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

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

CODE

 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