bits but that’s still way better than 16 :slight_smile: it’s not a big problem in my opinion if you have to round before the power amp.

I’m not sure about the “acceptable” latency, it seems kinda very personnal to me !

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what I really wanted to say is “more than 16 bit is mandatory”

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Agreed. Having grown up in the era where samplers went from 12 to 16 to 20 to 32 bits, the difference in harmonic distortion for each jump up to 20 is huge. 20 to 32 matters less than 16 to 20 but there’s still a measurable difference, it’s just not so audible. 16 to 20 you can easily audibly tell the difference in harmonic distortion.

All the best samplers were 12 bit for a while and they were thoroughly obsoleted except for use in stuff like industrial, where you still see them living on :rofl:

Classic example:

These things were pretty thick on the ground. So many bands had them, or Ensoniq Mirages. Still in use today by some bands.

I would recommend 20 bits minimum.

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