Pittsford, New York. Signal Wizard II™ is a unique new, real-time DSP-based filter design system for conditioning audio bandwidth signals with exceptional sharpness. Its combined hardware and software enables design engineers, more familiar with analog filter design, to achieve significantly superior results – and without any DSP expertise! A key feature of Signal Wizard II™ is that it allows designers to design and produce a high-performance filter in seconds without any DSP knowledge, or even understanding the mathematics involved behind digital filter designs.
Signal Wizard II™ uses a Freescale (Motorola) DSP to filter unwanted signals from analog input signals. Conditioned signals are then converted to analog output in real time. The filter is specified using an easy-to-use visual software interface where the user simply enters the required frequency cut-off points and downloads them to the DSP chip. The response shape can be tailored by changing the window type and number of coefficients. At the other end of the scale, the software can handle complex input functions. Each time a change is made, you’re given an up-to-date response display. Once you’re satisfied with the design, the filter is downloaded to the DSP chip and is ready for use. All you have to do then is connect the inputs and outputs to the hardware and begin using the filter – a “no-brainer”!
Signal Wizard II™ allows you to make much sharper filters than any analog counterpart. You can mimic analog filter responses with 24-bit, 48 kHz sampling accuracy. Low pass, high bass, multiple band-stop/band-pass filters may be combined to produce very complex filters for frequencies up to 24 kHz, as well as standard infinite impulse response (IIR) and adaptive types. The software can also accept measured responses to define a filter template. This can be used to provide measurement equalization or to search out signal signatures in noisy environments. In fact, it is a simple matter to produce filters with completely arbitrary frequency magnitude and phase characteristics using the finite impulse response (FIR) method, with no phase distortion, no matter how sharp the filter is – try doing that with an analog filter! Alternatively, arbitrary phase distortion can be introduced if this is desirable. It is even possible to design and execute real-time deconvolution (inverse) filters using the special invert mode. Because the processing module is so fast, it is possible to design filters with responses far beyond what is possible with traditional analog techniques. The control program runs under Windows, and provides a user-friendly filter design tool that de-mystifies the process of specifying the filter. The filter design process simply becomes one of describing the desired frequency response. The design package shows the response that will be produced and also the deviation from that specified. User designs may be stored for re-use and actual responses may be entered from measurements for simulation or equalization purposes. The filters are calculated and downloaded to the hardware within seconds. Results are shown on a real-time dual channel software scope and spectrum analyzer function. Frequency responses can be plotted as magnitude, dB, square, root, real, imaginary or phase; log or linear frequency axis; pole-zero plots and coefficient export.
There are many more applications for this device than simple filtering. It is possible, for example, to make an electric violin sound like its acoustic counterpart. Any audio-bandwidth signals in instrumentation applications can be processed, for example. Full anti-aliasing and reconstruction filters are incorporated, and there is no phase distortion - a problem that plagues analogue filters.
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