IIRInterpolator

Overview

Audio sample interpolator with IIR filter

Discussion

The IIR interpolator module implements an upsampler followed by an IIR low-pass filter. The IIR 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.

This module accepts any input block size.
The output block size equals the input block size times the interpolation factor D.
The output sample rate equals the input sample rate times D

Module Pack

Advanced

ClassID

classID =

Type Definition

typedef struct _ModuleIIRInterpolator
{
ModuleInstanceDescriptor instance;            // Common Audio Weaver module instance structure
} ModuleIIRInterpolatorClass;

Variables

Properties

Name	Type	Usage	Default Value	Range	Units
cutoffFreq	float	parameter	10800	20:24000	Hz
Rs	float	parameter	60	10:100	dB
L	int	const	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