Apply pulse shaping by downsampling signal using raised cosine FIR filter
Comm Filters
The Raised Cosine Receive Filter block filters the input signal
using a normal raised cosine FIR filter or a square root raised cosine
FIR filter. It also downsamples the filtered signal if you set the Output
mode parameter to Downsampling
.
The FIR Decimation block implements
this functionality. The Raised Cosine Receive Filter block's icon
shows the filter's impulse response.
Characteristics of the raised cosine filter are the same as in the Raised Cosine Transmit Filter block, except that the length of the filter's input response has a slightly different expression: L * Filter span in symbols + 1, where L is the value of the Input samples per symbol parameter (not the Output samples per symbol parameter, as in the case of the Raised Cosine Transmit Filter block).
The block normalizes the filter coefficients to unit energy.
If you specify a Liner amplitude filter gain other
than 1
, then the block scales the normalized filter
coefficients using the gain value you specify.
To have the block decimate the filtered signal, set the Decimation
factor parameter to a value greater than 1
.
If K
represents the Decimation
factor parameter value, then the block retains 1/K of
the samples, choosing them as follows:
If the Decimation offset parameter is zero, then the block selects the samples of the filtered signal indexed by 1, K+1, 2*K+1, 3*K+1, etc.
If the Decimation offset parameter is a positive integer less than M, then the block initially discards that number of samples from the filtered signal and downsamples the remaining data as in the previous case.
To preserve the entire filtered signal and avoid decimation,
set Decimation factor to 1
.
This setting is appropriate, for example, when the output from the
filter block forms the input to a timing phase recovery block such
as Squaring Timing Recovery. The timing
phase recovery block performs the downsampling in that case.
This block accepts a column vector or matrix input signal. For information about the data types each block port supports, see the Supported Data Type table on this page.
If you set Decimation factor to 1
,
then the input and output signals share the same sampling mode, sample
time, and vector length.
If you set Decimation factor to K,
which is greater than 1
, then K and
the input sampling mode determine characteristics of the output signal:
When you set the Rate options parameter
to Enforce singlerate processing
, the
input and output of the block have the same sample rate. To genereate
the output while maintaining the input sample rate, the block resamples
the data in each column of the input such that the frame size of the
output (M_{o}) is 1/K times
that of the input (M_{o} = M_{i}/K),
In this mode, the input frame size, M_{i},
must be a multiple of K.
When you set the Rate options parameter
to Allow multirate processing
, the input
and output of the block are the same size, but the sample rate of
the output is K times slower than that of the input.
When the block is in multirate processing mode, you must also specify
a value for the Input processing parameter:
When you set the Input processing parameter
to Elements as channels (sample based)
,
the block treats an MbyN matrix
input as M*N independent channels,
and processes each channel over time. The output sample period (T_{so})
is K times longer than the input sample period
(T_{so} = K*T_{si}),
and the input and output sizes are identical.
When you set the Input processing parameter
to Columns as channels (frame based)
, the
block treats an M_{i}byN matrix
input as N independent channels. The block processes
each column of the input over time by keeping the frame size constant
(M_{i}=M_{o}),
and making the output frame period (T_{fo}) K times
longer than the input frame period (T_{fo} = K*T_{fi}).
To examine or manipulate the coefficients of the filter that this block designs, select Export filter coefficients to workspace. Then set the Coefficient variable name parameter to the name of a variable that you want the block to create in the MATLAB^{®} workspace. Running the simulation causes the block to create the variable, overwriting any previous contents in case the variable already exists.
For information pertaining to the latency of the block, see details in FIR Decimation.
Specify the filter shape as Square root
or Normal
.
Specify the rolloff factor of the filter. Use a real number
between 0
and 1
.
Specify the number of symbols the filter spans as an even, integervalued
positive scalar. The default is 10
. Because the
ideal raised cosine filter has an infinite impulse response, the block
truncates the impulse response to the number of symbols that this
parameter specifies.
An integer greater than 1 representing the number of samples that represent one symbol in the input signal.
Specify the decimation factor the block applies to the input signal. The output samples per symbol equals the value of the input samples per symbol divided by the decimation factor. If the decimation factor is one, then the block only applies filtering. There is no decimation.
Specify the decimation offset in samples. Use a value between 0
and Decimation
factor 1
.
Specify a positive scalar value that the block uses to scale
the filter coefficients. By default, the block normalizes filter coefficients
to provide unit energy gain. If you specify a gain other than 1
,
the block scales the normalized filter coefficients using the gain
value you specify.
Specify how the block processes the input signal. You can set this parameter to one of the following options:
Columns as channels (frame based)
—
When you select this option, the block treats each column of the input
as a separate channel.
Elements as channels (sample based)
—
When you select this option, the block treats each element of the
input as a separate channel.
Specify the method by which the block should filter and downsample the input signal. You can select one of the following options:
Enforce singlerate processing
—
When you select this option, the block maintains the input sample
rate and processes the signal by decreasing the output frame size
by a factor of K. To select this option, you must
set the Input processing parameter to Columns
as channels (frame based)
.
Allow multirate processing
—
When you select this option, the block processes the signal such that
the output sample rate is K times slower than the
input sample rate.
Select this check box to create a variable in the MATLAB workspace that contains the filter coefficients.
The name of the variable to create in the MATLAB workspace. This field appears only if Export filter coefficients to workspace is selected.
If you click this button, then MATLAB launches the Filter
Visualization Tool, fvtool
, to analyze the raised
cosine filter whenever you apply any changes to the block's parameters.
If you launch fvtool
for the filter, and subsequently
change parameters in the mask, fvtool
will not
update. You will need to launch a new fvtool
in
order to see the new filter characteristics. Also note that if you
have launched fvtool
, then it will remain open
even after the model is closed.
Select the rounding mode for fixedpoint operations. The block
uses the Rounding mode when the result of a fixedpoint
calculation does not map exactly to a number representable by the
data type and scaling storing the result. The filter coefficients
do not obey this parameter; they always round to Nearest
.
For more information, see Rounding Modes (DSP System Toolbox) in the DSP System Toolbox™ documentation
or Rounding Mode: Simplest (FixedPoint Designer) in
the FixedPoint Designer™ documentation.
Select the overflow mode for fixedpoint operations. The filter coefficients do not obey this parameter; they are always saturated.
Choose how you specify the word length and the fraction length of the filter coefficients (numerator and/or denominator).
See the Coefficients section of the FIR Decimation help page and Filter Structure Diagrams (DSP System Toolbox) in DSP System Toolbox Reference Guide for illustrations depicting the use of the coefficient data types in this block:
See the Coefficients subsection of the Digital Filter help page for descriptions of parameter settings.
When you select Same word length as input
,
the word length of the filter coefficients match that of the input
to the block. In this mode, the fraction length of the coefficients
is automatically set to the binarypoint only scaling that provides
you with the best precision possible given the value and word length
of the coefficients.
When you select Specify word length
,
you are able to enter the word length of the coefficients, in bits.
In this mode, the fraction length of the coefficients is automatically
set to the binarypoint only scaling that provides you with the best
precision possible given the value and word length of the coefficients.
When you select Binary point scaling
,
you are able to enter the word length and the fraction length of the
coefficients, in bits. If applicable, you are able to enter separate
fraction lengths for the numerator and denominator coefficients.
When you select Slope and bias scaling
,
you are able to enter the word length, in bits, and the slope of the
coefficients. If applicable, you are able to enter separate slopes
for the numerator and denominator coefficients. This block requires
poweroftwo slope and a bias of zero.
The filter coefficients do not obey the Rounding
mode and the Overflow mode parameters;
they are always saturated and rounded to Nearest
.
Use this parameter to specify how you would like to designate the product output word and fraction lengths. See Filter Structure Diagrams (DSP System Toolbox) and Multiplication Data Types (DSP System Toolbox) in DSP System Toolbox Reference Guide for illustrations depicting the use of the product output data type in this block:
When you select Same as input
,
these characteristics match those of the input to the block.
When you select Binary point scaling
,
you are able to enter the word length and the fraction length of the
product output, in bits.
When you select Slope and bias scaling
,
you are able to enter the word length, in bits, and the slope of the
product output. This block requires poweroftwo slope and a bias
of zero.
Use this parameter to specify how you would like to designate the accumulator word and fraction lengths. See Filter Structure Diagrams (DSP System Toolbox) and Multiplication Data Types (DSP System Toolbox) for illustrations depicting the use of the accumulator data type in this block:
When you select Same as input
,
these characteristics match those of the input to the block.
When you select Same as product output
,
these characteristics match those of the product output.
When you select Binary point scaling
,
you are able to enter the word length and the fraction length of the
accumulator, in bits.
When you select Slope and bias scaling
,
you are able to enter the word length, in bits, and the slope of the
accumulator. This block requires poweroftwo slope and a bias of
zero.
Choose how you specify the output word length and fraction length:
When you select Same as input
,
these characteristics match those of the input to the block.
When you select Same as accumulator
,
these characteristics match those of the accumulator.
When you select Binary point scaling
,
you are able to enter the word length and the fraction length of the
output, in bits.
When you select Slope and bias scaling
,
you are able to enter the word length, in bits, and the slope of the
output. This block requires poweroftwo slope and a bias of zero.
Select this parameter to prevent any fixedpoint scaling you specify in this block mask from being overridden by the autoscaling tool in the FixedPoint Tool.
This block supports HDL code generation using HDL Coder™. HDL Coder provides additional configuration options that affect HDL implementation and synthesized logic. For more information on implementations, properties, and restrictions for HDL code generation, see Raised Cosine Receive Filter in the HDL Coder documentation.
Port  Supported Data Types 

In 

Out 
