Evaluate Acoustic Metrics

Select the correct Audio Toolbox function for acoustic measurement, apply proper calibration, and choose appropriate standards — instead of reimplementing algorithms or calling nonexistent functions.

When to Use

• Computing psychoacoustic metrics (loudness, sharpness, roughness, fluctuation)
• Evaluating speech quality or intelligibility (ViSQOL, STOI, STI, SII)
• Assessing noise compliance (NC rating, RC rating, RNC, SIL)
• Measuring SPL from recordings (offline, calibrated)
• Computing broadcast loudness (EBU R 128 / LUFS)
• Measuring reverberation time (RT60) from impulse responses
• Detecting tonal content (tone-to-noise ratio, prominence ratio)
• Converting microphone sensitivity to CalibrationFactor
• User mentions PESQ or POLQA (redirect to correct alternatives)

When NOT to Use

• Real-time streaming audio processing loops — use matlab-process-streaming-audio
• Audio device I/O, recording, or live calibrateMicrophone — use matlab-play-record-audio
• Audio plugin authoring (VST/AU) — use matlab-write-audio-plugin
• General filter design unrelated to measurement — use matlab-design-digital-filter
• Spatial audio / HRTF / SOFA processing

Workflow

Follow this decision tree to route the user's goal to the right function:

1. Identify the measurement goal

Goal	Category	Go to
"How loud is it?" (perceived)	Psychoacoustic	→ Psychoacoustic Metrics
"Does it comply with broadcast loudness?"	Broadcast	→ Broadcast Loudness
"Is the noise level acceptable for this space?"	Noise compliance	→ Noise Compliance
"Can people understand speech in this room?"	Speech intelligibility	→ STI/SII Measurement
"How good does processed speech sound?"	Speech quality	→ Speech Quality
"What's the SPL?"	Level measurement	→ SPL Measurement
"What's the reverberation time?"	Room acoustics	→ RT60
"Is there a tonal component?"	Tonality	→ Tonality

2. Ask before computing

Before writing code, confirm these with the user:

• Calibration: Do they have a CalibrationFactor? Did they calibrate with calibrateMicrophone? (See Calibration section)
• Standard: Which standard or method applies? (ISO 532-1 vs 532-2, NC vs RC vs RNC)
• Sound field: Free field or diffuse field recording?
• Application context: What space type? What's the concern? (Drives NC vs RC vs RNC selection)

3. Select function and apply

Use the Key Functions table and the category-specific patterns below.

Functions That Do NOT Exist

Function	Status	Use Instead
pesq	Does NOT exist (proprietary ITU-T P.862 license)	visqol(degraded, reference, fs, Mode="speech")
polqa	Does NOT exist (proprietary ITU-T P.863 license)	visqol(degraded, reference, fs)

Never suggest pesq or polqa. They are not in Audio Toolbox and cannot be added due to licensing.

Key Functions

Function	Purpose	Standard	Available From
acousticLoudness	Perceived loudness (sones)	ISO 532-1/2	R2020a
acousticSharpness	Perceived sharpness (acum)	DIN 45692	R2020a
acousticRoughness	Perceived roughness (asper)	—	R2021a
acousticFluctuation	Fluctuation strength (vacil)	—	R2020b
integratedLoudness	Broadcast loudness (LUFS)	EBU R 128 / ITU-R BS.1770	R2016b
loudnessMeter	Streaming broadcast loudness meter	EBU R 128 / ITU-R BS.1770	R2018a
visqol	Full-reference audio quality (MOS)	—	R2024a
stoi	Short-time objective intelligibility	—	R2024a
mnru	Modulated noise reference unit	ITU-T P.810	R2024a
dBov	Power level relative to overload (dB)	ITU-T G.100.1	R2024a
speechTransmissionIndex	STI from IR or test signals	IEC 60268-16	R2026a
speechIntelligibilityIndex	SII from IR, test signals, or levels	ANSI/ASA S3.5-1997	R2026b
stipaExcitation	Generate STI/STIPA test signals	IEC 60268-16	R2026a
siiExcitation	Generate SII test signals	ANSI/ASA S3.5-1997	R2026b
noiseCriteria	NC rating + SIL + spectrum imbalance + rattle risk	ANSI/ASA S12.2	R2026b
roomNoiseCriteria	RNC rating (detects surging/fluctuations)	ANSI/ASA S12.2	R2026b
roomCriteria	RC Mark II rating (HVAC systems)	ANSI/ASA S12.2	R2026b
noiseCriteriaCurves	Look up NC curve values	ANSI/ASA S12.2	R2026b
noiseCriteriaRecommendations	Recommended NC levels by room type	ANSI/ASA S12.2	R2026b
speechInterferenceLevel	SIL from recording	ANSI/ASA S12.2	R2026b
rt60	Reverberation time from impulse response	ISO 3382-1/2	R2025a
acousticToneToNoiseRatio	Detect/quantify tones	ECMA-418-1	R2023b
acousticProminenceRatio	Tonality via critical band comparison	ECMA-418-1	R2023b
splMeter	SPL measurement (time-weighted, per-band)	IEC 61672	R2018a
calibrateMicrophone	Compute calibration factor from known tone	—	R2018b
weightingFilter	A/C/Z/K frequency weighting	IEC 61672-1	R2016b
octaveFilter	Single octave-band filter	ANSI S1.11-2004	R2017b
octaveFilterBank	Multi-band octave filtering	ANSI S1.11-2004	R2019a

See references/function-arguments.md for complete argument lists, defaults, and valid values for each function — consult when constructing calls with non-default parameters.

Calibration

CalibrationFactor from microphone sensitivity

Convert dB sensitivity (dBV/Pa) to the linear CalibrationFactor:

sensitivity_dBV = -26;  % Example: -26 dBV/Pa microphone
calibrationFactor = 1 / 10^(sensitivity_dBV/20);

The CalibrationFactor is the 3rd positional argument for acousticLoudness, acousticSharpness, acousticRoughness, and acousticFluctuation (default: sqrt(8)). For noise compliance functions, pass it as a name-value: CalibrationFactor=value.

System-level calibration advice

Always ask the user about their signal chain. If the recording path includes a preamp, audio interface, or ADC with unknown gain, the mic sensitivity spec alone is insufficient. Recommend:

1. Play a 1 kHz tone through the system and record it through the full signal chain (mic → preamp → ADC → WAV). Use a pistonphone for a known 94 dB reference, or place a physical SPL meter next to the microphone to get an SPL reading.
2. Compute the system calibration factor: calibrationFactor = calibrateMicrophone(calRecording, fs, SPLreading, FrequencyWeighting="Z-weighting")
3. Pass that factor to the measurement function

calibrateMicrophone expects a 1 kHz tone recording as input. The SPLreading argument is the known SPL in dB (e.g., 94 from a pistonphone, or the value shown on a physical SPL meter).

Cross-reference: for the physical recording workflow with calibrateMicrophone, see the matlab-play-record-audio skill.

Patterns

Psychoacoustic Metrics

[audioIn, fs] = audioread("recording.wav");
calFactor = 1 / 10^(-26/20);

loudness = acousticLoudness(audioIn, fs, calFactor, ...
    Method="ISO 532-1", SoundField="diffuse", TimeVarying=true);

sharpness = acousticSharpness(audioIn, fs, calFactor, ...
    SoundField="diffuse", TimeVarying=true);

roughness = acousticRoughness(audioIn, fs, calFactor, SoundField="diffuse");

fluctuation = acousticFluctuation(audioIn, fs, calFactor, SoundField="diffuse");

Return types: acousticLoudness returns a scalar (sones) for stationary analysis. With TimeVarying=true, it returns a column vector (time-varying loudness in sones). The second output (specificLoudness) is a matrix (time x critical bands). It does NOT return a struct or timetable.

Ask the user:

• ISO 532-1 (Zwicker, stationary/time-varying) or ISO 532-2 (Moore-Glasberg)?
• Free-field or diffuse-field recording?
• Time-varying analysis needed?

See references/psychoacoustic-metrics.md for method selection guidance (ISO 532-1 vs 532-2), interpretation tables (sone values, sharpness/roughness/fluctuation scales), tonality metric selection, and combined sound quality assessment workflows.

Unit Conversions

Use built-in functions — do NOT reimplement conversion formulas manually:

Conversion	Function	Method selection
phon → sone	phon2sone(phon, standard)	"ISO 532-1" (default), "ISO 532-2"
sone → phon	sone2phon(sone, standard)	"ISO 532-1" (default), "ISO 532-2"
Hz → ERB	hz2erb(freq)	Single formula (no variants)
ERB → Hz	erb2hz(erb)	Single formula (no variants)
Hz → bark	hz2bark(freq)	Single formula (no variants)
bark → Hz	bark2hz(bark)	Single formula (no variants)
Hz → mel	hz2mel(freq, MelStyle=s)	"oshaughnessy" (default), "slaney"
mel → Hz	mel2hz(mel, MelStyle=s)	"oshaughnessy" (default), "slaney"

When the user specifies a standard (e.g., "ISO 532-2"), pass it as the second positional argument for phon2sone/sone2phon. For hz2mel/mel2hz, use the MelStyle name-value pair.

Noise Compliance

Choose the function based on the application:

Situation	Function	Why
General background noise rating	noiseCriteria	Standard NC curves (ANSI/ASA S12.2)
HVAC system noise specifically	roomCriteria	RC Mark II designed for HVAC evaluation
Noise with audible surging/fluctuation	roomNoiseCriteria	RNC penalizes time-varying noise (requires ≥15 s recording)
Quick speech interference check	speechInterferenceLevel	Direct SIL value
Need recommended levels for room type	noiseCriteriaRecommendations	Lookup table by room use

[audioIn, fs] = audioread("office_noise.wav");

[NC, SIL, specImbalance, rattleRisk] = noiseCriteria(audioIn, fs, ...
    CalibrationFactor=2.3);

[RC, LMF, QAI, rattleRisk] = roomCriteria(audioIn, fs, ...
    CalibrationFactor=2.3);

SIL = speechInterferenceLevel(audioIn, fs, CalibrationFactor=2.3);

rec = noiseCriteriaRecommendations("NC", Occupancy="all");

Return types:

• NC is a string (e.g., "NC-35" or "NC-56 (1000 Hz)" if a band exceeds)
• RC is a string (e.g., "RC-32(N)" — the letter indicates spectrum quality: N/R/H/RV)
• SIL and LMF are double scalars (dB)
• QAI is a double scalar (dB; 0 = ideal neutral)
• specImbalance and rattleRisk are structs with .Summary (string) and .Details (table) — access .Summary for display. Exception: specImbalance degrades to a char scalar (not a struct) when NC exceeds the curve range (e.g., "Above NC-70") or falls below NC-15. Guard with isstruct(specImbalance) before dot-indexing.
• noiseCriteriaRecommendations requires a criteria name argument: "NC", "RNC", or "RC"

Ask the user:

• What type of space? (office, classroom, hospital, concert hall)
• Is the noise source HVAC? → prefer roomCriteria
• Is the noise steady or fluctuating? → if fluctuating, use roomNoiseCriteria

See references/noise-compliance.md for detailed output interpretation and selection guidance.

Speech Quality

[degraded, fs] = audioread("processed_speech.wav");
[reference, ~] = audioread("reference_speech.wav");

mosScore = visqol(degraded, reference, fs, Mode="speech");

intelligibility = stoi(degraded, reference, fs);

Critical: stoi and visqol argument order is (processed, reference, fs) — processed/degraded signal FIRST.

visqol modes: "speech" for narrowband/wideband speech (outputs 1-5 MOS), "audio" (default) for general audio.

STI/SII Measurement

Two methods are available for both speechTransmissionIndex and speechIntelligibilityIndex:

IR-based method (simpler — no alignment needed):

[ir, fs] = audioread("room_impulse_response.wav");
sti = speechTransmissionIndex(ir, fs);
sii = speechIntelligibilityIndex(ir, fs);

Direct method (from test signals transmitted through the system):

fs = 48000;
reference = stipaExcitation(fs, Duration=15);
% Transmit reference through system, record as 'processed'
[processed, ~] = audioread("received_stipa.wav");

% IMPORTANT: Align signals before computing (see Signal Alignment)
processed = alignsignals(processed, reference);

sti = speechTransmissionIndex(processed, reference, fs);

Ask the user:

• Do they have an impulse response? → use IR method (simpler, no alignment needed)
• Measuring live? → generate test signal, transmit, record, align, compute
• STI or SII? STI for room/PA systems (IEC 60268-16); SII for hearing/communication (ANSI S3.5)

See references/sti-measurement.md for STI NV arguments, post-processing adjustment workflow (Annex M), test signal generation, and result interpretation. See references/sii-measurement.md for SII standard clause selection (5.1-5.3), BandImportanceFunction guidance, measurement positioning, and result interpretation.

SPL Measurement (Offline)

[audioIn, fs] = audioread("recording.wav");

spl = splMeter(SampleRate=fs, ...
    Bandwidth="Full band", ...
    FrequencyWeighting="A-weighting", ...
    TimeWeighting="fast", ...
    CalibrationFactor=2.3);

[Lt, Leq, Lpeak, Lmax] = spl(audioIn);

% For a single overall value, read the final row
fprintf("Leq: %.1f dB(A)\n", Leq(end));

Valid Bandwidth values: "Full band" (overall level), "1 octave", "2/3 octave", "1/3 octave". There is no "broadband" option — use "Full band" for a single overall SPL value.

Output arity: All outputs (Lt, Leq, Lpeak, Lmax) have one row per input sample. For a 12-second file at 44.1 kHz, that is 539,648 rows. Never fprintf or disp the full output — it will hang. Instead:

• Single overall value: read Leq(end) (the final accumulated value)
• Time-interval readings: downsample to interval boundaries: Leq(round(fs*T):round(fs*T):end, :) where T is the desired interval in seconds

TimeInterval controls the integration window for Leq computation but does NOT reduce the number of output rows.

Do not manually compute 20*log10(rms(audioIn)/20e-6) — splMeter handles calibration, weighting, and time integration correctly per IEC 61672.

Multi-band (octave-band) SPL: Use splMeter with Bandwidth="1 octave" — do NOT use octaveFilterBank + manual dB conversion. Key properties for octave-band SPL:

[audioIn, fs] = audioread("recording.wav");

spl = splMeter(SampleRate=fs, ...
    Bandwidth="1 octave", ...
    FrequencyWeighting="Z-weighting", ...
    TimeWeighting="fast", ...
    FrequencyRange=[15 8000], ...
    OctaveFilterOrder=8, ...
    CalibrationFactor=2.3);

[Lt, Leq, Lpeak, Lmax] = spl(audioIn);
cf = getCenterFrequencies(spl);

Critical gotchas for octave-band splMeter:

• Property name is OctaveFilterOrder (not FilterOrder). FilterOrder does not exist on splMeter.
• FrequencyRange compares against true center frequencies (fc = 1000 × 10^(3b/10) per IEC 61260), not nominal values. A band is included only if its true center falls within [lower, upper]. Bands below 1 kHz have true centers slightly above nominal (e.g., "63 Hz" = 63.10); bands above 1 kHz have true centers below nominal (e.g., "2 kHz" = 1995.26, "8 kHz" = 7943.28). Use floor(true_center) for lower bounds and ceil(true_center) for upper bounds. Common case "16 Hz to 8 kHz": use [15 8000] — the 16 Hz true center is 15.85 (so 15 ≤ 15.85 includes it) and 8 kHz true center is 7943 (so 8000 ≥ 7943 includes it). Do NOT use [16 8000] — this drops the 16 Hz band.
• Output is per-sample (see Output arity note above). Never print the full output. For time-interval readings, downsample: Lt100ms = Lt(round(fs*0.1):round(fs*0.1):end, :);

See references/spl-measurement.md for weighting selection guidance (A/C/Z and fast/slow), SPL interpretation tables, occupational exposure limits, splMeter vs poctave equivalency, and common measurement scenarios.

Broadcast Loudness (EBU R 128)

[audioIn, fs] = audioread("broadcast_content.wav");
[loudness, loudnessRange] = integratedLoudness(audioIn, fs);

% Check compliance: target is -23 LUFS +/- 1 LU
isCompliant = abs(loudness - (-23)) <= 1;

Use integratedLoudness for offline compliance checks. Use loudnessMeter for streaming/real-time monitoring (see matlab-process-streaming-audio).

RT60

[ir, fs] = audioread("room_ir.wav");
rtsummary = rt60(ir, fs, Bandwidth="1 octave");

Ask the user:

• What is the room type? (drives the target RT60)
• Do they have an IR, or do they need to measure one?
• T30 (standard) vs T20 (when noise floor limits dynamic range)?

See references/rt60-measurement.md for target RT60 values by space type, T20/T30/EDT interpretation, frequency-dependent analysis, and relationship to STI.

Tonality

[audioIn, fs] = audioread("machine_noise.wav");

[tnr, tnrFreq, isProminent] = acousticToneToNoiseRatio(audioIn, fs);

[pr, prFreq, isProminent] = acousticProminenceRatio(audioIn, fs);

All outputs are empty (0×1) when no tonal components are detected. Check isempty(tnr) before indexing.

Standards-Compliant Filters (Custom Preprocessing)

The metric functions above perform standards-compliant filtering internally. Use standalone filter objects only for custom measurement workflows where no single metric function applies AND you do not need SPL output:

% A-weighting filter
wf = weightingFilter("A-weighting", fs);
weightedSignal = wf(audioIn);

% Octave-band filter bank (for waveform-level band decomposition only)
ofb = octaveFilterBank("1 octave", fs, ...
    FrequencyRange=[31.5 16000], FilterOrder=8);
bandSignals = ofb(audioIn);

If you need octave-band SPL levels, use splMeter with Bandwidth="1 octave" instead of octaveFilterBank + manual dB conversion. See the Multi-band SPL section above.

Signal Alignment

Function	Internal alignment?	Action
visqol	Yes	No action needed
stoi	No	Align before calling
speechTransmissionIndex	No	Align for direct method; IR method avoids issue
speechIntelligibilityIndex	No	Align for direct method; IR method avoids issue

For the direct (processed, reference, fs) method, always align signals:

processed = alignsignals(processed, reference);

SII is more sensitive to misalignment than STI. At 50 ms delay: SII error ~0.016, STI negligible. At 200 ms: SII ~0.036, STI ~0.017. The IR-based (ir, fs) method avoids alignment issues entirely.

Common Mistakes

Mistake	Why it fails	Correct approach
Call pesq(reference, degraded, fs)	Function does not exist	visqol(degraded, reference, fs, Mode="speech")
stoi(reference, processed, fs)	Wrong argument order	stoi(processed, reference, fs) — processed FIRST
acousticFluctuationStrength(...)	Wrong function name	acousticFluctuation(audioIn, fs, calFactor)
Implement NC from scratch with hardcoded curves	Misses SIL, spectrum imbalance, rattle risk	noiseCriteria(audioIn, fs, CalibrationFactor=cf)
20*log10(rms(x)/20e-6) for SPL	No weighting, no time integration, no calibration	splMeter with appropriate settings
Implement STI algorithm manually	~100 lines, error-prone, misses NV options	speechTransmissionIndex(ir, fs)
Divide signal by calibration factor manually	Incorrect; factor is a multiplier, not a divisor	Pass as positional arg or CalibrationFactor=value
Skip alignment for direct STI/SII	Silent accuracy degradation	alignsignals(processed, reference) before calling
Use acousticLoudness for broadcast loudness	Wrong standard (ISO 532 = sones, not LUFS)	integratedLoudness for EBU R 128 compliance
string(rattleRisk) or fprintf("%s", specImbalance)	These are structs, not strings	Access .Summary field: rattleRisk.Summary
specImbalance.Summary when NC is out of curve range	specImbalance degrades to char (not struct) when "Above NC-70" or below NC-15	Guard: if isstruct(specImbalance), disp(specImbalance.Summary); end
roomNoiseCriteria on recordings shorter than 15 s	ANSI/ASA S12.2-2019 requires ≥150 Lt samples at 100 ms intervals	Ensure recording is at least 15 seconds; use noiseCriteria for shorter clips
round(NC) or arithmetic on NC/RC output	NC and RC are strings, not numbers	Use directly: fprintf("%s\n", NC)
noiseCriteriaRecommendations() with no args	First argument (criteria name) is required	noiseCriteriaRecommendations("NC")
loudness.OverallLoudness after acousticLoudness	Returns a scalar or vector, not a struct	Use directly: fprintf("%.1f sones\n", loudness)
splMeter(..., FilterOrder=8)	Property does not exist on splMeter	Use OctaveFilterOrder=8
splMeter(..., FrequencyRange=[16 8000]) for 16 Hz–8 kHz	Drops 16 Hz band (true center is 15.85 Hz, which is below 16)	Use FrequencyRange=[15 8000] — use floor(true_center) for lower bound
octaveFilterBank + 20*log10(rms(band)/pRef) for SPL	Manual, no time weighting, no calibration integration	Use splMeter with Bandwidth="1 octave"
Treating splMeter output as a single value	All outputs have one row per input sample (e.g., 539k rows for 12s at 44.1 kHz)	Use Leq(end) for a single value, or downsample to interval boundaries

Conventions

• Always ask about the user's measurement goal before selecting a function
• Always ask about calibration — recommend calibrateMicrophone for system-level accuracy
• Default to SoundField="free" for anechoic/outdoor, "diffuse" for reverberant rooms
• Default to Method="ISO 532-1" for loudness (Zwicker method, most common)
• Use IR-based methods for STI/SII when an impulse response is available (avoids alignment)
• Pass CalibrationFactor explicitly rather than pre-scaling the signal
• For noise compliance, prefer roomCriteria for HVAC and noiseCriteria for general background noise

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