Multistage samplerate conversion
Signal Operations
dspsigops
The SampleRate Converter block implements a multistage FIR samplerate converter. This multistage FIR converter converts the rate of each channel of the input signal from the input sample rate to the output sample rate. Multistage implementations minimize the amount of computation required by the samplerate conversions by first reducing the sample rate of the input signal. Next, the block determines the optimal number of decimators and interpolators required based on the parameters specified in the block dialog box. Then the block designs filters in the individual stages accordingly.
The input frame size must be a multiple of the decimation factor of the rate converter. The decimation factor depends on the parameter setting of the converter. To determine the decimation factor, in the block dialog box, click View Info .
Each column of a twodimensional input signal is treated as a separate channel. If the input is a twodimensional signal, the first dimension represents the channel length (or frame size), and the second dimension represents the number of channels. If the input is a onedimensional signal, then it is interpreted as a single channel. The inputs to the block can be single or double, and real or complex.
This block supports SIMD code generation. For details, see Code Generation.
Sample rate of the input signal, specified as a positive scalar in Hz. The
input sample rate must be greater than the bandwidth of interest. The
default is 48e3
.
Sample rate of the output signal, specified as a positive scalar in Hz.
The output sample rate must be greater than the bandwidth of interest. The
default is 96e3
.
Maximum allowed tolerance for output sample rate, specified as a positive
scalar in the range [0
,1
]. The default
is 0
.
The actual output sample rate varies but is within the specified range. For example, suppose that you set the Tolerance for output sample rate, to 0.01. Then the actual output sample rate is in the range given by sample rate of output signal ± 1%. This flexibility allows for a simpler filter design.
Twosided bandwidth of interest (after the rate of conversion), specified
as a positive scalar in Hz. The default is 40e3
.
Minimum amount of attenuation for aliased components in the stopband,
specified as a positive scalar in dB. The default is 80
.
This parameter is the minimum amount by which any aliasing involved in the
process is attenuated.
Opens the Filter Visualization Tool FVTool and displays the magnitude/phase response of the SampleRate Converter. The response is based on the block dialog box parameters. Changes made to these parameters update FVTool.
To update the magnitude response while FVTool is running, modify the dialog box parameters and click Apply.
Displays information about the filter system of the SampleRate Converter block:
Overall Interpolation Factor
Overall Decimation Factor
Number of Filters
Multiplication per Input Sample
Number of Coefficients
Filters
The button brings the functionality of the info
method into the Simulink^{®} environment.
Type of simulation to run. You can set this parameter to:
Code generation
(default)
Simulate model using generated C code. The first time you run
a simulation, Simulink generates C code for the block. The C code is
reused for subsequent simulations, as long as the model does not
change. This option requires additional startup time but
provides faster simulation speed
than Interpreted
execution
.
Interpreted execution
Simulate model using the MATLAB^{®} interpreter. This option shortens startup
time but has slower simulation speed than Code
generation
.
Port  Supported Data Types 

Input 

Output 

dsp.SampleRateConverter  DSP System Toolbox 
Farrow Rate Converter  DSP System Toolbox 
This block brings the capabilities of the dsp.SampleRateConverter
System object™ to the Simulink environment.
For information on the algorithms used by this block, see the Algorithms section of dsp.SampleRateConverter
.