Multirate Digital Signal Processing

Contents
1) What is multirate DSP?
2) Downsampling and Decimation
3) Upsampling and Interpolation
4) FIR filters
5) IIR filters
a) Direct form filter
b) Cascaded form filter
6) Polyphase filters
7) Advantages of multirate DSP
8) Applications of multirate DSP
a) Design of phase shifts
b) Interfacing of digital systems with different sampling rates
c) Implementation of digital filter banks
d) Subband coding of speech signals
e) Quadrature mirror filters (QMFs)
f) Transmultiplexers
g) Oversampling A/D and D/A conversion

Introduction

Interest in signal processing long predates computers. As long as people have

tried to send or receive information through electronic media, such as telegraphs,
telephones, television, radar, etc., there has been the realization that these signals
may be affected by the system used to acquire, transmit, or process them.
Sometimes these systems are imperfect and introduce noise, distortion, or other
artifacts. Understanding the effects these systems have and finding ways to
correct them is the foundation of signal processing. There are many types of
signal processing. Among those Digital signal processing is more efficient and
widely used. Multirate systems are building blocks commonly used in digital
signal processing (DSP). Their function is to alter the rate of the discrete-time
signals, which is achieved by adding or deleting a portion of the signal samples.
"Multirate" simply means "multiple sampling rates". A multirate DSP system
simply uses more than one sampling rate within the system. In many systems,
multrate DSP increases processing efficiency, which reduces DSP hardware
requirements. Also, a few systems are inherently multirate, for example, a
"sampling rate converter" system that converts an input sampling rate to a
different output sampling rate. Multirate systems play a central role in many areas
of signal processing, such as filter bank theory and multiresolution theory, they
are essential in various standard signal-processing techniques such as signal
analysis, denoising, compression and so on. During the last decade, however, they
have increasingly found applications in new and emerging areas of signal
processing, as well as in digital communications.

"Multirate" means "multiple sampling rates". A multirate DSP system uses multiple
sampling rates within the system. Whenever a signal at one rate has to be used by a
system that expects a different rate, the rate has to be increased or decreased, and some
processing is required to do so. Therefore "Multirate DSP" refers to the art or science of
changing sampling rates.
"Resampling" means combining interpolation and decimation to change the sampling
rate by a rational factor. Resampling is done to interface two systems with different
sampling rates. Ex: Professional audio equipment uses a sampling rate of 48 kHz, but
consumer audio equipment uses a rate of 44.1 kHz. To transfer music from a professional
recording tape to a CD, the sampling rate must be changed by a factor of 44100 / 48000 =
441/480=147/160.Therefore we would interpolate by a factor of L=147 then decimate by
a factor of M=160.The resampling factor is 147 / 160 = 0.91875.The Nyquist criteria
must be met relative to the resulting output sampling rate to prevent aliasing. Since
resampling includes interpolation and decimation, we require an interpolation and a
decimation filter
Multirate DSP consists of:

1. Decimation: It is a process to decrease the sampling rate.

2. Interpolation: It is a process to increase the sampling rate.

"Downsampling" is a process of removing some samples, without the lowpass

filtering. A signal is downsampled only when it is "oversampled"(i.e. sampling
rate > Nyquist rate). This combined operation of filtering and downsampling is
called Decimation. To downsample by a factor of M, we must keep every Mth

sample as it is and remove the (M-1) samples in between. Ex: To decimate by 4,

keep every fourth sample, and remove three out of every four samples.

Symbol of downsampler

The graphical representation for M=4 is

Block diagram of a decimator

"Upsampling" is the process of inserting zero-valued samples between original

samples to increase the sampling rate. (This is called "zero-stuffing"). Given a
sequence x[n] , we can define

Where xu[n] is the sequence up-sampled from x[n] by a factor of L.This means
that xu[n] is generated by padding (L-1) zeros between every sample of x[n].

Symbol for up-sampler

Graphical representation for L=2

"Interpolation" is the process of upsampling followed by filtering (to remove the

undesired spectral images.) The result is a signal sampled at a higher rate. The
interpolation factor (L) is the ratio of the output rate to the input rate.

Block diagram of an interpolator

Interpolation consists of two processes:

1) Zero stuffing :Inserting (L-1) zero-valued samples between each pair of

input samples. The zero stuffing creates a higher-rate signal whose
spectrum is the same as the original over the original bandwidth, but has
images of the original spectrum centered on multiples of the original
sampling rate.

2) Lowpass-filtering: The lowpass filtering eliminates the images.

There are 3 different kinds of filters. They are:

1) FIR filter

2) IIR filter : a) Direct form

b) Cascaded form

3) Polyphase filter

1) FIR filter: A causal FIR filter has the following difference equation

Where M is the order. The result y[n] is the discrete convolution of x[n] with
the (finite) impulse response:

2) IIR filter: The input x[n] and output y[n] of a causal IIR filter satisfy the
Nth order linear constant-coefficients difference equation of the form.

Often the coefficient a0 is assumed to be 1 and we can rewrite the difference

equation as

Where k=1,2N.

a) Direct form: The system function is of the following form.

b) Cascaded form: The system function is

One advantage of cascaded form over the direct form is that a small change of a
coefficient (ex:quantization)moves only the pair of poles(or zeros)of the
corresponding stage and not all others.Furthermore,the amount of displacement is
less than for the overall higher order direct form filter.

Polyphase filtering

Polyphase filtering is a technique that allows us to reduce the computational

requirements when performing convolution followed by down sampling. For
example, consider the basic lowpass filtering followed by down-sampling
structure:

This direct implementation is extremely inefficient since the tapped delay line is
computing all the samples at its output and yet (M-1) of them are thrown away.

Advantages of Multirate DSP.

1) With interpolation and decimation, the computational and/or memory

requirements of the resampling filtering can sometimes be greatly reduced by
using multiple stages.

2) Sampling rate conversion of a digital signal can be accomplished in two methods. One
method is to pass the digital signal through a D/A converter, filter it and then resample
the resulting analog signal at the desired rate. The second method is to perform the
sampling rate conversion entirely in the digital domain. The advantage of the first
method is that the new sampling rate can be arbitrarily selected and neednt have any
special relationship to the old sampling rate.
3) Multirate Digital signal processing is more efficient, distortion less and flexible type of
signal processing.
Applications of Multirate digital signal processing
1) Used for the design of phase shifters
2) Interfacing of digital systems with different sampling rates
3) Implementation of digital filter banks
Filter banks are used for performing spectrum analysis and signal synthesis. The filter
banks are basically two types. They are Analysis filter banks and Synthesis filter
banks.

a) Synthesis filter bank

b) Analysis filter bank

4) Subband coding of speech signals.

Multirate signal processing notions provide efficient implementations of the subband
encoder. Subband coding is a method, where the speech signal is subdivided into several
frequency bands and each band is digitally encoded separately. Subband coding is also an
effective method to achieve data compression in image signal processing.
5) Quadrature mirror filters (QMF).

Two channel QMF bank.

QMfs split the input signal into two output signals with bandwidth half of the original
bandwidth. Thus the sampling rate of the output signals can be decimated by a factor of
two. The output of the QMF filter bank after being processed (encoding, decoding,
individual amplification etc) is recombined to a single signal using the synthesis filter
bank also composed of QMFs .
6) Transmultiplexers.
These are the devices used for converting between Time Division Multiplexed (TDM)
signals and Frequency Division Multiplexed (FDM) signals.
7) Oversampling A/D and D/A conversion.
An oversampling A/D converter is implemented by a cascade of an analog sigma-delta
modulator (SDM) followed by a digital anti-aliasing decimation filter and a digital

highpass filter. The analog SDM produces a one-bit per sample output at a very
highsampling rate, which is passed through a digital lowpass filter, which provides a high
precision output that is decimated to a lower sampling rate. This output is then passed to
a digital highpass filter that serves to attenuate the quantiztion noise at the lower
frequencies.

The digital signal is passed through a highpass filter whose output is fed to a digital
interpolator. This high sampling rate signal is the input to the digital SDM that provides a
high sampling rate, one-bit per sample output, which is then converted to an analog signal
by lowpass filtering and further smoothing with analog filters.