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.
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 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
Apply weighting filters to measure sound pressure. A user interface (UI) allows the user to control various parameters while the simulation is running.
Examine the Audio Device Writer block in a Simulink® model and specify a nondefault channel mapping.
Examine the Audio Device Writer block in a Simulink® model, determine underrun, and decrease underrun.
Communicate between a digital audio workstation (DAW) and MATLAB using the user datagram protocol (UDP). The information shared between the DAW and MATLAB can used to perform
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
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.
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
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.
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
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.
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
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
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.
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.
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
Use the Levinson-Durbin and Time-Varying Lattice Filter blocks for low-bandwidth transmission of speech using linear predictive coding.
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.
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.
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.
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
Extract an audio source from a stereo mix based on its panning coefficient. This example illustrates MATLAB® and Simulink® implementations.
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