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(8.D.1.2) IIRInterpolator

Overview

Audio sample interpolator with IIR filter

Discussion

The IIR interpolator subsystem implements upsampler followed by an IIR filter. IIR low-pass filter is realized by adding allpass pair subsystem output. Vaiable cutoffFreq must be smaller than Fs*0.5/D, where Fs is a sampling rate and D is a 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

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