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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
"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:
Symbol of downsampler
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].
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.
Where k=1,2N.
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
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.
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.
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.