I'm working on a project which requires that we identify some musical notes played, where the music to be played is single-tone.
it's basically by using filters.
The way I think of it, the signal from the microphone will go through a low pass filter and then that signal will pass through 12 different band pass filters (we have 12 notes) in order to identify which note was played...... (by comparing the frequency of the input signal to the ranges of the pass band filters).

So I understand how a filter works, for example a low pass filter is ideally supposed to make the voltage after a certain frequency quite low (in a sense blocking it.......)
However how low does a voltage have to be so that we may neglect it, or consider the signal coming to be "blocked".
For example in the attached circuit below, our cutoff frequency is 1000 rad/s or 159Hz.... at 159Hz you'd expect the voltage of the capacitor to be 0.707 * Vin which equals 7V...... as we increase the voltage further the output voltage should be significantly small thus blocking signals with a higher frequency than the cuttoff frequency......... but it doesn't work like that....

Furthermore, what if a signal with a frequency higher than the cutoff frequency, generates an output voltage of 6V......... that's still a high voltage.... and in no proper manner can be considered a "blocked signal"

So how do filters work then???

 

Hi,

Filters work by blocking some frequencies and allowing others to pass. However, the response is also due to signal amplitude because the output/input ratio is based not only on frequency but on amplitude as well.

So for example, if you have a 2v input at 1kHz and your filter passes that (2v output) but cuts a 2v 2khz signal down to just 1v, then if you boost the amplitude of the 2k input to 4v you will then get 2v out for the 2kHz signal. This means you get 2v out for either 2v 1kHz or 4v 2kHz, so you see you cant detect either one being greater or lower than the other because if the amplitude changes they both produce the same output.

However, if you can limit the input to a constant amplitude like say 2v, then the output for 1kHz will be 2v but for the 2kHz signal it will be only 1v, regardless what the real input was because the original input is limited to 2v.

Of course there is also the chance that the signal will be lower than 2v, so you'd have to boost that. The complete input circuit then would have to look like an automatic gain control circuit.

There are other ways of doing this, but you'd have to go to a different kind of circuit like a frequency to voltage circuit, which i am not sure you are allowed to do here.

 

Filters do not actually block signals outside the pass band, they attenuate them (reduce their relative amplitude).
The further you get from the passband, the more the attenuation.
For your application you need a number of bandpass filters.
Here's a short tutorial on those.
For better discrimination between frequencies you likely will need to go to an active bandpass filter such as discussed here.

 

To reliably detect 12 different frequencies over the audio range you may need filters with a fairly high "Q" factor (see above link).
It all depends what the audio range you will be using. The standard range of say a piano keyboard - or the whole audio range (50Hz to 17kHz)?

These days it's often easier to do this using digital techniques - but that's a whole different ball game!.

 

I don't know if it still is available, but there used to be a small phased locked loop (PLL) part made by Signetics (then Philips, then NXP, then ..._called the NE567. This was designed specifically to be a tone decoder. As the datasheet shows, a group of them could be piled up to detect touchtones.

https://www.sonoma.edu/users/m/marivani/datasheets/misc/NE567.pdf

ak

 

Updated version still available. Now called the LM567. From Farnell UK.