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DIGITAL FILTERS MADE EASY


Circuit Cellar Online
THE MAGAZINE FOR COMPUTER APPLICATIONS
Circuit Cellar Online offers articles illustrating creative solutions
and unique applications through complete projects, practical
tutorials, and useful design techniques.

DIGITAL FILTERS MADE EASY

Lessons from the Trenches Part 2: The Rest of the Story
by Dan Cross-Cole

Start ý All the Pieces ý Using the DSP/FFT System ý Sources and PDF

USING THE DSP/FFT SYSTEM

Digital band-pass filters provide two useful functions. Not only do they allow only a designated band of frequencies to pass through, but they also block the other frequencies, as well as the noise in those frequency bands (see Photos 1ý4).

Photo 1ýThe horizontal scale is in hundreds of hertz. The display shows two signals at about 500 and 950 Hz. This was taken using the DSP filter that passes signals between 500 and 1500 Hz (click here to see Photos 2, 3, and 4)

The DSP/FFT system can be used to check the flatness of response of an audio system. A faster sampling is possible with a faster CPU. My laptop is a ý486 running at 40 MHz. With todayýs computers operating at ten times that speed, you can expect sampling rates of 80 kHz. This rate can handle both stereo channels.

The current system displays the data at a rate of about five refreshes per second. Again, a faster microprocessor could work wonders. The limiting module is the ADC. It has to sample at 40 kHz per channel. The current one (ADC0809) is designed for about 10,000 samples per second. This is fine for sampling audio up to about 5 kHz, which covers speech adequately and is surprisingly fair for music, but if you want to experiment with faster CPUs, you should use delay statements in the program to find the maximum sampling rate. Otherwise, the ADC chip hangs up.

The DSP/FFT system can measure the response from an acoustic guitar. The limited range is adequate for the lower spectrum of interest. This corresponds to the tones that distinguish rich sounding guitars from those that lack a broad bass response.

Using the DSP filter can also eliminate unwanted audio energy (see Photo 4), resulting in a more linear measurement of the frequencies of interest.

The software is available. You will need the QuickBASIC compiler or some other BASIC platform to run the program. The program takes 512 samples but only displays 90 frequency components on the screen, with about 20 Hz between components. This means that you can display frequency components near the maximum sampling frequency. Again, a faster ADC will let you use this system to display the entire audio range.

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