Its saying the control voltage can be 3V to 10VDC. What can that mean other than input 4 can be upto 10V more positive than input 3?
If that's the case, you don't have to worry.
Type: Posts; User: MarkT
Its saying the control voltage can be 3V to 10VDC. What can that mean other than input 4 can be upto 10V more positive than input 3?
If that's the case, you don't have to worry.
BTW what signal quality requirements do you have?
"Control voltage range 3V to 10V" - that's pretty clear to me.
Yes I think so, the risetime of the signal is not instantaneous, and the thresholding will likely be different for each microphone
due to signal level variation. Cross-correlation uses all the...
This is the correct link, https://www.ap.com/blog/fft-spectrum-and-spectral-densities-same-data-different-scaling/
The basic insight is that a discrete spectrum is measured differently from a...
3V is possibly not safe on low voltage CMOS chips. Many standard multimeters use 0.2V full-scale except on diode mode.
The 56k you see on the powered down chip is because you are back-driving the...
I've not played with the T3.6, but I wouldn't be surprized if you can generate suitable square waves from the PWM units in it?
This thread seems relevant. ...
I think you mean correlation, not convolution (very different things), and secondly its cross-correlation,
an autocorrelation cannot be between two signals!
Let me say it again, FFT plots of signal spectrum are not useful for measuring noise floor, you need a PSD plot for that.
Noise is measured in W/Hz, or V/√Hz, and an FFT bin is typically not exactly...
Audio is an ac signal, typically I2S audio ADCs have a DC-removal filter built in so the result is strictly zero-centred.
Oddly typical audio DACs are not DC-blocking, which is a problem (if the...
Hmm, perhaps that level shifter isn't able to drive the opto-coupler LEDs properly - can't see any
datasheet for it. Opto coupled inputs probably want 15mA or so, which is more than some logic...
Sounds like a job for a DDS module, not a microcontroller. Cheap AD9833 module perhaps?
Or you could use square wave output and a high-order low-pass filter on the output to kill the harmonics.
Sounds like a job for a DDS module, not a microcontroller. Cheap AD9833 module perhaps?
That's not a small motor. This is the small type of stepper: https://www.hackster.io/news/drive-a-tiny-stepper-directly-with-arduino-911571f8cd0e
Try with AccelStepper library and 1000...
SDRAM needs about 26 pins to connect to it, several control lines and an 8-bit databus and 12 or so address bits.
You mention an existing diode, but don't explain...
For a strain-guage wheatstone bridge configuration you need an instrumentation amp to boost the differential mode signal.
There are many instrumentation amps you can get, its basically 3 opamps in a...
If you want to integrate pulses, use an integrator before sampling. Then you need a much lower sample rate.
The pulse time can be captured with a comparator, the pulse energy by differencing the...
This is the classic scheduling problem, and a good data-structure to use is a heapsorted list, aka binary heap.
inserts and extracts are O(log(N)) so it scales well to 100's or 1000's of entries,...
I guess I was saying either use a standard Bressenham's or write something cleaner without the arbitrary mix of global variables and
passed arguments. For me a nice clean interface would be a...
Indeed you'll never get a stepper to jump to high speed from standstill without speed ramping, unless its a really tiny motor.
Most 'scope front-ends aren't linear enough for any kind of audio quality measurement work! They are designed for a large bandwidth with constant group-delay,
and are typically 8-bit so only have...
10MSPS ADC needs a separate (presumably parallel) ADC chip and you need a datapath for it too, this is high speed acquisition.
FPGA's are often used once you get to these kinds of speeds as the...
All pn junctions are photodiodes and LEDs, if the material is transparent at the relevant wavelength, since photons generate
electron-hole pairs and recombination can emit a photon (although the...
I think you'll need to do some low-pass filtering then. Try an RCRC to start with, that might have enough roll-off and
is very simple. For differential RC filter you have two resistors and one...
The logic doesn't look good in linearInterpolation... First you are not dealing with any edge cases, such as when X_old == pTS.x.
Second you are overwriting wave1Values[Xold] the first time round...
The default volume is 0.3 and 4 x 0.3 > 1.0 so your mixer output is clipping I think. Try 0.25.
Ah, but I think your oscillators are at 0.3 already so that won't be it....
Ah, looked again,...
I'd say should use, not must use :)
For low voltage circuitry many old style opamps are bad news as they require dual rail supplies - a modern
true rail-to-rail audio spec'd opamp is a better...
Absolutely not, you'll be driving them into slew-rating limiting which is massively distorting (the feedback loop
completely loses lock). Also the step change in input may destroy the front-end of...
That 33uF on the output of the buffer should definitely not be there! Opamps don't drive capacitive loads!
Inductance is measured in henries, not farads... 33µH perhaps?
If you drive audio amp from a switch-mode supply (not recommended), then make sure its one with an
ultrasonic switching frequency...
Ah, xyproblem again.
You can't process the output of a class D amplifier in the analog domain as its a digital signal.
First you need to low-pass filter the signal.
The MCP6002 won't do audio, far too slow(*). Its an ultra low power opamp, so the bandwidth and slew-rate are woeful.
Your virtual ground reference network, the two 47k resistors, lacks...
Is this a layout issue? Cables too long? Missing or misrouted ground return wire?
If you want to find out more, this is the paper that inspired me: https://holometer.fnal.gov/GH_FFT.pdf
The PCM1808 datasheet clearly shows the analog supply is 5V and digital 3.3V. If a module doesn't have a separate regulator
on it then both voltages need to be brought into the module.
Its...
No, you want to reduce the sampling rate or increase the number of FFT bins or both.
The FFT output is always equally spaced frequency bins, nothing you can do to change that, you simply want
to present that information recscaled logarithmically. For each x position map to an...
Then you'll need to map your x coord to frequency through an exponential function, and use that to decide which FFT bin corresponds
to that frequency. And you'll probably want to interpolate...
I'm just trying to correct the common misconception about noise-floor measurements using the FFT - its very
rarely appreciated that noise isn't measured the same as a single tone. Noise is measured...
Noise floor is measured as power spectral density, that plot is showing the power spectrum w.r.t. the loudest tone, so the
-78dB isn't meaningful as a measure of noise, since changing the FFT number...
These are CMOS isolators, not opto-isolators, so standard CMOS input and output behaviour. Outputs good to 4mA from
what I can tell, fine for any logic input. The WS2812 input is just CMOS logic...
You do need sin to work reasonably on largish arguments though, a classic example being in implementing sinc().
Well if you don't need well defined phase response IIR filters are probably more straight-forward, and if you need several their coefficient
arrays are _much_ more compact.
For simple frequency...
If you followed the forum rule, we might have solved this already. One thing to watch out for is correct block handling
in the update() method, always checking for NULL blocks and releasing each...
Like switch-case?
But do you need phase information?
Ah, I wasn't expecting a +/-15V preamp and +15V phantom, +48V is standard for phantom power, and a 5V rail to rail preamp would be
more usual in a microcontroller context these days :)
Make them both a few k.
Your phantom power circuit will destroy the mike preamp, you don't have diode clamps to protect it from the 48V phantom-power switch transients.
See the diodes here for...
BTW why use FIR for simple audio LP/BP/HP filtering? Do you actually need linear-phase?