Use MATLAB Compiler™ to create a standalone application from a MATLAB function. The function implements an audio processing algorithm and plays the result through your audio output
Generate a standalone executable for parametric equalization using MATLAB Coder™ and use it on an audio file. multibandParametricEQ is used for the equalization algorithm. The example
The following tutorial walks through the steps of loading and configuring an external audio plugin in parameter display mode.
Accelerate a real-time audio application using C code generation with MATLAB® Coder™. You must have the MATLAB Coder™ software installed to run this example.
The following tutorial walks through the steps of loading and configuring an external audio plugin in property display mode.
Measure total harmonic distortion and noise level of audio input and output devices.
Presents a utility that can be used to analyze the timing performance of signal processing algorithms designed for real-time streaming applications.
Measure the latency of an audio device. The example uses audioLatencyMeasurementExampleApp which in turn uses audioPlayerRecorder along with a test signal and cross correlation to
Examine the Audio Device Writer block in a Simulink® model and specify a nondefault channel mapping.
Examine the Audio Device Reader block in a Simulink® model, modify parameters, and explore overrun.
The impulse response (IR) is an important tool for characterizing or representing a linear time-invariant (LTI) system. The impulseResponseMeasurer enables you to measure and capture
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.
Use the Parametric EQ Filter block and the multibandParametricEQ System object™ from the Audio System Toolbox™ to implement a parametric audio equalizer model. You can run the model on your
Use the Crossover Filter block and compressor System object™ from the Audio System toolbox™ to implement a multiband dynamic range compressor model. You can run the model on your host
Use System objects™ from Audio System Toolbox™ to implement echo and reverberation effects in a Simulink® model. You can run the model on your host computer or deploy it to an Apple iOS device.
Use the Levinson-Durbin and Time-Varying Lattice Filter blocks for low-bandwidth transmission of speech using linear predictive coding.
This model uses if-else block signal routing to replace regions of no speech with zeros.
This model detects voice activity using a frequency-domain audio signal.
Demonstrates a machine learning approach to identify people based on features extracted from recorded speech. The features used to train the classifier are: pitch of the voiced segments of
Generate a stereo signal from a multichannel audio signal using matrix encoding, and how to recover the original channels from the stereo mix using matrix decoding. This example
Acquire and process live multichannel audio. It also presents a simple algorithm for estimating the Direction Of Arrival (DOA) of a sound source using multiple microphone pairs within a
Design octave-band and fractional octave-band filters. Octave-band and fractional-octave-band filters are commonly used in acoustics. For example, octave filters are used to perform
Demonstrates how to measure sound pressure levels of octave frequency bands. A user interface (UI) enables you to experiment with various parameters while the measurement is displayed.
Obtain designs for the most common weighting filters - A-weighting, C-weighting, C-message, ITU-T 0.41, and ITU-R 468-4 - using the weightingFilter System object and audio weighting
Several basic aspects of audio signal positioning. The listener occupies a location in the center of a circle, and the position of the sound source is varied so that it remains within the
Use tools from Audio System Toolbox (TM) to measure loudness, loudness range, and true-peak value. It also shows how to normalize audio to meet the EBU R 128 standard compliance.
Illustrates microphone array beamforming to extract desired speech signals in an interference-dominant, noisy environment. Such operations are useful to enhance speech signal quality
This examples shows how to create ambisonic plugins using MATLAB higher order ambisonic (HOA) demo functions. Ambisonics is a spatial audio technique which represents a
Decode ambisonic audio into binaural audio using virtual loudspeakers. A virtual loudspeaker is a sound source positioned on the surface of a sphere, with the listener located at the center
Visualize the magnitude response of a tunable filter. The filters in this example are implemented as audio plugins. This example uses the visualize and audioTestBench functionality of the