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P. 1

DSP5|Views: 0|Likes: 0

Published by Brijesh Singh

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https://www.scribd.com/doc/100048444/DSP5

07/14/2012

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Patrick A. Naylor

Digital Signal Processing – p.1/25

Contents

Applications of multirate signal processing Fundamentals decimation interpolation Resampling by rational fractions Multirate identities Polyphase representations Maximally decimated ﬁlter banks aliasing amplitude and phase distortion perfect reconstruction conditions

Digital Signal Processing – p.2/25

Introduction

In single-rate DSP systems, all data is sampled at the same rate no change of rate within the system. In multirate DSP systems, sample rates are changed (or are different) within the system Multirate can offer several advantages reduced computational complexity reduced transmission data rate.

Digital Signal Processing – p.3/25

**Example: Audio sample rate conversion
**

recording studios use 192 kHz CD uses 44.1 kHz wideband speech coding using 16 kHz master from studio must be rate-converted by a factor 44.1 192

Digital Signal Processing – p.4/25

5/25 .1 kHz 2 Requires high order analogue ﬁlter such as elliptic ﬁlters that have very nonlinear phase characteristics hard to design. Bandwidth for audio is 20 Hz < f < 20 kHz Antialiasing ﬁlter required has very demanding speciﬁcation |H(jω)| = 0 dB. f ≥ 44. Digital Signal Processing – p. f < 20 kHz |H(jω)| < 96 dB. expensive and bad for audio quality.Example: Oversampling ADC Consider a Nyquist rate ADC in which the signal is sampled at the desired precision and at a rate such that Nyquist’s sampling criterion is just satisﬁed.

Digital Signal Processing – p.6/25 .Nyquist Rate Conversion Anti-aliasing Filter.

Digital Signal Processing – p. New (over-sampled) sampling rate is 44.1 × 64 kHz.Consider oversampling the signal at. f ≥ (44. say.7/25 .1 kHz with full precision.1 kHz 2 Could be implemented by simple ﬁlter (eg. Then use multirate techniques to convert sample rate back to 44. Requires simple antialiasing ﬁlter |H(jω)| = 0 dB. f < 20 kHz |H(jω)| < 96 dB.1 × 64) − 44. RC network) Recover desired sampling rate by downsampling process. 64 times the Nyquist rate but with lower precision.

8/25 .Oversampled Conversion Antialiasing Filter Digital Signal Processing – p.

Overall System This is a simpliﬁed version In these lectures we will study blocks like G(z) and ↓ 64 Digital Signal Processing – p.9/25 .

certain frequency bands are less important perceptually because they contain less signiﬁcant information bands with less information or lower perceptual importance may be quantized with lower precision . This gives an adequate dynamic range. Digital Signal Processing – p.fewer bits. How many bits are required? In general. Divide the spectrum of the signal into several subbands then allocate bits to each band appropriately. 16 bits precision per sample is normally used for audio. In practice.Example: Subband Coding Consider quantizing the samples of a speech signal.10/25 .

5 × 16 + 5 × 8 = 120 kbits/s 4:3 compression Reconstructed signal has no noticeable reduction is signal quality. 10 kHz sampling frequency gives 160 kbits/s Divide into 2 bands: high frequency and low frequency subbands. still satisfying Nyquist criterion.16 bits per sample. Digital Signal Processing – p. Therefore use only 8 bits per sample The sampling frequency can be reduced by a factor of 2 since bandwidth is halved.11/25 . High frequencies of speech are less important to intelligibility.

12/25 .Fundamental Multirate Operations Downsampling by a factor M ﬁlter and M-fold decimator Upsampling by a factor L L-fold expander and ﬁlter Digital Signal Processing – p.

13/25 . yD (n) = x(M n) Aliasing will occur in yD (n) unless x(n) is sufﬁciently bandlimited loss of information.M-fold Decimator For an input sequence x(n). Digital Signal Processing – p. select only the samples which occur at integer multiples of M . The other samples are thrown away.

M = 2 Digital Signal Processing – p.Eg.14/25 .

no aliasing. insert L − 1 zeros between each sample.L-fold Expander For an input sequence x(n).15/25 . Digital Signal Processing – p. yE (n) = x(M n) x(n) can always be recovered from yE (n) no loss of information.

16/25 . L = 2 Digital Signal Processing – p.Eg.

17/25 .Frequency Domain View of the Expander From the deﬁnition of the z-transform ∞ YE (z) = n=−∞ ∞ yE (n)z −n yE (kL)z −kL k=−∞ ∞ = = k=−∞ x(k)z −kL = X(z L ) For frequency response write z = ejω giving YE (ejω ) = X(ejωL ) YE is a compressed version of X Multiple images of X(ejω ) are created in YE (ejω ) between ω = 0 and ω = 2π Digital Signal Processing – p.

To use the expander for interpolation. Digital Signal Processing – p.18/25 . a lowpass ﬁlter is applied after the expander to remove the images (shaded).

Frequency Domain View of the Decimator From the deﬁnition of the z-transform ∞ ∞ YD (z) = n=−∞ yD (n)z −n = n=−∞ x(M n)z −n Let x1 (n) = Then ∞ ∞ x(n) 0 if n is an integer multiple of M otherwise YD (z) = n=−∞ x1 (M n)z −n = k=−∞ x1 (k)z −k/M since x1 (k) = 0 unless k is a multiple of M . Digital Signal Processing – p.19/25 .

20/25 .Therefore 1 1/M YD (z) = X1 (z )= M M −1 k X z 1/M WM k=0 as will be shown on the next slide and using k WM = e−j2πk/M For frequency response write z = ejω to give 1 YD (ejω ) = M M −1 X ej(ω−2πk)/M k=0 Digital Signal Processing – p.

We arrive at the previous expression for YD (z) by considering a new sequence cM (n) = and then writing x1 (n) = cM (n)x(n) Further consideration of cM (n) tells us that cM (n) is the inverse Fourier transform of unity and can be written 1 cM (n) = M M −1 −kn WM k=0 1 if n is an integer multiple of M 0 otherwise Digital Signal Processing – p.21/25 .

22/25 .Then 1 X1 (z) = M 1 = M 1 = M M −1 ∞ −kn x(n)WM z −n k=0 n=−∞ M −1 ∞ x(n) k=0 n=−∞ M −1 k X zWM k=0 −n k WM z from the deﬁnition of the z-transform. So ﬁnally 1 YD (z) = M M −1 k X z 1/M WM k=0 Digital Signal Processing – p.

23/25 .What does YD (z) = 1 M M −1 k=0 k X z 1/M WM represent? stretching of X(ejω ) to X(ejω/M ) creating M − 1 copies of the stretched versions shifting each copy by successive multiples of 2π and superimposing (adding) all the shifted copies dividing the result by M Digital Signal Processing – p.

Digital Signal Processing – p.To use a decimation process we must ﬁrst bandlimit the signal to π |ω| < M .24/25 .

Summary Downsampling Upsampling Digital Signal Processing – p.25/25 .

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