Ultrasound recording and Detectors using Digital Signal Processing

Detectors using DSP work by amplifying the signal from the microphone, then sampling it at a very high frequency, and processing it as digital data.

How DSP detection works:

The signal from the microphone is amplified then "sampled" at very high rates often as many as 380 000 samples each second. Measurements of the signal are made at regular intervals.  Here the blue wave represents the amplified signal from the microphone.  A sampling wave is produced ( the green line) with a value that can be either 0 or 1. When the two are multiplied together a series of samples (indicated by the red wave) is produced.
Here you can see the circuit that does this conversion.  The signal from the microphone is amplified and sampled at regular intervals controlled by the waveform from the clock signal generator. The sampled signal (red wave above) is then digitised (converted to numbers) by an Analog to Digital Converter (ADC).  As long as the sample rate is fast enough the result will be a good representation of the signal from the microphone.
The result is a "data stream" a set of data values as shown here.  These would then be stored in memory.	252, 94, 28, 241, 130, 10, 220, 165, 1, 193
Sound analysis packages like Soundruler use sound that is recorded on a computer as a .wav file - this uses the same process as above, the sound being fed into a sound card which converts the signal to a data stream similar to the above.	For best results from sound analysis this data stream should be recorded directly onto the analysis computer in digital form. Converting back to sound then digitising again introduces noise and distortion.
The digitized signal can be restored by converting the numbers to voltages using a Digital-to-Analog Converter (DAC), and played back at any desired speed; for example ten times slower.  This gives us our time expansion detector. Of course we are not limited to just playing the signal back.  One option is to measure the frequency of the incoming signal via a Fast Fourier Transform (FFT) and use this to generate a signal at a corresponding but lower frequency that humans can hear. This technique is used in hearing aids and is called "frequency compression"

Many other opportunities exist for using the digital signal to produce a sound that represents the bat call in a way that is useful and informative for humans.