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
Use a phase vocoder to implement time dilation and pitch shifting of an audio signal.
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
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.
Extract an audio source from a stereo mix based on its panning coefficient. This example illustrates MATLAB® and Simulink® implementations.
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
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
Design parametric equalizer filters. Parametric equalizers are digital filters used in audio for adjusting the frequency content of a sound signal. Parametric equalizers provide
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
Simulate a digital audio multiband dynamic range compression system.
Remove a 250 Hz interfering tone from a streaming audio signal using a notch filter.
Implement a phase vocoder to time stretch and pitch scale an audio signal.
An audio plugin designed to enhance the perceived sound level in the lower part of the audible spectrum.
Apply adaptive filters to the attenuation of acoustic noise via active noise control.
Suppress the volume of loud sounds and visualize the applied dynamic range control gain.
Examine the Weighting Filter block in a Simulink® model and tune parameters.
Use the Compressor block to suppress loud sounds and visualize the applied compression gain.
Examine the Parametric EQ Filter block in a Simulink® model and tune parameters.
Use the Expander block to attenuate low-level noise and visualize the applied dynamic range control gain.
Measure momentary and short-term loudness before and after compression of a streaming audio signal in Simulink®.
Examine the Reverberator block in a Simulink® model and tune parameters. The reverberation parameters in this model mimic a large room with hard walls, such as a gymnasium.
Divide a mono signal into a stereo signal with distinct frequency bands. To hear the full effect of this simulation, use a stereo speaker system, such as headphones.
Demonstrates two forms of graphic equalizers constructed using building blocks from Audio System Toolbox. It also shows how to export them as VST plugins to be used in a Digital Audio
Use a multistage/multirate approach to sample rate conversion between different audio sampling rates.
This model minimizes the plosives of a speech signal by applying highpass filtering and low-band compression.
This model enables you to apply dynamic range compression to an audio signal while staying inside a preset loudness range. In this model, a Compressor block increases the loudness and
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