Audio Processing Algorithm Design

Audio processing tools, algorithm design and modularization, stream processing

Audio Toolbox™ is optimized for real-time audio stream processing. Use these features individually or as part of a larger algorithm to create effects, analyze signals, and process audio.

Functions

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Filter Design

`designVarSlopeFilter`	Design variable slope lowpass or highpass IIR filter
`designParamEQ`	Design parametric equalizer
`designShelvingEQ`	Design shelving equalizer

Resampling

`audioresample`	Sample-rate conversion of audio signals (Since R2023b)
`designAudioResampler`	Design sample-rate converter for audio signals (Since R2023b)

Filters

`crossoverFilter`	Audio crossover filter
`multibandParametricEQ`	Multiband parametric equalizer
`graphicEQ`	Standards-based graphic equalizer
`octaveFilter`	Octave-band and fractional octave-band filter
`octaveFilterBank`	Octave and fractional-octave filter bank
`weightingFilter`	Frequency-weighted filter
`shelvingFilter`	Second-order IIR shelving filter (Since R2022a)
`gammatoneFilterBank`	Gammatone filter bank

Dynamic Range Control

`compressor`	Dynamic range compressor
`expander`	Dynamic range expander
`limiter`	Dynamic range limiter
`noiseGate`	Dynamic range gate

Effects

`reverberator`	Add reverberation to audio signal
`shiftPitch`	Shift audio pitch
`stretchAudio`	Time-stretch audio
`audioTimeScaler`	Apply time scaling to streaming audio

Transforms and Windows

`kbdwin`	Kaiser-Bessel-derived window
`mdct`	Modified discrete cosine transform
`imdct`	Inverse modified discrete cosine transform

Blocks

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Dynamic Range Control

Compressor	Dynamic range compressor
Expander	Dynamic range expander
Limiter	Dynamic range limiter
Noise Gate	Dynamic range gate

Effects

Reverberator

Add reverberation to audio signal

Filters

Crossover Filter	Audio crossover filter
Weighting Filter	Weighted frequency response filter
Octave Filter	Octave-band and fractional octave-band filter
Octave Filter Bank	Octave-band and fractional octave-band filter bank (Since R2021a)
Single-Band Parametric EQ	Second-order parametric equalizer filter
Graphic EQ	Standards-based graphic equalizer
Multiband Parametric EQ	Multiband parametric equalizer (Since R2021b)
Gammatone Filter Bank	Gammatone filter bank (Since R2021b)
Shelving Filter	Second-order IIR shelving filter (Since R2022a)

Filter Design

Parametric Equalizer Design	Design tunable parametric equalizer coefficients (Since R2024a)
Shelving Equalizer Design	Design tunable shelving equalizer coefficients (Since R2024a)
Variable Slope Filter Design	Design tunable lowpass or highpass filter coefficients (Since R2024a)

Topics

Real-Time Audio in MATLAB
Create a script to process and analyze real-time audio signals.
Real-Time Audio in Simulink
Create a model using the Simulink^® templates and blocks for audio processing.
Equalization
Understand the foundations of audio equalization and how equalizers are implemented in Audio Toolbox.
Dynamic Range Control
Understand the foundations of dynamic range control and how compressors, limiters, expanders, and noise gates are implemented in Audio Toolbox.
Parametric Equalizer Design
Design parametric equalizer filters. Parametric equalizers are digital filters used in audio for adjusting the frequency content of a sound signal.

Featured Examples

Automated Design of Audio Filters for Room Equalization

Combine Optimization Toolbox™ and Audio Toolbox to develop an algorithm that automatically tunes a set of filter parameters.

Open Live Script

Multistage Sample-Rate Conversion of Audio Signals

Use a multistage/multirate approach to sample rate conversion between different audio sampling rates.

Open Script

Graphic Equalization

Demonstrates two forms of graphic equalizers constructed using building blocks from Audio Toolbox™. It also shows how to export them as VST plugins to be used in a Digital Audio Workstation (DAW).

Open Live Script

Multiband Dynamic Range Compression

Simulate a digital audio multiband dynamic range compression system.

Open Live Script

Delay-Based Audio Effects

Design and use three audio effects that are based on varying delay: echo, chorus and flanger. The example also shows how the algorithms, developed in MATLAB, can be easily ported to Simulink.

Open Live Script

Psychoacoustic Bass Enhancement for Band-Limited Signals

An audio plugin designed to enhance the perceived sound level in the lower part of the audible spectrum.

Open Live Script

Audio Phaser Using Multiband Parametric Equalizer

Implement a real-time audio "phaser" effect which can be tuned by a user interface (UI). It also shows how to generate a VST plugin for the phaser that you can import into a Digital Audio Workstation (DAW).

Open Live Script

Identification and Separation of Panned Audio Sources in a Stereo Mix

Extract an audio source from a stereo mix based on its panning coefficient. This example illustrates MATLAB® and Simulink® implementations.

Open Model

Acoustic Noise Cancellation Using LMS

Use the Least Mean Square (LMS) algorithm to subtract noise from an input signal. The LMS adaptive filter uses the reference signal on the Input port and the desired signal on the Desired port to automatically match the filter response. As it converges to the correct filter model, the filtered noise is subtracted and the error signal should contain only the original signal.

Open Script

Active Noise Control Using a Filtered-X LMS FIR Adaptive Filter

Apply adaptive filters to the attenuation of acoustic noise via active noise control.

Open Script

Pitch Shifting and Time Dilation Using a Phase Vocoder in MATLAB

Implement a phase vocoder to time stretch and pitch scale an audio signal.

Open Live Script

Pitch Shifting and Time Dilation Using a Phase Vocoder in Simulink

Use a phase vocoder to implement time dilation and pitch shifting of an audio signal.

Open Model

Remove Interfering Tone From Audio Stream

Remove a 250 Hz interfering tone from a streaming audio signal using a notch filter.

Open Live Script

Vorbis Decoder

Implement a Vorbis decoder, which is a freeware, open-source alternative to the MP3 standard. This audio decoding format supports the segmentation of encoded data into small packets for network transmission.

Open Model

Add Reverberation Using Freeverb Algorithm

Apply reverberation to audio by using the Freeverb reverberation algorithm. The reverberation can be tuned using a user interface (UI) in MATLAB or through a MIDI controller. This example illustrates MATLAB® and Simulink® implementations.

Open Script

Cochlear Implant Speech Processor

Simulate the design of a cochlear implant that can be placed in the inner ear of a profoundly deaf person to restore partial hearing. Signal processing is used in cochlear implants to convert sound to electrical pulses. The pulses can bypass the damaged parts of a deaf person's ear and be transmitted to the brain to provide partial hearing.

Open Model

Dynamic Range Compression Using Overlap-Add Reconstruction

Compress the dynamic range of a signal by modifying the range of the magnitude at each frequency bin. This nonlinear spectral modification is followed by an overlap-add FFT algorithm for reconstruction. This system might be used as a speech enhancement system for the hearing impaired. The algorithm in this simulation is derived from a patented system for adaptive processing of telephone voice signals for the hearing impaired originally developed by Alvin M. Terry and Thomas P. Krauss at US West Advanced Technologies Inc., US patent number 5,388,185.

Open Model

Acoustic Echo Cancellation (AEC)

Apply adaptive filters to acoustic echo cancellation (AEC).

Open Script

Time-Frequency Masking for Harmonic-Percussive Source Separation

Time-frequency masking is the process of applying weights to the bins of a time-frequency representation to enhance, diminish, or isolate portions of audio.

Open Live Script