I am a recent grad and my new project at work deals with taking the signal from an electret microphone and getting SPL values for 10 standard frequency bands.
The analog amplification/conditioning is completed so I am now trying to figure out what do in terms of sampling/filtering. My frequency bands are:
- 31.5, 63, 125, 250, 500, 1000, 2000, 4000, 8000, and 16000 Hz
I decided that I should be able to use a 160kHz sampling rate (apparently 5-10 * Nyquist rate is recommended according to my old notes) and use 10 2nd order Butterworth bandpass filters that were designed using mkfilter. After this point I am unsure how to proceed.
Would I simply use the 16kHz filter every sample, the 8kHz filter every 2 samples, the 4kHz filter every 4 samples and so on? ( I am getting these numbers by using 5*Nyquist rate for each band, e.g. 1/(8000Hz*2*5)/(1/160000Hz) ) ? Then I would then convert the values to a pressure and do the math necessary for a SPL measurement. Is there a better way to approach this? Is my understanding correct?
The hardware I am working with has a 48MHz clock and 32kB of flash and 4kB of RAM.
Best Answer
No, if you feed in a signal containing frequencies to 16 kHz, your filter will need to sample at at least 2*16 kHz for all bands. If you sample only at 160 kHz, you'll have to design each filter for that sampling rate.
You could use a lower sampling rate for the lower frequency filters, while keeping the sampling rate above the highest frequency present (16 kHz ?). Thus, taking every 2nd sample and using the 16 kHz (160 kHz sampling) would work for 8 kHz -- but conflicts with the rationale for choosing 5x oversampling in the 1st place (that's not strictly necessary -- it depends on the highest frequency components in the signal).
Note that if you extended this thought to the 31.5 Hz band, you'd use every 500th sample == 160k/500 = 320 Hz, which wouldn't work with the whole signal since it contains components to (and beyond ?) 16 kHz.