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Problems                                                             185

        4.23 A digital signal can be characterized by the sum of two sinusoids:





            where f\ = 0.8 kHz, /2 = 1.6 kHz, and fsample - 4 kHz. The sample rate is
             decreased to 2 kHz by first applying a digital lowpass filter and then
             discarding every other output sample.
            (a) Determine first the digital spectrum if the lowpass filter is not used.
            (b) Determine the digital spectrum if the lowpass filter meets the following
                specification: A max = 0.1 dB, (O CT = n/4,A mi n = 40 dB, and Q) ST = 0.8 n.
        4.24 The sample rate of the digital signal in Problem 4.23 is increased to 8 kHz by
            inserting zeros and removing the unwanted images using a lowpass filter.
            (a) Determine first the digital spectrum after zeros have been introduced.
            (b) Determine proper cutoff and stopband angles for the lowpass filter.
            (c) Determine the digital spectrum if the lowpass filter meets the following
                specification A max = 0.1 dB andA mj ra = 40 dB.
            (d) Determine the required order if a Cauer filter is used as the interpolator
                filter.
        4.25 (a) Determine the specification for an ideal interpolator that increases the
                sampling frequency by a factor two for a bandlimited signal contained
                within the band 0 to a^T < n.
            (b) Determine the frequency response of the ideal interpolator filter.
            (c) Determine the impulse response of the ideal interpolator filter.

        4.26 Show that it is not efficient to interpolate the input sample rate by a factor
            two by repeating the input samples instead of introducing zeros.
        4.27 Suggest a system for interpolation of the sample frequency by a factor 1.75.
            Determine also the cutoff and stopband edges of the filters involved.
        4.28 The following scheme is used in a simple approach to double the sample
            frequency. The original samples are retained and new samples are created
            between the original samples by linear interpolation, i.e.,




            (a) Determine the difference equation for the interpolator. Hint: Create a
                new sequence from the original by interleaving zero samples between
                the original samples.
            (b) Determine the transfer function for the interpolator.
            (c) Determine the poles and zeros.
            (d) Sketch the frequency response.
            (e) Sketch the Fourier transform for the original, input, and output signals
                to the interpolator.
            (f) Determine the impulse response of the interpolator.
        4.29 Determine the work load required for the interpolator (case study) if the
            allpass filter is placed after the interpolator filter.
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