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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

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