Overview
Audio sample interpolator with IIR filter
Discussion
The IIR interpolator subsystem implements an upsampler followed by an IIR filter. The IIR low-pass filter is implemented as an 8th order elliptical filter. The module variable cutoffFreq must be smaller than Fs*0.5/D, where Fs is the sampling rate and D is the interpolation factor.
The block size of the output pin must be divisible by the interpolation factor D. The input block size equals the output block size divided by D. The input sample rate equals the output sample rate divided by D.
Type Definition
-Not Shown-
Variables
Properties
Name | Type | Usage | isHidden | Default value | Range | Units |
cutoffFreq | float | parameter | 0 | 10800 | 20:24000 | Hz |
Rs | float | parameter | 0 | 60 | 10:100 | dB |
L | int | const | 0 | 2 | Unrestricted |
Pins
Input Pins
Name: in
Description: Audio input
Data type: float
Channel range: Unrestricted
Block size range: Unrestricted
Sample rate range: Unrestricted
Complex support: Real
Output Pins
Name: out
Description: Audio input
Data type: float
Scratch Pins
Channel count: 1
Block size: 64
Sample rate: 96000
Channel count: 1
Block size: 64
Sample rate: 96000
Channel count: 1
Block size: 64
Sample rate: 96000
Channel count: 1
Block size: 64
Sample rate: 96000
MATLAB Usage
File Name: iir_interpolator_subsystem.m
SYS=iir_interpolator_subsystem(NAME, L)
Create a sub-system that interpolates audio samples with IIR filter(s). IIR
filters are implemented with allpass_pair sub-system with Elliptic filter
option.
Arguments:
NAME - name of the module.
L - Interpolation factor.
Copyright 2018. DSP Concepts, Inc. All Rights Reserved.
Author: Taka Unno