freqzmr
Compute DTFT approximation of the impulse response of a multirate or a single-rate filter
Since R2024a
Description
[
outputs the Discrete-Time Fourier Transform (DTFT) of the impulse response of the filter
object and the corresponding frequencies.Y
,Fout
] = freqzmr(filtObj
)
If the input filter is a single-rate FIR or IIR filter, the behavior of
freqzmr
and freqz
functions is equivalent. If the filter is an irreducible multirate
filter, use the freqzmr
function to analyze the frequency response of
the filter. For more details on the differences between these two functions, see Comparison of freqzmr and freqz functions. For more information on
reducible and irreducible multirate filters, see Overview of Multirate Filters.
freqzmr(
plots the magnitude and phase
of the impulse response DTFT. The function also color-codes the distortion that occurs due
to the rate changes. For more information on distortion, see Output Distortion.filtObj
)
Examples
Compute the Impulse Response DTFT of an FIR Rate Converter
Compute and plot the impulse response DTFT of the FIR rate converter using the freqzmr
function.
firRC = dsp.FIRRateConverter(5,7); freqzmr(firRC)
Compute and Plot the Impulse Response DTFT of Sample Rate Converter
Compute the impulse response DTFT of the sample rate converter using the freqzmr
function.
SRC = dsp.SampleRateConverter(InputSampleRate=77,... OutputSampleRate=32*24,... Bandwidth=23,... StopbandAttenuation=120); [Y,F] = freqzmr(SRC,MinimumFFTLength=1024);
When you do not specify any output arguments, the function plots the output DTFT on a convenience plot.
Plot the impulse response DTFT with maximum distortion sensitivity. Note that in the red regions, one can expect some distortion due to the rate conversion aliasing.
freqzmr(SRC,MinimumFFTLength=1024,DistortionThreshold=-inf);
To hide distortion coloring, set DistortionThreshold
to 0.
freqzmr(SRC,MinimumFFTLength=1024,DistortionThreshold=0);
Specify the frequency range to be "centered"
.
freqzmr(SRC,MinimumFFTLength=1024,DistortionThreshold=-inf,FrequencyRange="centered");
To compare, pass a random white noise signal to the sample rate converter. View the spectrum of the corresponding output in the spectrum analyzer.
You can see that the impulse response DTFT that the freqzmr
function computes is exactly the same as the spectrum of the output signal when you input a random white noise signal.
sa = spectrumAnalyzer(SampleRate=SRC.OutputSampleRate,PlotAsTwoSidedSpectrum=true,YLimits=[-200 20]); for k=1:10 u = randn(4096,1); y = SRC(u); sa(y); end
Compute and Plot Impulse Response DTFT of Composite Filter
Design a composite filter which is a cascade of these filters:
A single-rate filter.
A parallel structure of multirate filters.
Design the single-rate filter using the dsp.LowpassFilter
object in its default configuration.
singleRateFilter = dsp.LowpassFilter
singleRateFilter = dsp.LowpassFilter with properties: FilterType: 'FIR' DesignForMinimumOrder: true PassbandFrequency: 8000 StopbandFrequency: 12000 PassbandRipple: 0.1000 StopbandAttenuation: 80 NormalizedFrequency: false SampleRate: 44100 Show all properties
The parallel structure contains a cascade of dsp.FIRInterpolator
and dsp.FIRDecimator
objects, and a dsp.FIRRateConverter
object.
Design the dsp.FIRInterpolator
and the dsp.FIRRateConverter
objects using the designBandpassFIR
function.
B1 = 2.5*designBandpassFIR(FilterOrder=256,... CenterFrequency=0.3,Bandwidth=1/5); B2 = 0.25*designBandpassFIR(FilterOrder=256,... CenterFrequency=0.7,Bandwidth=1/20);
Create the parallel structure of multirate filters using the parallel
and cascade
functions.
multirateFilter = parallel(cascade(dsp.FIRInterpolator(5,B1),...
dsp.FIRDecimator(2)),dsp.FIRRateConverter(10,4,B2))
multirateFilter = dsp.ParallelFilter with properties: Branch1: [1×1 dsp.FilterCascade] Branch2: [1×1 dsp.FIRRateConverter] CloneBranches: true
Create a composite filter which is a cascade of the single-rate filter and the parallel filter.
compositeFilter = cascade(singleRateFilter,multirateFilter)
compositeFilter = dsp.FilterCascade with properties: Stage1: [1×1 dsp.LowpassFilter] Stage2: [1×1 dsp.ParallelFilter] CloneStages: true
Compute and plot the impulse response DTFT of this filter using the freqzmr
function. Set MinimumFFTLength
to 512. Set the frequency range to be "centered"
.
freqzmr(compositeFilter,MinimumFFTLength=512,FrequencyRange='centered')
Input Arguments
filtObj
— Input filter
filter System object™
Input filter, specified as one of these filter System objects:
Name-Value Arguments
Specify optional pairs of arguments as
Name1=Value1,...,NameN=ValueN
, where Name
is
the argument name and Value
is the corresponding value.
Name-value arguments must appear after other arguments, but the order of the
pairs does not matter.
Example: src =
dsp.SampleRateConverter(InputSampleRate=77,OutputSampleRate=32*24,Bandwidth=23,StopbandAttenuation=120);
[Y,Fout] =
freqzmr(src,MinimumFFTLength=1024,DistortionThreshold=-inf);
MinimumFFTLength
— Shortest discrete spectrum length
512 (default) | positive integer
Shortest discrete spectrum length, specified as a positive integer. When the
function resamples the spectrum internally (for example during downsampling), the size
of the resampled spectrum is no shorter than MinimumFFTLength
.
For more information, see DTFT of Downsampled Signal.
Data Types: double
InputSampleRate
— Sample rate of input signal in Hz
positive integer
Sample rate of input signal FsIn in Hz, specified as a positive integer.
If you do not specify the input sample rate, the function inherits this value from the filter object. If the input filter object contains an explicit property to specify the input sample rate, the function uses that property value. If the filter object does not contain any such property, the function uses a value of 1.
Data Types: double
FrequencyRange
— Frequency range of DTFT
"auto"
(default) | "onesided"
"twosided"
| "centered"
Frequency range of the DTFT, specified as one of these:
"auto"
–– The function automatically determine the appropriate choice ("onesided"
or"twosided"
) based on the system symmetry. The function chooses"onesided"
when the system spectrum is Hermitian symmetric, and chooses"twosided"
otherwise."onesided"
–– Impulse response DTFT contains only the positive frequencies. Frequency range is [0 Fsout/2]."twosided"
–– Frequency range is [0 Fsout]."centered"
–– Frequency range is [−Fsout/2 Fsout/2].
Fsout is the output sample rate and is given by the equation, Fsout = FsIn×(L/M), where L/M is the rate conversion ratio of the input filter object.
DistortionThreshold
— Output ambiguity distortion threshold in dB
-40
(default) | scalar in the range (−∞ 0]
Output ambiguity distortion threshold in dB of the convenience magnitude plot,
specified as a scalar in the range (−∞ 0]. The
DistortionThreshold
argument is the threshold over which the
signal is considered ambiguous.
Decrease the threshold value for a more sensitive detection of distortion. The function colors the convenience magnitude plot according to the distortion ratio. Red indicates maximum distortion (two or more branches have the same power), while blue indicates no distortion. If the distortion threshold is 0 dB, the function hides all distortion coloring.
The function ignores any distortion value below the value of
DistortionThreshold
and treats the output as
non-ambiguous.
Here is a plot of the impulse response DTFT of an FIR rate converter with
DistortionThreshold
set to −∞. This plot shows the most
sensitive detection of distortion.
Increase the threshold value to −40 dB. The function ignores all distortions below −40 dB.
Increasing the threshold value to 0 dB shows no distortion on the plot since the function ignores all the distortion values.
For more information, see Output Distortion.
Data Types: double
Output Arguments
Y
— DTFT approximation of output signal
column vector
DTFT approximation of impulse response, returned as a column vector. The length of
the impulse response DTFT depends on the MinimumFFTLength
value and
the rate conversion factors of the multirate filter stages.
Data Types: double
Complex Number Support: Yes
Fout
— Output frequencies in Hz
column vector
Frequencies in Hz corresponding to the impulse response DTFT approximation, returned
as a column vector. The length of Fout
equals the length of
Y
. The range of Fout
depends on the
FrequencyRange
you choose.
Data Types: double
More About
Discrete-Time Fourier Transform (DTFT) and its relation to DFT and FFT
The discrete-time Fourier transform (DTFT) of a sequence x[m] sampled at rate T is a periodic complex-valued function and is given by:
where, T is the sample rate of the signal. The DTFT plays a central role in the analysis of discrete signals and systems.
For computational purposes, DTFT is sampled on a finite set of frequencies and can be expressed in terms of DFT and FFT.
Comparison of freqzmr
and freqz
functions
The freqz
function plots the frequency response of
a single-rate FIR or IIR filter while the freqzmr
function plots the
DTFT approximation of the filter impulse response. This distinction stems from the fact that
multirate filters do not have a frequency response in general because a single input tone
can generate multiple output tones. The freqzmr
function accounts for
multirate operations in its DTFT approximation, while freqz
assumes
that the filter is a single-rate FIR or IIR filter.
Here are a few examples which illustrate the behavior of freqz
and
freqzmr
for various filters.
Single-Rate Filters
For single-rate FIR or IIR filters, the frequency response is equal to the DTFT of their
impulse response, and the behavior of freqzmr
and
freqz
functions is equivalent.
Consider the dsp.LowpassFilter
object which is a single-rate
filter.
lpFilter = dsp.LowpassFilter
Plot the frequency response of this filter using freqz
, and the
DTFT of the impulse response using freqzmr
. Overlay the two
plots.
The two plots are exactly the same.
Multirate Filters
Multirate filters cannot be characterized by a single transfer function, and do not have a frequency response in the usual sense. However, multirate filters have a well defined impulse response, whose DTFT gives valuable information about the spectral content of their outputs.
The freqzmr
function computes the DTFT approximation of the
multirate filter impulse response, and accounts for multirate operations while doing so.
This behavior allows the freqzmr
function to capture multirate effects
such as imaging and aliasing, while the freqz
function does not.
Here is an example. This filter is an antialiasing FIR filter with a downsample factor of 4.
firDec = dsp.FIRDecimator(4)
The freqz
function inspects this filter as a single-rate filter.
The freqzmr
function computes the DTFT approximation of the filter
impulse response. The DTFT plot accounts for aliasing and looks different compared to the
frequency response plot. The aliased components compensate to a flat response. The plot
shows these components by coloring them in red. For more information, see Output Distortion.
Multirate filters can be reducible or irreducible. A multirate filter is said to be
reducible if it can be modified through noble identities to produce a single-rate FIR or IIR
filter, as firDec
in the previous example. For more information, see Overview of Multirate Filters. If the
input filter is a reducible multirate filter, you can use the freqz
function to analyze the filter. The function analyzes the equivalent FIR or IIR stage, and
treats the filter as a single-rate FIR or IIR filter.
Multirate Filter Reducing to a Single-Stage
Consider this FIR rate converter which is trivially reducible for it has a single FIR stage.
firrc = dsp.FIRRateConverter(3,2,triang(9))
firrc = dsp.FIRRateConverter with properties: Main InterpolationFactor: 3 DecimationFactor: 2 NumeratorSource: 'Property' Numerator: [9×1 double]
With an input sample rate of 1 kHz, the output sample rate is 1.5 kHz. Plot the
frequency response of the filter using freqz
, and the DTFT of the
filter impulse response using freqzmr
. You can see that there is
distortion in the frequency range [0.4 0.75 kHz]. The DTFT plot shows this
distortion.
Consider this filter which is a cascade of a lowpass filter and an FIR rate converter. This filter can be reduced to a single stage.
lpfilt = dsp.LowpassFilter(SampleRate=1e3,... PassbandFrequency=80,StopbandFrequency=100,FilterType='IIR'); firrc = dsp.FIRRateConverter(3,4); cascFilt = cascade(lpfilt,firrc)
cascFilt = dsp.FilterCascade with properties: Stage1: [1×1 dsp.LowpassFilter] Stage2: [1×1 dsp.FIRRateConverter] CloneStages: true
Plot the frequency response of this filter using freqz
and the DTFT
of the filter impulse response using freqzmr
. Overlay and compare the
two plots.
The two functions generate plots which look similar but are not the same. Also, the
freqzmr
plot shows distortion regions in red, where there is rate
conversion aliasing.
Irreducible Multirate Filter
Lastly, if the input filter is an irreducible multirate filter, use the
freqzmr
function to analyze the filter in the frequency domain. For
more information on irreducible filters, see Overview of Multirate Filters.
Consider this filter which is a cascade of two FIR rate converters. This filter is irreducible.
firrc1 = dsp.FIRRateConverter; firrc2 = dsp.FIRRateConverter(3,4); cascFilt = cascade(firrc1,firrc2)
cascFilt = dsp.FilterCascade with properties: Stage1: [1×1 dsp.FIRRateConverter] Stage2: [1×1 dsp.FIRRateConverter] CloneStages: true
When you try to analyze this filter with freqz
, the function errors
out.
To analyze this filter, use the freqzmr
function, which in its
default configuration shows the DTFT of the filter impulse response.
freqzmr(src)
To summarize, for irreducible multirate filters, use the freqzmr
function. For single-rate FIR or IIR filters, the behavior of freqz
and
freqzmr
is equivalent.
Here is a table that compares the two functions and lists the key differences.
freqzmr | freqz | |
---|---|---|
Computation | DTFT approximation of filter impulse response h DTFT(h) | Frequency response of the filter, which is the transfer function evaluated on the unit circle |
Single-rate convolution filters | ✓ | ✓ |
Reducible multirate filters | ✓ | ✓ Performs analysis on the single-rate FIR or IIR
equivalent. The plot looks different compared to the
|
Irreducible multirate filters | ✓ | ╳ |
Captures multirate effects such as imaging an aliasing | ✓ | ╳ |
Support for normalized frequencies | ╳ | ✓ |
Algorithms
The freqzmr
algorithm treats the input filter object as a combination
of FIR or IIR (single-rate) filters and multirate operations (upsample and downsample). The
function processes the DTFT through each stage consecutively. For more information on DTFT,
see Discrete-Time Fourier Transform (DTFT) and its relation to DFT and FFT.
Single-Rate FIR or IIR Filters
Here is the block diagram of a single-rate filter with an input sequence x[m] and an output sequence y[m].
where, h is the impulse response of the single-rate filter.
The freqzmr
function computes the impulse response DTFT, that is,
DTFT(h) .
DTFT of Downsampled Signal
For a given input signal x[m], the output of the downsampling operation is given by y[n] = x[Mm].
The DTFT of the downsampled signal is periodic and its period 1/(MT) is 1/M of the period of X1/T.
For multirate filters, the components can overlap and interfere with each other resulting in aliasing. To resolve aliasing, it is important that one of the components dominates the rest in its value. This component contributes to the sum while the rest are 0 resolving the issue of aliasing. In most cases, this component corresponds to m=0 (baseband component), that is, Y(1/MT)(f)≅X1/T(f/M).
Multirate distortion is the level of ambiguity between these components. You can specify
the threshold in dB over which the function considers the signal ambiguous using the
DistortionThreshold
property. For more information, see Output Distortion.
Computing the DTFT of Downsampled Signal
DTFT downsampling is done through reshaping of the DTFT samples vector and taking a sum across columns. To downsample a signal with DTFT samples vector X by a decimation factor M, the downsampled DTFT is:
NX = size(X,1); Y = (1/M)*sum(reshape(X,NX/M,M),2);
Note that the DTFT array X is usually resampled at a higher frequency
resolution (increase by NX) prior to reshaping. This is done in order to
make NX length divisible by M, and to ensure that the
resolution of the downsampled DTFT Y is no less than
MinimumFFTLength
.
DTFT of Upsampled Signal
For a given input signal x[m], the output of the upsampling operation is given by:
y[n] operates at a rate L/T. The DTFT of the upsampled signal y is identical to the DTFT of x, that is, YL/T(f) = X1/T(f). The only difference is in the period, which is L/T, that is, L times larger on the frequency domain.
The freqzmr
function determines the DTFT of the upsampled signal by
replicating the input DTFT samples vector L times.
function Y = upsampleDTFT(X,L) Y = repmat(X,L,1); end
DTFT of Composite Filters (Cascade and Parallel)
Cascade Filters
For a cascade of filters, the freqzmr
function computes the output
DTFT by passing the output DTFT of each stage as the input DTFT of the following stage. The
first stage input is an impulse.
freqzmr(cascade(H1,H2)) = freqzmr(H2,...
InputSpectrum=freqzmr(H1))
If you have multiple stages in the cascade, the freqzmr
function
computes the impulse response DTFT
using:
function Y = cascadeDTFT(obj,X,MinimumFFTLength) Y = X; for k = 1:NumStages Y = outputDTFT(Stage(k),Y,MinimumFFTLength); end end
Parallel Filters
For parallel filters, the DTFT output is a sum of the DTFTs of the filter outputs on the individual branches.
Output Distortion
In multirate systems, an input frequency generates several output frequencies due to aliasing and imaging. As a result, an output frequency component is often a combination of several different frequency components.
The output DTFT of a multirate filter can be written as a sum of the components Q0, Q1, …., QM-1.
where:
M is the total decimation factor
Q0, Q1, …., QM-1 are DTFT components that the function computes through rate change operations.
Ideally, only one branch contributes to the sum while the rest are 0 (no-aliasing), that is, Qm = 0 for all m except for one. As soon as more than one of the branches has a nonzero value, Y(f) has aliasing.
The distortion is the level of ambiguity of the frequency branches Qm(f) in the sum above. It is defined as the inverse power ratio between the dominant component and the average power of the remaining components.
The distortion ratio ranges between 0 and 1 (or −∞ to 0 in dB), where higher values indicate more ambiguity. A 1:1 ratio (0 dB) indicates that there is no dominant component, and the signal is totally ambiguous. In contrast, a 0-ratio (−∞ dB) indicates that the output frequency is non-ambiguous.
The freqzmr
function sorts
m1, ….,
mM at every frequency according to the
corresponding Qm magnitude values. Note that the
distortion ratio is invariant under permutations, that is, changing the order of the
branches does not alter the distortion. The distortion ratio is a real number between 0 and
1. These values indicate where aliasing is present in a multirate system.
The function colors the impulse response DTFT depending on the value of the distortion
ratio. Red indicates maximum distortion (two or more branches have the same power), while
blue indicates no distortion. The function ignores any distortion value below the value of
DistortionThreshold
.
The following figure shows the distortion ratio in two frequencies. In the lower
frequency (f = 4.375 Hz), there is a fairly low distortion since the
dominant lobe is about 80 dB above the rest of the lobes. In the higher frequency
(f = 12.125 Hz), there is far more significant interference as alias
lobe 4 is much closer to the dominant lobe 1. This is indicated by the red color on the
freqzmr
plot.
Version History
Introduced in R2024a
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