Directform FIR polyphase sample rate converter
mfilt.firsrc
will be removed in a future
release. Use dsp.FIRRateConverter
instead.
hm = mfilt.firsrc(l,m,num)
hm = mfilt.firsrc(l,m,num)
returns
a directform FIR polyphase sample rate converter. l
specifies
the interpolation factor. It must be an integer and when omitted in
the calling syntax, it defaults to 2.
m
is the decimation factor. It must be an
integer. If not specified, m
defaults to 1. If l
is
also not specified, m
defaults to 3 and the overall
rate change factor is 2/3.
You specify the coefficients of the FIR lowpass filter used
for sample rate conversion in num
. If omitted,
a lowpass Nyquist filter with gain l
and cutoff
frequency of π/max
(l
,m
)
is the default.
Combining an interpolation factor and a decimation factor lets
you use mfilt.firsrc
to perform fractional interpolation
or decimation on an input signal. Using an mfilt.firsrc
object
applies a rate change factor defined by l
/m
to
the input signal. For proper rate changing to occur, l
and m
must
be relatively prime — meaning the ratio l
/m
cannot
be reduced to a ratio of smaller integers.
When you are doing samplerate conversion with large values
of l
or m
, such as l
or m
greater
than 20, using the mfilt.firsrc
structure
is the most effective approach.
Make this filter a fixedpoint or singleprecision filter by
changing the value of the Arithmetic
property
for the filter hm
as follows:
To change to singleprecision filtering, enter
set(hm,'arithmetic','single');
To change to fixedpoint filtering, enter
set(hm,'arithmetic','fixed');
You can use the realizemdl
method
to create a Simulink^{®} block of a filter created using mfilt.firsrc
.
The following table describes the input arguments for creating hm
.
Input Argument  Description 

 Interpolation factor for the filter. 
 Vector containing the coefficients of the FIR lowpass
filter used for interpolation. When 
 Decimation factor for the filter. 
This section describes the properties for both floatingpoint filters (doubleprecision and singleprecision) and fixedpoint filters.
Every multirate filter object has properties that govern the
way it behaves when you use it. Note that many of the properties are
also input arguments for creating mfilt.firsrc
objects.
The next table describes each property for an mfilt.firsrc
filter
object.
Name  Values  Description 


 Defines the arithmetic the filter uses. Gives you the
options 
 String  Reports the type of filter object. You cannot set this
property — it is always read only and results from your choice
of 
 Integers  Contains a value derived from the number of input samples
and the decimation factor — 
 Vector  Vector containing the coefficients of the FIR lowpass filter used for decimation. 

 Determines whether the filter states get restored to
zeros for each filtering operation. The starting values are the values
in place when you create the filter if you have not changed the filter
since you constructed it. 
 Positive integers. [2 3]  Specifies the interpolation and decimation factors [ 

 Contains the filter states before, during, and after filter operations. States act as filter memory between filtering runs or sessions. 
This table shows the properties associated with the fixedpoint
implementation of the mfilt.firsrc
filter.
The table lists all of the properties that a fixedpoint filter can have. Many of the properties listed are dynamic, meaning they exist only in response to the settings of other properties. To view all of the characteristics for a filter at any time, use
info(hm)
where hm
is a filter.
For further information about the properties of this filter
or any mfilt
object, refer to Multirate Filter Properties.
Name  Values  Description 

 Any positive or negative integer number of bits. [32]  Specifies the fraction length used to interpret data output by the accumulator. This is a property of FIR filters. 
 Any integer number of bits [39]  Sets the word length used to store data in the accumulator. 
 fixed for fixedpoint filters  Setting this to 
 [true], false  Specifies whether the filter automatically chooses the
proper fraction length to represent filter coefficients without overflowing.
Turning this off by setting the value to 
 Any integer number of bits [16]  Specifies the word length to apply to filter coefficients. 
 [FullPrecision], SpecifyPrecision  Controls whether the filter automatically sets the output
word and fraction lengths, product word and fraction lengths, and
the accumulator word and fraction lengths to maintain the best precision
results during filtering. The default value, 
 Any positive or negative integer number of bits [15]  Specifies the fraction length the filter uses to interpret input data. 
 Any integer number of bits [16]  Specifies the word length applied to interpret input data. 
 Any positive or negative integer number of bits [  Sets the fraction length used to interpret the numerator coefficients. 
 Any positive or negative integer number of bits [32]  Determines how the filter interprets the filter output
data. You can change the value of 
 Any integer number of bits [39]  Determines the word length used for the output data.
You make this property editable by setting 
 saturate, [wrap]  Sets the mode used to respond to overflow conditions
in fixedpoint arithmetic. Choose from either 
 Positive integers [2 3]  Specifies the interpolation and decimation factors [ 
 [  Sets the mode the filter uses to quantize numeric values when the values lie between representable values for the data format (word and fraction lengths).
The choice you make affects only the accumulator and output arithmetic. Coefficient and input arithmetic always round. Finally, products never overflow — they maintain full precision. 
 [true], false  Specifies whether the filter uses signed or unsigned fixedpoint coefficients. Only coefficients reflect this property setting. 

 Contains the filter states before, during, and after
filter operations. States act as filter memory between filtering runs
or sessions. The states use 
This is an example of a common audio rate change process —
changing the sample rate of a high end audio (48 kHz) signal to the
compact disc sample rate (44.1 kHz). This conversion requires a rate
change factor of 0.91875, or l
=
147 and m
= 160.
l = 147; m = 160; % Interpolation/decimation factors. hm = mfilt.firsrc(l,m); % Use the default FIR filter. fs = 48e3; % Original sample freq: 48 kHz. n = 0:10239; % 10240 samples, 0.213 seconds long. x = sin(2*pi*1e3/fs*n); % Original signal, sinusoid at 1 kHz. y = filter(hm,x); % 9408 samples, still 0.213 seconds. stem(n(1:49)/fs,x(1:49)) % Plot original sampled at 48 kHz. hold on % Plot fractionally decimated signal (44.1 kHz) in red stem(n(1:45)/(fs*l/m),y(13:57),'r','filled') xlabel('Time (sec)');ylabel('Signal Value')
Fractional decimation provides you the flexibility to pick and
choose the sample rates you want by carefully selecting l
and m
,
the interpolation and decimation factors, that result in the final
fractional decimation. The following figure shows the signal after
applying the rate change filter hm
to the original
signal.