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(1)
Where the notation * denotes convolution sum. Table 1 lists the product and summation values for the first five values of n in (1).
TABLE I
Hierarchical blocks h, and hs, are for plotting the signals at different stages of the Flip and Slip process, with the impulse response signal flipped in time before multiplication and summation. A non-causal 750 us (6 delays = 6x125us) reference point is shown in Figure 2, and is achieved by multiplying the input signal by exp(6sT ) . The impulse response signal is also delayed by exp(4sT ) , which is two delays less to accommodate the signal length. The blocks contain Laplace analogue behavioral models (ABM) parts to generate the signals defined in (2) and (3). The delayed input signal, expressed as a function of the z-transform, is: X ( z ) = (1 + 0.5 z 1 + 0.25 z 2 ) z 6 (2) Similarly, the impulse response signal is: H ( z ) = (1 + 2 z 1 + 3z 2 ) z 4 (3) Flipping the impulse response is achieved by assigning the filter coefficients to the impulse response values, but in a reverse, or flipped, order using the useful Param part. This example helps students understand the difficult concept of convolving signals and the flipping and slipping technique. Note: The student may use another schematic to multiply the two convolution signals and compare the result to the convolution output, since convolution in time is equivalent to multiplication in the z-domain.
Session R4B
f0 =
The filter amplitude response is obtained by applying a sine wave and sweeping the input frequency. Alternatively, a narrow impulse applied yields the impulse response from which the amplitude can be obtained. A 10 us impulse signal will produce an infinite number of terms in the output, of which the first four terms are: h(0) = (0) + 1.2h(0 1) 0.91h(0 2) = 1 + 0 0 = 1 h(1) = (1) + 1.2h(1 1) 0.91h(1 2) = 0 + 1.2 x1 0 = 1.2 h(2) = (2) + 1.2h(n 1) 0.91h(n 2) = 0 + 1.2 x1.2 0.91 = 0.53 h(3) = (3) + 1.2h(n 1) 0.91h(n 2)...... (6) We may compare theses four impulse response values to the values displayed in the impulse response in Figure 4. The cursor shows time/amplitude pairs, e.g. the second pulse at 130 us, has an amplitude of 1.2 V.
Session R4B
From the impulse response output plot, select the Fast Fourier transform (FFT) icon to change the impulse response to a sinc-type display of amplitude versus frequency. Magnifying the initial portion of this display will show a repeating bandpass amplitude response (the repeating response, at multiples of the sampling frequency, is characteristic of sampled signals). However, to get meaningful measurements, such as the maximum gain, Q-factor, and bandwidth, requires carrying out an ac sweep by changing the input signal to a sine signal (rename the bubble), in order to plot the frequency response shown in Figure 5. The effect on the frequency response of varying the first filter coefficient is also shown. Changes in the Q-factor and resonant frequency, is evident when b1 is varied from 0.95 to 0.99 in steps of 0.01. As b1 approaches unity, the Q-factor becomes higher and the bandwidth narrows.
Session R4B
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FIGURE 6 QUADRATURE SIGNAL PRODUCTION
Session R4B
Figure 9 shows the 1 V input impulse signal and the impulse response pulses whose amplitudes corresponding to the filter coefficients. Figure 10 shows the amplitude response with ripple in the passband region that can be reduced by
increasing the filter order. The phase response is plotted using a pair of phase markers showing the required constant 90 degree phase shift over the desired frequency band.
Session R4B
SINGLE SIDEBAND MODULATOR
The schematic in Figure 11 is a single sideband modulator (SSB), for suppressing the upper sideband and carrier. This increases the overall transmission efficiency and the signal to noise ratio compared to a double sideband suppressed/full carrier modulator (DSBSC/FC). The Hilbert Transform hierarchical block, HilbertTransform, shifts the two sinusoidal modulating test signals by 90 degrees before being applied to each balanced modulator. The sine and cosine discrete local oscillators, examined previously, are located in the QuadOsc block and produce quadrature DSBSC outputs. Subtracting the two outputs, produce an SSB signal whose output spectra in Figure 12, shows how the carrier and upper sideband have been eliminated from the output spectrum.
Figure 11
SSB MODULATOR
CONCLUSIONS
Understanding the fundamental concepts of DSP (indeed any engineering subject), is important in order for the student to progress satisfactorily to more advanced treatment of the subject. An ever-increasing worrying trend throughout engineering education is a gradual diminution of fundamental principles, in order to accommodate the vast amounts of software/protocol techniques being created. With PSpice, the student is forced to construct the schematic from basic principles. I have found over the last six years that this method produces a better understanding when compared to the Matlab approach. However, Matlab excels for more advanced DSP analysis. PSpice has also been used to examine decimation, interpolation, digital receivers, and filter design methods such as the bilinear transform, impulse invariant, and window technique. Echo/reverb [4] and other musical effects etc. have also been investigated successfully using a segment of speech (in ASCII format) imported into PSpice. A shareware program, Wav2Ascii [3], enables a twosecond segment of speech to be recorded and inputted into PSpice. This program also allows the user to listen to the simulated output thus creating an air of realism into the simulation. Other topics not covered, because of space limitation, are artificial neural networks (ANN). Here again, simple circuits were created from basic concepts and were useful in introducing students to ANN network learning such as the delta and perceptron learning paradigms.
REFERENCES
[1] [2] [3] [4] Tobin, Paul A, PSpice for Telecommunications and DSP with over 500 worked examples (To be published in 2006). Bateman, Andrew and Paterson-Stephens, Iain. The DSP handbook, Prentice Hall PTR; CD-Rom edition (October 16, 2002) Tobin Lee E Shareware software copyright Mitra K Sanjit, Digital Signal Processing. A computer-based Approach WCB/McGraw-Hill, 1st edition (1998)