# splMeter

Measure sound pressure level of audio signal

## Description

The `splMeter`

System object™ computes sound pressure level measurements. The object returns measurements for:

frequency-weighted sound levels

fast or slow time-weighted sound levels

equivalent-continuous sound levels

peak sound levels

maximum sound levels

To implement SPL metering:

Create the

`splMeter`

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

### Description

`SPL = splMeter`

creates a System object, `SPL`

, that performs SPL metering.

`SPL = splMeter(`

sets each
property `Name,Value`

)`Name`

to the specified `Value`

. Unspecified
properties have default values.

**Example: **```
SPL =
splMeter('FrequencyWeighting','C-weighting','SampleRate',12000)
```

creates a System object, `SPL`

, that performs C-weighting and operates at 12
kHz.

## Properties

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.

`Bandwidth`

— Width of analysis bands

`'Full band'`

(default) | `'1 octave'`

| `'2/3 octave'`

| `'1/3 octave'`

Width of analysis bands, specified as `'Full band'`

, ```
'1
octave'
```

, `'2/3 octave'`

, or ```
'1/3
octave'
```

. If `Bandwidth`

is specified as ```
'Full
band'
```

, the SPL meter returns one set of measurements for the whole frequency
band. If `Bandwidth`

is specified as `'1 octave'`

,
`'2/3 octave'`

, or `'1/3 octave'`

, the SPL meter
returns one set of measurements per octave or fractional-octave band.

**Tunable: **No

**Data Types: **`char`

| `string`

`FrequencyRange`

— Frequency range of filter bank (Hz)

`[22 22050]`

(default) | two-element row vector of positive monotonically increasing values

Frequency range of the filter bank in Hz, specified as a two-element row vector of
positive monotonically increasing values. Frequency bands centered above
`SampleRate`

/2 are excluded.

**Tunable: **No

#### Dependencies

To enable this property, set `Bandwidth`

to ```
'1
octave'
```

, `'2/3 octave'`

, or ```
'1/3
octave'
```

.

**Data Types: **`single`

| `double`

| `int8`

| `int16`

| `int32`

| `int64`

| `uint8`

| `uint16`

| `uint32`

| `uint64`

`OctaveFilterOrder`

— Order of octave filter

`2`

(default) | even integer

Order of the octave filter, specified as an even integer.

**Tunable: **No

#### Dependencies

To enable this property, set `Bandwidth`

to ```
'1
octave'
```

, `'2/3 octave'`

, or ```
'1/3
octave'
```

.

**Data Types: **`single`

| `double`

`FrequencyWeighting`

— Frequency weighting applied to input

`'A-weighting'`

(default) | `'C-weighting'`

| `'Z-weighting'`

(no weighting)

Frequency weighting applied to input, specified as `'A-weighting'`

,
`'C-weighting'`

, or `'Z-weighting'`

, where
Z-weighting corresponds to no weighting. The frequency weighting is designed and
implemented using the `weightingFilter`

System object.

**Tunable: **No

**Data Types: **`char`

| `string`

`TimeWeighting`

— Time weighting (s)

`'Fast'`

(default) | `'Slow'`

Time weighting, in seconds, for calculation of time-weighted sound level and maximum
time-weighted sound level, specified as `'Fast'`

or
`'Slow'`

. The `TimeWeighting`

property is used to
specify the coefficient of a lowpass filter.

`'Fast'`

– 1/8`'Slow'`

– 1

**Tunable: **Yes

**Data Types: **`char`

| `string`

`PressureReference`

— Reference pressure for dB calculations (Pa)

`2e-5`

(default) | positive scalar

Reference pressure for dB calculations in Pa, specified as a positive scalar.

**Tunable: **Yes

**Data Types: **`single`

| `double`

`TimeInterval`

— Time interval for reporting level measurements (s)

`1`

(default) | positive scalar

Time interval, in seconds, to report equivalent-continuous, peak, and maximum time-weighted sound levels, specified as a positive scalar integer.

**Tunable: **No

**Data Types: **`single`

| `double`

`CalibrationFactor`

— Calibration factor multiplied by input

1 (default) | positive finite scalar or vector

Scalar (mono input) or vector (multichannel input) calibration factor multiplied by input.

To set the calibration factor using a reference tone, use `calibrate`

.

**Tunable: **No

**Data Types: **`single`

| `double`

`SampleRate`

— Input sample rate (Hz)

`44100`

(default) | positive scalar

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

**Tunable: **No

**Data Types: **`single`

| `double`

## Usage

### Description

### Input Arguments

`audioIn`

— Audio input to SPL meter

column vector | matrix

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

**Data Types: **`single`

| `double`

### Output Arguments

`Lt`

— Time-weighted sound level (dB)

column vector | matrix | 3-D array

Time-weighted sound level in dB, returned as a column vector, matrix, or 3-D array
the same type as `audioIn`

.

Size and interpretation of the outputs depend on what the Bandwidth property is set to:

`'Full band'`

(default) ––`Lt`

,`Leq`

,`Lpeak`

, and`Lmax`

are returned as column vectors or matrices the same size as`audioIn`

.`'1 octave'`

,`'2/3 octave'`

, or`'1/3 octave'`

––`Lt`

,`Leq`

,`Lpeak`

, and`Lmax`

are returned as*L*-by-*B*-by-*C*arrays.*L*–– Number of rows in`audioIn`

*B*–– Number of octave bands*C*–– Number of columns in`audioIn`

**Data Types: **`single`

| `double`

`Leq`

— Equivalent-continuous sound level (dB)

column vector | matrix | 3-D array

Equivalent-continuous sound level in dB, returned as a column vector, matrix, or
3-D array the same type as `audioIn`

.

Size and interpretation of the outputs depend on what the Bandwidth property is set to:

`'Full band'`

(default) ––`Lt`

,`Leq`

,`Lpeak`

, and`Lmax`

are returned as column vectors or matrices the same size as`audioIn`

.`'1 octave'`

,`'2/3 octave'`

, or`'1/3 octave'`

––`Lt`

,`Leq`

,`Lpeak`

, and`Lmax`

are returned as*L*-by-*B*-by-*C*arrays.*L*–– Number of rows in`audioIn`

*B*–– Number of octave bands*C*–– Number of columns in`audioIn`

**Data Types: **`single`

| `double`

`Lpeak`

— Peak sound level (dB)

column vector | matrix | 3-D array

Peak sound level in dB, returned as a column vector, matrix, or 3-D array the same
type as `audioIn`

.

Size and interpretation of the outputs depend on what the Bandwidth property is set to:

`'Full band'`

(default) ––`Lt`

,`Leq`

,`Lpeak`

, and`Lmax`

are returned as column vectors or matrices the same size as`audioIn`

.`'1 octave'`

,`'2/3 octave'`

, or`'1/3 octave'`

––`Lt`

,`Leq`

,`Lpeak`

, and`Lmax`

are returned as*L*-by-*B*-by-*C*arrays.*L*–– Number of rows in`audioIn`

*B*–– Number of octave bands*C*–– Number of columns in`audioIn`

**Data Types: **`single`

| `double`

`Lmax`

— Maximum time-weighted sound level (dB)

column vector | matrix | 3-D array

Maximum time-weighted sound level in dB, returned as a column vector, matrix, or
3-D array the same type as `audioIn`

.

Size and interpretation of the outputs depend on what the Bandwidth property is set to:

`'Full band'`

(default) ––`Lt`

,`Leq`

,`Lpeak`

, and`Lmax`

are returned as column vectors or matrices the same size as`audioIn`

.`'1 octave'`

,`'2/3 octave'`

, or`'1/3 octave'`

––`Lt`

,`Leq`

,`Lpeak`

, and`Lmax`

are returned as*L*-by-*B*-by-*C*arrays.*L*–– Number of rows in`audioIn`

*B*–– Number of octave bands*C*–– Number of columns in`audioIn`

**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)

### Specific to splMeter

`calibrate` | Calibrate meter using calibration tone with known level |

`getCenterFrequencies` | Center frequencies of filters |

## Examples

### Measure SPL of Audio Signal

Use the `splMeter`

System object™ to measure the A-weighted sound pressure level of a streaming audio signal. Specify a two second time-interval for reporting and a fast time-weighting. Visualize the SPL measurements using the `timescope`

object.

Create a `dsp.AudioFileReader`

object to read in an audio file frame by frame. Create an `audioDeviceWriter`

object to listen to the audio signal. Create a `timescope`

object to visualize SPL measurements. Create an `splMeter`

to measure the sound pressure level of the audio file. Use the default calibration factor of 1.

source = dsp.AudioFileReader('Ambiance-16-44p1-mono-12secs.wav'); fs = source.SampleRate; player = audioDeviceWriter('SampleRate',fs); scope = timescope('SampleRate',fs, ... 'TimeSpanOverrunAction','Scroll', ... 'TimeSpanSource','Property','TimeSpan',3,'ShowGrid',true, ... 'YLimits',[20 110],'AxesScaling','Auto', ... 'ShowLegend',true,'BufferLength',4*3*fs, ... 'ChannelNames', ... {'Lt_AF','Leq_A','Lpeak_A','Lmax_AF'}, ... 'Name','Sound Pressure Level Meter'); SPL = splMeter('TimeWeighting','Fast', ... 'FrequencyWeighting','A-weighting', ... 'SampleRate',fs, ... 'TimeInterval',2);

In an audio stream loop:

Read in the audio signal frame.

Play the audio signal to your output device.

Call the SPL meter to return the time-weighted, equivalent-continuous, peak, and maximum time-weighted sound levels in dB.

Display the sound levels using the scope.

As a best practice, release your objects once complete.

while ~isDone(source) x = source(); player(x); [Lt,Leq,Lpeak,Lmax] = SPL(x); scope([Lt,Leq,Lpeak,Lmax]) end release(source) release(player) release(SPL) release(scope)

### Octave SPL Metering

The `splMeter`

enables you to monitor sound pressure level for octave and fractional-octave bands. In this example, you monitor the equivalent-continuous sound pressure level of 1/3-octave bands.

Create a `dsp.AudioFileReader`

object to read in an audio file frame by frame. Create an `audioDeviceWriter`

object so you can listen to the audio signal. Create an `splMeter`

to measure the octave sound pressure level of the audio file. Use the default calibration factor of 1. Create a `dsp.ArrayPlot`

object to visualize the equivalent-continuous SPL for each octave band.

source = dsp.AudioFileReader('JetAirplane-16-11p025-mono-16secs.wav'); fs = source.SampleRate; player = audioDeviceWriter('SampleRate',fs); SPL = splMeter( ... 'Bandwidth','1/3 octave', ... 'SampleRate',fs); centerFrequencies = getCenterFrequencies(SPL); scope = dsp.ArrayPlot(... 'XDataMode','Custom', ... 'CustomXData',centerFrequencies, ... 'XLabel','Octave Band Center Frequencies (Hz)', ... 'YLabel','Equivalent-Continuous Sound Level (dB)', ... 'YLimits',[20 90], ... 'ShowGrid',true, ... 'Name','Sound Pressure Level Meter');

In an audio stream loop:

Read in the audio signal frame.

Play the audio signal to your output device.

Call the SPL meter to return the equivalent-continuous sound pressure level in dB.

Display the sound levels using the scope. Update the scope only when the equivalent-continuous sound pressure level has changed.

As a best practice, release your objects once complete.

LeqPrevious = zeros(size(centerFrequencies)); while ~isDone(source) x = source(); player(x); [~,Leq] = SPL(x); for i = 1:size(Leq,1) if LeqPrevious ~= Leq(i,:) scope(Leq(i,:)') LeqPrevious = Leq(i,:); end end end release(source) release(player) release(SPL) release(scope)

## Algorithms

Sound pressure level calculations follow the algorithms described in [1]. You can specify property values to conform to standards [2] and [3].

### Calibration

To account for environmental and input device effects in SPL measurements, the audio input is multiplied by a calibration factor:

$$x=\text{audioIn}\times \text{CalibrationFactor}$$

The `CalibrationFactor`

property can be set directly, or by using the
`calibrate`

function, which compares a known level with acquired data. The known level is determined
using a physical calibrator.

### Frequency Weighting

A-, C-, or Z-frequency weighting is applied. The frequency weighting is implemented
using the `weightingFilter`

System object.

### Analysis Bands

If you specify the `Bandwidth`

property as ```
'1
octave'
```

, `'2/3 octave'`

or `'1/3 octave'`

,
then the SPL calculations are applied to each octave or fractional-octave band. These
analysis bands are determined after frequency weighting.

### Time-Weighted Sound Level

Time-weighted sound level is defined as the ratio of the time-weighted root mean squared sound pressure to the reference sound pressure, converted to dB. That is,

$$\begin{array}{l}Lt=10{\mathrm{log}}_{10}\left\{\frac{\left(\raisebox{1ex}{$1$}\!\left/ \!\raisebox{-1ex}{$\tau $}\right.\right){\displaystyle {\int}_{{t}_{s}}^{t}y{(\xi )}^{2}{e}^{-\left(t-\xi \right)/\tau}d\xi}}{{p}_{o}^{2}}\right\}\\ \text{\hspace{0.17em}}\text{\hspace{0.17em}}\text{\hspace{0.17em}}=10{\mathrm{log}}_{10}\left\{\frac{h({y}^{2})}{{p}_{o}^{2}}\right\}\end{array}$$

*h*(*y*^{2}) can be
interpreted as the convolution of *y*^{2} with a
filter with impulse response $$\left(\raisebox{1ex}{$1$}\!\left/ \!\raisebox{-1ex}{$\tau $}\right.\right){e}^{-\raisebox{1ex}{$t$}\!\left/ \!\raisebox{-1ex}{$\tau $}\right.}$$. *y* is the output of the frequency-weighting filter. The
impulse response corresponds to a lowpass filter of the form $$H\left(s\right)=\frac{\raisebox{1ex}{$1$}\!\left/ \!\raisebox{-1ex}{$\tau $}\right.}{s+\raisebox{1ex}{$1$}\!\left/ \!\raisebox{-1ex}{$\tau $}\right.}$$. Using impulse invariance, the discrete filter can be interpreted
as,

$$H\left(z\right)=\frac{\raisebox{1ex}{$1$}\!\left/ \!\raisebox{-1ex}{$\left(\tau \times fs\right)$}\right.}{1-{e}^{-\raisebox{1ex}{$1$}\!\left/ \!\raisebox{-1ex}{$\left(\tau \times fs\right)$}\right.}{z}^{-1}}\text{\hspace{0.17em}}.$$

*τ*is specified by the time-weighting coefficient as 0.125 (if`TimeWeighting`

is set to`'Fast'`

) or 1 (if`TimeWeighting`

is set to`'Slow'`

).*fs*is the sample rate specified by the`SampleRate`

property.

### Equivalent-Continuous Sound Level

Equivalent-continuous sound level is also called time-average sound level. It is defined as the ratio of root mean squared sound pressure to the reference sound pressure, converted to dB. That is,

$$\begin{array}{l}Leq=10{\mathrm{log}}_{10}\left\{\frac{\left(\raisebox{1ex}{$1$}\!\left/ \!\raisebox{-1ex}{$T$}\right.\right){\displaystyle {\int}_{{t}_{1}}^{{t}_{2}}{y}^{2}dt}}{{p}_{o}^{2}}\right\}\\ \text{\hspace{1em}}=20{\mathrm{log}}_{10}\left(\text{rms}\left(y\right)/{p}_{\text{o}}\right)\end{array}$$

where

*y*is the output of the frequency-weighting filter.*p*_{o}is the reference sound pressure, specified by the`PressureReference`

property.

### Peak Sound Level

Peak sound level is defined as the ratio of peak sound pressure to the reference sound pressure, converted to dB. That is,

$$Lpeak=20{\mathrm{log}}_{10}\left(\mathrm{max}\left(\left|y\right|\right)/{p}_{o}\right)$$

where

*y*is the output of the frequency-weighting filter.*p*_{o}is the reference sound pressure, specified by the`PressureReference`

property.

### Max Time-Weighted Sound Level

Maximum time-weighted sound level is defined as the greatest time-weighted sound level within a stated time interval.

## References

[1] Harris, Cyril M. *Handbook of
Acoustical Measurements and Noise Control*. 3rd ed. American Institute of
Physics, 1998.

[2] International Electrotechnical Commission. Electroacoustics - Sound level meters - Part 1: Specifications. IEC 61672-1:2013.

[3] American National Standards Institute. ANSI S1.4: Specification for Sound Level Meters. 1983.

## 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 R2018a**

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