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Multirate Signal Processing

This document discusses multirate digital signal processing. It begins by defining multirate signal processing as processing data at more than one sampling rate, using decimation to reduce the rate and interpolation to increase it. It then covers the need for multirate DSP, decimation and interpolation by integer factors, and conversion by non-integer factors. The document also discusses filter requirements, multistage approaches, and applications like subband coding of speech signals and transmultiplexers.

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100% found this document useful (1 vote)
241 views25 pages

Multirate Signal Processing

This document discusses multirate digital signal processing. It begins by defining multirate signal processing as processing data at more than one sampling rate, using decimation to reduce the rate and interpolation to increase it. It then covers the need for multirate DSP, decimation and interpolation by integer factors, and conversion by non-integer factors. The document also discusses filter requirements, multistage approaches, and applications like subband coding of speech signals and transmultiplexers.

Uploaded by

Harish
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
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UNIT 5

Multirate Digital Signal


Processing
WHAT IS MULTIRATE SIGNAL PROCESSING

• To process data at more than one sampling rate is called


multirate signal processing.
• The two primary operations are 1]decimation 2]Interpolation
• Decimation reduces sampling rate (i.e. sampling frequency) by
effectively compressing data and retaining only the desired
information.
• Interpolation increases the sampling rate so that it is easier to
process or to achieve compatibility with another system.

2
NEED OF MULTIRATE DIGITAL SIGNAL PROCESSING

System
x(n) y(n)

3
SAMPLING RATE REDUCTION: DECIMATION BY INTEGER FACTOR(M/D)
• Figure shows the process of
decimating a signal x(n) by an
integer factor M.
• It consists of digital anti aliasing
filter h(k).
• Sampling rate compressor
,symbolized by down arrow and
decimation factor M.
• The rate compressor reduces the
To prevent aliasing at lower rate the digital filter sampling rate from Fs to Fs/M.
used to bandlimit the input signal to less than Fs/2M
beforehand. Thus signal x(n) is first band limited
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The sampling rate reduction is achieved by discarding M-1 samples for every M samples
of the filtered signal, w(n).
For M=3 ,Two samples out of
every three samples of x(n) are
discarded.

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SAMPLING RATE INCREASE: INTERPOLATION BY INTEGER FACTORS
Interpolation is the digital equivalent of D-A Converter
X(n) is signal at sampling frequency Fs ,the interpolation process increases the sampling
rate by L to LFs.

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For each sample of x(n) ,the
expanlder inserts L-1 zero
valued samples to form the
new signal w(m) at a rate of
LFs.
This signal is then applied to
LPF to remove images created
by the rate increase to yield
y(m).
If L=3 ,for each sample of
x(n,)three output samples are
produced.
This is due to two zero valued
samples inserted by expander.
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SAMPLING RATE CONVERSION BY NON INTEGER FACTORS
OR L/M OR I/D CONVERSION
• In many applications we need to change the sampling rate by non integer factor.
• In practice such non-integer factor is represented by rational number that is the ratio of
two integers i.e L/M or I/D.
• L and M are integer such that Ratio of L/M or I/D is as close to the desired factor as
possible.
• The sampling frequency change is achieved by first interpolating the data by L and then
decimating the data by M.

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The two LPF h1(k) and h2(k) can be combined into a single filter since they are in
cascade and have common sampling frequency.
If M>L then resulting operation is decimation process by a non integer and when M<L
then it is interpolation.

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• In figure interpolation by
factor 3/2.
• The sample rate is first
increased by 3 by inserting
two non zero valued samples
for each sample of x(n).
• Then LPF is applied.
• The filtered data is then
reduced by factor of 2

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NUMERICAL: REFEREE SOLVED PROBLEMS IN IFEACHOR BOOK(PROBLEM 9.1 TO 9.4)

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MULTISTAGE APPROACH TO SAMPLING RATE CONVERSION
• When large change in the sampling rate is required it is more efficient to change
the two or more stages than in one single stage.
• In fact more practical multirate systems employ the multistage approach because it
allows a gradual reduction or increase in the sampling rate, leading to a significant
relaxation in the requirements of anti-aliasing or anti-imaging filter at each stage.

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Multistage Decimation Process
Figure shows I-stage decimation process. Each stage is independent decimator.
The overall decimation factor M is expressed as:
M=M1M2M3…….Mi
If M>>1 ,the multistage approach leads to much reduced computational and
storage requirments.

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FILTER REQUIRMENTS FOR INDIVIDUAL STAGES:
For multistage decimator the filter requirements for each stage to ensure that the
overall filter requirments are met are:
Fi=output sampling frequency
Fi=𝐹𝑖−1 /𝑀𝑖 i=1,2,3….I

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b)Multistage structure
c)Filter Specifications for stage i

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DETERMINING THE NUMBER OF STAGES AND DECIMATION FACTORS

Multiplication per second=MPS=σ𝐼𝑖=1 𝑁𝑖𝐹𝑖


Total storage requirement=TSR= σ𝐼𝑖=1 𝑁𝑖

IN general for optimum MPS and TSR the


decimation facor satisfy the following relation
M1> M2 > ………MI

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POLYPHASE FILTER STRUCTURE
Polyphase structure for FIR filter were developed for the efficient implementation of
sampling rate converters.
M-component polyphase decomposition and Pi(z)
is polyphase components of H(z).

Each subsequence is obtained by down sampling


a delayed version of the original impulse
response

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Blk diagm of polyphase filter structure for M=3

18
APPLICATIONS OF MULTIRATE DSP:
Subband Coding of Speech Signal:
• A variety of technique have been developed to efficiently represent
speech signals in digital form for either transmission or storage.
• Most of the speech energy is contained in lower frequencies, we would
like tot encode the lower frequency band with more bits than high
frequency band.
• Subband coding is a method where speech signal is subdivided into
several frequency bands and each band is digitally encoded
separately.

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• Speech signal is sampled at a rate Fs samples per second.
• The first frequency subdivision splits signal spectrum into two equal width segments
• Low pass signal (0≤F≤Fs/4) and High pass signal (Fs/4≤F≤Fs/2).
• The second frequency subdivision splits low pass signal from first stage into two equal bands
low pass signal (0≤F≤Fs/8) and high pass signal (Fs/8≤F≤Fs/4).
• Similarly third subdivision takes place.

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• Decimation by a factor 2 is performed after frequency subdivision.
• By allocating a different number of bits per samples to the signal in the four subbands, we can
achieve a reduction in the bit rate of the quantized speech signal.
• To avoid aliasing use quadrature mirror filter

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Subband speech coder
• The synthesis method for subband
encoded speech signal is basically
the reverse of the encoding process.
• The signals in adjacent lowpass and
highpass frequency bands are
interpolated, filtered and combined
as shown in fig.

• Application of subband coding:


• It is efficient method to achieve
data compression in image signal
processing.

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Synthesis of subband encoded signals
TRANSMULTIPLEXERS
• Transmultiplexers are devices for converting between time-division-multiplexed (TDM)
signals and frequency–division –multiplexed(FDM) signals.
• Consider FDM to TDM Conversion:

23 FDM TO TDM transmultiplexer


TDM TO FDM TRANSMULTIPLEXER

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REFERENCES:

2 .Emmanuel Ifeachor, “Digital Signal Processing”, 2nd edition.

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