Multirate Signal Processing PDF
Multirate Signal Processing PDF
Multirate Signal Processing PDF
Patrick A. Naylor
Contents
Applications of multirate signal processing Fundamentals decimation interpolation Resampling by rational fractions Multirate identities Polyphase representations Maximally decimated lter banks aliasing amplitude and phase distortion perfect reconstruction conditions
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.
Requires high order analogue lter such as elliptic lters that have very nonlinear phase characteristics hard to design, expensive and bad for audio quality.
Consider oversampling the signal at, say, 64 times the Nyquist rate but with lower precision. Then use multirate techniques to convert sample rate back to 44.1 kHz with full precision. New (over-sampled) sampling rate is 44.1 64 kHz. Requires simple antialiasing lter |H (j )| = 0 dB, f < 20 kHz |H (j )| < 96 dB, f (44.1 64) 44.1 kHz 2
Could be implemented by simple lter (eg. RC network) Recover desired sampling rate by downsampling process.
Overall System
This is a simplied version In these lectures we will study blocks like G(z ) and 64
Digital Signal Processing p.9/25
16 bits per sample, 10 kHz sampling frequency gives 160 kbits/s Divide into 2 bands: high frequency and low frequency subbands. High frequencies of speech are less important to intelligibility. Therefore use only 8 bits per sample The sampling frequency can be reduced by a factor of 2 since bandwidth is halved, still satisfying Nyquist criterion. 5 16 + 5 8 = 120 kbits/s 4:3 compression Reconstructed signal has no noticeable reduction is signal quality.
M-fold Decimator
For an input sequence x(n), select only the samples which occur at integer multiples of M . The other samples are thrown away.
yD (n) = x(M n) Aliasing will occur in yD (n) unless x(n) is sufciently bandlimited loss of information.
Eg. M = 2
L-fold Expander
For an input sequence x(n), insert L 1 zeros between each sample.
yE (n) = x(M n) x(n) can always be recovered from yE (n) no loss of information, no aliasing.
Eg. L = 2
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 = ej giving YE (ej ) = X (ejL ) YE is a compressed version of X Multiple images of X (ej ) are created in YE (ej ) between = 0 and = 2
Digital Signal Processing p.17/25
To use the expander for interpolation, a lowpass lter is applied after the expander to remove the images (shaded).
Digital Signal Processing p.18/25
YD (z ) =
n=
yD (n)z n =
n=
x(M n)z n
x(n) 0
YD (z ) =
n=
x1 (M n)z n =
k=
x1 (k )z k/M
Therefore 1 1/M YD (z ) = X1 (z )= M
M 1 k X z 1/M WM k=0
X ej (2k)/M
k=0
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
Then 1 X 1 (z ) = M 1 = M 1 = M
M 1 kn n x(n)WM z k=0 n= M 1
x(n)
k=0 n= M 1 k X zWM k=0
n k WM z
What does YD (z ) =
1 M
M 1 k=0
k X z 1/M WM represent?
stretching of X (ej ) to X (ej/M ) creating M 1 copies of the stretched versions shifting each copy by successive multiples of 2 and superimposing (adding) all the shifted copies dividing the result by M
Summary
Downsampling
Upsampling