reverberator

Add reverberation to audio signal

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Description

The reverberator System object™ adds reverberation to mono or stereo audio signals.

To add reverberation to your input:

Create the reverberator object and set its properties.
Call the object with arguments, as if it were a function.

To learn more about how System objects work, see What Are System Objects?

Creation

Syntax

reverb = reverberator

reverb = reverberator(Name,Value)

Description

reverb = reverberator creates a System object, reverb, that adds artificial reverberation to an audio signal.

reverb = reverberator(Name,Value) sets each property Name to the specified Value. Unspecified properties have default values.

Example: reverb = reverberator('PreDelay',0.5,'WetDryMix',1) creates a System object, reverb, with a 0.5 second pre-delay and a wet-to-dry mix ratio of one.

Properties

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Unless otherwise indicated, properties are nontunable, which means you cannot change their values after calling the object. Objects lock when you call them, and the release function unlocks them.

If a property is tunable, you can change its value at any time.

For more information on changing property values, see System Design in MATLAB Using System Objects.

`PreDelay` — Pre-delay for reverberation (s)
`0` (default) | real positive scalar

Pre-delay for reverberation in seconds, specified as a real scalar in the range [0, 1].

Pre-delay for reverberation is the time between hearing direct sound and the first early reflection. The value of PreDelay is proportional to the size of the room being modeled.

Tunable: Yes

Data Types: single | double

`HighCutFrequency` — Lowpass filter cutoff (Hz)
`20000` (default) | real positive scalar

Lowpass filter cutoff in Hz, specified as a real positive scalar in the range 0 to $(\frac{S a m p l e R a t e}{2})$ .

Lowpass filter cutoff is the –3 dB cutoff frequency for the single-pole lowpass filter at the front of the reverberator structure. It prevents the application of reverberation to high-frequency components of the input.

Tunable: Yes

Data Types: single | double

`Diffusion` — Density of reverb tail
`0.5` (default) | real scalar

Density of reverb tail, specified as a real positive scalar in the range [0, 1].

Diffusion is proportional to the rate at which the reverb tail builds in density. Increasing Diffusion pushes the reflections closer together, thickening the sound. Reducing Diffusion creates more discrete echoes.

Tunable: Yes

Data Types: single | double

`DecayFactor` — Decay factor of reverb tail
`0.5` (default) | real scalar

Decay factor of reverb tail, specified as a real positive scalar in the range [0, 1].

DecayFactor is inversely proportional to the time it takes for reflections to run out of energy. To model a large room, use a long reverb tail (low decay factor). To model a small room, use a short reverb tail (high decay factor).

Tunable: Yes

Data Types: single | double

`HighFrequencyDamping` — High-frequency damping
`0.0005` (default) | real scalar

High-frequency damping, specified as a real positive scalar in the range [0, 1].

HighFrequencyDamping is proportional to the attenuation of high frequencies in the reverberation output. Setting HighFrequencyDamping to a large value makes high-frequency reflections decay faster than low-frequency reflections.

Tunable: Yes

Data Types: single | double

`WetDryMix` — Wet-dry mix
`0.3` (default) | real scalar

Wet-dry mix, specified as a real positive scalar in the range [0, 1].

Wet-dry mix is the ratio of wet (reverberated) to dry (original) signal that your reverberator System object outputs.

Tunable: Yes

Data Types: single | double

`SampleRate` — Input sample rate (Hz)
`44100` (default) | positive scalar

Input sample rate in Hz, specified as a positive scalar.

Tunable: Yes

Data Types: single | double

Usage

Syntax

audioOut = reverb(audioIn)

Description

audioOut = reverb(audioIn) adds reverberation to the input signal, audioIn, and returns the mixed signal, audioOut. The type of reverberation is specified by the algorithm and properties of the reverberator System object, reverb.

example

Input Arguments

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`audioIn` — Audio input to reverberator
column vector | N-by-2 matrix

Audio input to the reverberator, specified as a column vector or two-column matrix. The columns of the matrix are treated as independent audio channels.

Data Types: single | double

Output Arguments

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`audioOut` — Audio output from reverberator
N-by-2 matrix

Audio output from the reverberator, returned as a two-column matrix.

Data Types: single | double

Object Functions

To use an object function, specify the System object as the first input argument. For example, to release system resources of a System object named obj, use this syntax:

release(obj)

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Specific to reverberator

`createAudioPluginClass`	Create audio plugin class that implements functionality of System object
`parameterTuner`	Tune object parameters while streaming

MIDI

`configureMIDI`	Configure MIDI connections between audio object and MIDI controller
`disconnectMIDI`	Disconnect MIDI controls from audio object
`getMIDIConnections`	Get MIDI connections of audio object

Common to All System Objects

`clone`	Create duplicate System object
`isLocked`	Determine if System object is in use
`release`	Release resources and allow changes to System object property values and input characteristics
`reset`	Reset internal states of System object
`step`	Run System object algorithm

Examples

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Add Reverberation to Audio Signal

Open Live Script

Use the reverberator System object™ to add artificial reverberation to an audio signal read from a file.

Create the dsp.AudioFileReader and audioDeviceWriter System objects. Use the sample rate of the reader as the sample rate of the writer.

fileReader = dsp.AudioFileReader('FunkyDrums-44p1-stereo-25secs.mp3','SamplesPerFrame',1024);
deviceWriter = audioDeviceWriter('SampleRate',fileReader.SampleRate);

Play 10 seconds of the audio signal through your device.

tic
while toc < 10
    audio = fileReader();
    deviceWriter(audio);
end
release(fileReader)

Construct a reverberator System object with default settings.

reverb = reverberator

reverb = 
  reverberator with properties:

                PreDelay: 0
        HighCutFrequency: 20000
               Diffusion: 0.5000
             DecayFactor: 0.5000
    HighFrequencyDamping: 5.0000e-04
               WetDryMix: 0.3000
              SampleRate: 44100

Construct a time scope to visualize the original audio signal and the audio signal with added artificial reverberation.

scope = timescope( ...
    'SampleRate',fileReader.SampleRate,...
    'TimeSpanOverrunAction','Scroll',...
    'TimeSpanSource','property',...
    'TimeSpan',3,...
    'BufferLength',3*fileReader.SampleRate*2, ...
    'YLimits',[-1,1],...
    'ShowGrid',true, ...
    'ShowLegend',true, ...
    'Title','Audio with Reverberation vs. Original');

Play the audio signal with artificial reverberation. Visualize the audio with reverberation and the original audio.

while ~isDone(fileReader)
    audio = fileReader();
    audioWithReverb = reverb(audio);
    deviceWriter(audioWithReverb);
    scope([audioWithReverb(:,1),audio(:,1)])
end
release(fileReader)
release(deviceWriter)
release(scope)

Tune Reverberator Parameters

Open Live Script

Create a dsp.AudioFileReader to read in audio frame-by-frame. Create an audioDeviceWriter to write audio to your sound card. Create a reverberator to process the audio data.

frameLength = 1024;
fileReader = dsp.AudioFileReader('RockDrums-44p1-stereo-11secs.mp3', ...
    'SamplesPerFrame',frameLength,'PlayCount',2);
deviceWriter = audioDeviceWriter('SampleRate',fileReader.SampleRate);
reverb = reverberator('SampleRate',fileReader.SampleRate);

Call parameterTuner to open a UI to tune parameters of the octaveFilter while streaming.

parameterTuner(reverb)

In an audio stream loop:

Read in a frame of audio from the file.
Apply reverberation.
Write the frame of audio to your audio device for listening.

While streaming, tune parameters of the reverberator and listen to the effect.

while ~isDone(fileReader)
    audioIn = fileReader();
    audioOut = reverb(audioIn);
    deviceWriter(audioOut);
    drawnow limitrate % required to update parameter
end

As a best practice, release your objects once done.

release(deviceWriter)
release(fileReader)
release(reverb)

Tips

The createAudioPluginClass and configureMIDI functions map tunable properties of the compressor to user-facing parameters:

Property	Range	Mapping	Unit
`PreDelay`	[0, 1]	linear	s
`HighCutFrequency`	[20, 20000]	log	Hz
`Diffusion`	[0, 1]	linear	none
`DecayFactor`	[0, 1]	linear	none
`HighFrequencyDamping`	[0, 1]	linear	none
`WetDryMix`	[0, 1]	linear	none

Algorithms

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The algorithm to add reverberation follows the plate-class reverberation topology described in [1] and is based on a 29,761 Hz sample rate.

The algorithm has five stages.

The description for the algorithm that follows is for a stereo input. A mono input is a simplified case.

Stereo-to-Mono

A stereo signal is converted to a mono signal: $x [n] = 0.5 \times (x_{R} [n] + x_{L} [n])$ .

Preconditioning

A delay followed by a lowpass filter preconditions the mono signal.

The pre-delay output is determined as $x_{p} [n] = x [n - k]$ , where the PreDelay property determines the value of k.
The signal is fed through a single-pole lowpass filter with transfer function
$L P (z) = \frac{1 - α}{1 - α z^{- 1}},$
where
$α = \exp (- 2 π \times \frac{f_{c}}{f_{s}}) .$
- f_c is the cutoff frequency specified by the HighCutFrequency property.
- f_s is the sampling frequency specified by the SampleRate property.

Decorrelation

The signal is decorrelated by passing through a series of four allpass filters.

The allpass filters are of the form

$A P (z) = \frac{β + z^{- k}}{1 + β z^{- k}},$

where β is the coefficient specified by the Diffusion property and k is the delay as follows:

For AP₁, k = 142.
For AP₂, k = 107.
For AP₃, k = 379.
For AP₄, k = 277.

Tank

The signal is fed into the tank, where it circulates to simulate the decay of a reverberation tail.

The following description tracks the signal as it progresses through the top of the tank. The signal progression through the bottom of the tank follows the same pattern, with different delay specifications.

The new signal enters the top of the tank and is added to the circulated signal from the bottom of the tank.
The signal passes through a modulated allpass filter:
$M o d u l a t e d A P_{1} (z) = \frac{- β + z^{- k}}{1 - β z^{- k}}$
- β is the coefficient specified by the Diffusion property.
- k is the variable delay specified by a 1 Hz sinusoid with amplitude = (8/29761) * SampleRate. To account for fractional delay resulting from the modulating k, allpass interpolation is used [2].
The signal is delayed again, and then passes through a lowpass filter:

$L P_{2} (z) = \frac{1 - φ}{1 - φ z^{- 1}}$
- φ is the coefficient specified by the HighFrequencyDamping property.
The signal is multiplied by a gain specified by the DecayFactor property. The signal then passes through an allpass filter:
$A P_{5} (z) = \frac{β + z^{- k}}{1 + β z^{- k}} .$
- β is the coefficient specified by the Diffusion property.
- k is set to 1800 for the top of the tank and 2656 for the bottom of the tank.
The signal is delayed again and then circulated to the bottom half of the tank for the next iteration.

A similar pattern is executed in parallel for the bottom half of the tank. The output of the tank is calculated as the signed sum of delay lines picked off at various points from the tank. The summed output is multiplied by 0.6.

Wet/Dry Mix

The wet (processed) signal is then added to the dry (original) signal:

$y_{R} [n] = (1 - κ) x_{R} [n] + κ x_{3 R} [n],$

$y_{L} [n] = (1 - κ) x_{L} [n] + κ x_{3 L} [n],$

where the WetDryMix property determines κ.

References

[1] Dattorro, Jon. "Effect Design, Part 1: Reverberator and Other Filters." Journal of the Audio Engineering Society. Vol. 45, Issue 9, 1997, pp. 660–684.

[2] Dattorro, Jon. "Effect Design, Part 2: Delay-Line Modulation and Chorus." Journal of the Audio Engineering Society. Vol. 45, Issue 10, 1997, pp. 764–788.

Extended Capabilities

C/C++ Code Generation
Generate C and C++ code using MATLAB® Coder™.

Usage notes and limitations:

System Objects in MATLAB Code Generation (MATLAB Coder)

Version History

Introduced in R2016a

reverberator

Description

Creation

Syntax

Description

Properties

PreDelay — Pre-delay for reverberation (s) 0 (default) | real positive scalar

HighCutFrequency — Lowpass filter cutoff (Hz) 20000 (default) | real positive scalar

Diffusion — Density of reverb tail 0.5 (default) | real scalar

DecayFactor — Decay factor of reverb tail 0.5 (default) | real scalar

HighFrequencyDamping — High-frequency damping 0.0005 (default) | real scalar

WetDryMix — Wet-dry mix 0.3 (default) | real scalar

SampleRate — Input sample rate (Hz) 44100 (default) | positive scalar

Usage

Syntax

Description

Input Arguments

audioIn — Audio input to reverberator column vector | N-by-2 matrix

Output Arguments

audioOut — Audio output from reverberator N-by-2 matrix

Object Functions

Specific to reverberator

MIDI

Common to All System Objects

Examples

Add Reverberation to Audio Signal

Tune Reverberator Parameters

Tips

Algorithms

Stereo-to-Mono

Preconditioning

Decorrelation

Tank

Wet/Dry Mix

References

Extended Capabilities

C/C++ Code Generation Generate C and C++ code using MATLAB® Coder™.

Version History

See Also

`PreDelay` — Pre-delay for reverberation (s)
`0` (default) | real positive scalar

`HighCutFrequency` — Lowpass filter cutoff (Hz)
`20000` (default) | real positive scalar

`Diffusion` — Density of reverb tail
`0.5` (default) | real scalar

`DecayFactor` — Decay factor of reverb tail
`0.5` (default) | real scalar

`HighFrequencyDamping` — High-frequency damping
`0.0005` (default) | real scalar

`WetDryMix` — Wet-dry mix
`0.3` (default) | real scalar

`SampleRate` — Input sample rate (Hz)
`44100` (default) | positive scalar

`audioIn` — Audio input to reverberator
column vector | N-by-2 matrix

`audioOut` — Audio output from reverberator
N-by-2 matrix

C/C++ Code Generation
Generate C and C++ code using MATLAB® Coder™.