Ideas? Detecting a narrow frequency within an audio signal

Can you describe the application a little more - to offer good advice I need to know how many different tones you want to detect, how fast, how many others present at the same time etc. Is it to be low cost for one or low cost to make lots ?

MK

What about hacking a electronic tuner?

My father used a harmonic harmonizer, but, he used it for his vocals. the only problem was back than you had to manually select the key by pressing a button on the harmonizer.

Not easily done wile playing a guitar and singing.

So, i used a Electronic tuner to detect the keys, and than, taped the led's on the tuner, so it would change the keys on the harmonizer.

If nothing else, mabe you could pull up a few schematics, and see how the tuner is able to detect the notes being played, and just use that part of the circuit.

Here is one using a MPS430 to detect the notes. but i dont know the programing they would use.

Just a thought.

Hi Gordon,

Why not use a microcontroller with an ADC for this? On this website⇒ http://www.earlevel.com/main/2012/11/26/biquad-c-source-code/ , and this nice widget Biquad calculator v2 | EarLevel Engineering you can find C code for a simple biquad filter, and you can also make a tunable notch with that. If you're using an arm core you can even use the CMSIS-DSP blocks for biquads.

This solution will be quite cheap, and very tunable.

Hi,

      This can be even accomplished using notch filter or twin notch filter for higher precision of selective frequency . You may use this in pre stages before feeding the original audio signal to the microcontroller application circuit. If we can filter out 60Hz frequency using notch, then 440hz is not a big deal. use op amp and other passive components. You may refer to following links.

http://www.youtube.com/watch?v=e67OmQbR3TI

Op Amp Notch Filter | Operational Amplifier Circuit | Radio-Electronics.Com

A notch filter can be even designed in MATLAB which is frequently getting signal from audio jack of PC/sound card. you would be having better visualization and interpretations of the desirable frequency marks and get good results for your understanding

You mayuse other complex software/ algorithms and get the desired signal from the PC audio jack and feed it to the microcontroller.

do reply if its helpful to you

Thanks,

Akshay

It is also possible to perform this in software

Google Software low pass, notch or high pass filters

Here is a C example of a filter

http://stackoverflow.com/questions/13882038/implementing-simple-high-and-low-pass-filters-in-c

regards

Peter

Hi Peter,

See my answer above; he'll need a notch, and biquads are very nice and generic for this approach

Hi vslluiter,

Thank-you for pointing to biquad filters and for providing the good resourse links. This seems like a very good place for me to start. Because I am already familiar with analog circuitry, I was thinking in those terms, but I see that the microprocessor-based approach offers many benefits to the application, first of which is simpler electronic fabrication. I am new to DSP so I will have a bit of a learning curve ahead of me!

Thanks again, super helpful.

cheers,

Gordon

Play with the link that V sent by all means (its a good start) but you may have some problems because the example code there is done using double length floats which will be somewhere between slow and dead slow on a low cost micro. (If you can stretch to an ARM Cortex M4 then you'll be OK. ARM have a free DSP library you can use.)

If the basic idea seems to be working for you then the next step is to implement the filters in fixed point if you can't afford an M4 - it's perfectly doable but harder to find example code.

And of course if you only need one filter running with a reasonable sample rate you may get away with doubles even on an M3 clocked at a meager 24Mhz.

If you do need to use fixed point then let us know and I'll give you an example.

MK

Hi Michael,

Thanks for your questions. The application is an investigation into using audio tones for communication over open air – say, across a large room. There will be people and ambient sounds that introduce noise into the communication channel, but for my purposes reliability is not a critical requirement. A 10% failure rate would be acceptable.

• How many different notes?

As a starting point I plan to use the tones from the chromatic musical scale, probably an octave on either side of A-440. So, maybe 25 different notes. Notes on the chromatic scale are spaced fairly closely in frequency... here, for example, are the five notes on around of A-440:

... 391.9954 hz, 415.3047 hz, 440.0000 hz, 466.1638 hz, 493.8833 hz, ...

• how many others present at the same time ?

Up to five notes/tones may be present when trying to detect the target note, and they may be notes adjacent on the musical scale.

• How fast?

Not fast, at least, not in electronic terms. The note duration is at least 100 ms long, so the rate is 10 hz or slower.

• Is it to be low cost for one or low cost to make lots ?

At this experimental stage there will be 25 to 100 circuit boards — so, I need to minimize cost for a small multiples.

cheers,

Gordon

Hi Lupe,

Harmonizers and musical instrument tuners would have similar problems to solve, so I will look into how those operate. Thanks for your idea!

cheers,

Gordon