Search:

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?

The way to protect any CMOS chip from overvoltage inputs is the standard resistor + 2 schottky clamps to the power/ground rails.
However the audio shield doesn't include this on the line inputs -...

The documentation for the arm primitives notes a restriction on scaling to prevent overflow - this might be why the 200 limit is there?

If you use an array of uint8_t for the top 8 bits, and pack the remaining bits into a subsiduary array you allow
faster operations on the 8 MSBs of the data, which might sometimes be useful.
...

You'd have to edit filter_fir.h to change the #define FIR_MAX_COEFFS to be more than 200.

Note the number of taps needs to be even and 4 or more to work with the ARM fir primitive used in...

Go to the Audio design tool and drag one of the sgtl5000's into the frame, you get all the docs displayed.
(It shows AUDIO_INPUT_LINEIN as the value to use)

You should be using a library that handles speed-ramping (acceleration) for steppers - this is essential to smooth operation
of a stepper. The classic library is AccelStepper but there's the...

Hmm, that's odd - perhaps this PCB has some hardware issue?

A bleeder resistor to ground on the outputs will ensure the output capacitors get charged up properly and avoid the loud crack/thump
on plugging them into some external amplifier. Something like...

You should be calling codec.lineOutLevel (). the volume method is for headphones only I'm pretty sure.

Both the low-pass and band-pass work perfectly for me on T4.0 + audio adapter. You are using the left channel for the low pass?

Shouldn't make any difference

I'd suggest using sampling and auto-correlation if you can't rely on well-timed edges since auto-correlation is powerful at extracting
signal from noise and doesn't care about phase distortion.

DXF file format is typically used for 2D CAD, such as laser cutters and the like, so that's a candidate.
The open source InkScape tool does SVG and can export to DXF and various other file formats....

If use use Eagle then Sparkfun have a useful gerber-generator CAM script for it, https://github.com/sparkfun/SparkFun_Eagle_Settings/tree/master/cam

Some fab houses will accept Eagle .brd files...

What bandwidth do you actually need for this - the nRF24L01 only supports 250k/1M/2Mbaud, so its sensitivity is correspondingly poor.

Sensitivity increases with narrower bandwidth, so if you only...

The speaker driver has to match the speaker enclosure, as they are both resonant systems and couple to each other.

The surroundings are another thing entirely - typical rooms trap lower...

My approach to this problem is to use photo diode in reverse-biased configuration (the fastest), and use a modified transimpedance
amp configuration using multiple pn-diodes to produce a...

There's a rule of thumb for speakers - you can't have all three of these qualities, but at most two of them:

1) small speaker enclosure
2) high power efficiency
3) good bass entension

So the...

The standard biquad will do upto 8 poles of arbitrary filtering, don't think there's anything to stop you creating an unstable filter using it, but
offhand I've no idea if it behaves well when...

240mAh is a capacity rating, not a current rating. To figure out if the battery can source enough current
you need to know the current rating.

For many batteries you'll see a C value, indicating...

Here's one approach:

First a filter layer to generate extra noteOff and noteOn messages/calls to ensure the live note count is <= 5.

Secondly a standard mapping from notes to oscillators which...

no idea where you got that from. FFTs always generate entire spectra.



What frequency resolution do you want? The audio lib 1024-point FFT at 44100 samples/second has a bin size of 43Hz...

That 8 pole bandpass will be very peaky as all the poles are coincident, leaving gaps between the
centre frequencies. It might be an idea to derive standard Butterworth poles if you want a...

Is this simply a consequence of 16 bit intermediate results in the freeverb object? 16 bits is enough for a signal, but not usually
sufficient for complex processing of signals as...

The FFTs in the audio lib are 256 and 1024 point, locked to the sample rate.

One FFT can do the full frequency analysis, then you would apply different frequency windows on
the results to sum...

General use, NE5532, great all-round performance, very cheap, although its not the best for a low impedance source. For
low voltage rail-to-rail the situation changes all the time, but anything...

That's noisy for a studio mic, 10--15dB is more reasonable. At 1m distance you are in studio conditions, and not being a mobile phone mic - there's
perhaps 20 to 30dB path-loss from moving the mic...

The voltage levels for the chip's I2C bus look to be for 5V logic, not necessarily reliable at 3.3V, note - this
must be an old design...

Don't forget the input bleeder resistors. I'd add RF...