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Extract voice quality parameters (NAQ, QOQ, H1-H2, HRF, PSP) per utterance

lst_covarep_vq.Rd

Computes five IAIF-derived glottal voice quality measures (NAQ, QOQ, H1-H2, HRF, PSP) summarised per utterance. For frame-by-frame trajectories see trk_covarep_vq_gci.

Usage

lst_covarep_vq(listOfFiles, beginTime = 0, endTime = 0, f0 = NULL, gci = NULL, gci_in_samples = FALSE, verbose = TRUE, toFile = FALSE, explicitExt = "cvq", outputDirectory = NULL)

Arguments

listOfFiles

Character vector of audio file paths. Any format supported by av is accepted; non-native inputs are transcoded automatically.

f0

Optional F0 estimate. Can be:

  • Scalar: Single F0 value for entire signal

  • Vector: F0 contour (median of voiced frames will be used)

  • NULL: H1-H2 will not be computed (returns NA)

gci

Optional glottal closure instants. Can be:

  • Numeric vector of sample indices

  • Numeric vector of times in seconds (will be converted to samples)

  • NULL: NAQ and QOQ will not be computed (return NA)

gci_in_samples

Logical; if TRUE, gci is in sample indices; if FALSE, gci is in seconds (default: FALSE)

toFile

Logical. If TRUE, write results to JSTF file. Default FALSE.

explicitExt

Character. File extension for output. Default "cvq".

beginTime

Start time for the extracted portion in seconds. Default: NULL (beginning of signal). Note: uses beginTime/endTime (seconds) matching DSP function conventions, unlike read_audio() which uses begin/end.

endTime

The end time of the section of the sound files that should be analysed (in seconds). Use 0 for end of file.

verbose

Logical. Show a progress bar (sequential path) or a progress-aware parallel apply (pbapply/pbmcapply, if installed).

outputDirectory

The directory where the slice file should be stored. If not defiled (NULL), the sparse slice file will placed in the same folder as the media file.

Value

If toFile=FALSE (default), for single file: Named list with voice quality parameters. For multiple files: List of named lists. If toFile=TRUE, invisibly returns the path(s) to the written JSTF file(s).

Each parameter list contains:

glottal_flow_max

Peak glottal flow amplitude

glottal_flow_min

Minimum glottal flow value

glottal_derivative_peak

Maximum flow derivative (MFDR)

NAQ

Normalized Amplitude Quotient (NA if no GCI)

QOQ

Quasi-Open Quotient (NA if no GCI)

H1_H2

First two harmonics difference in dB (NA if no F0)

HRF

Harmonic Richness Factor

PSP

Parabolic Spectral Parameter

Details

Internally applies IAIF (Iterative Adaptive Inverse Filtering, (Alku 1992) ) to extract the glottal source, then computes GCI-anchored measures from the glottal flow and its derivative. Implemented in native C++ (no Python dependency).

References

(Alku 1992)

(Kane and Gobl 2013)

See also

trk_covarep_iaif for glottal waveforms, trk_pitch_srh for F0 estimation

Examples

if (FALSE) { # \dontrun{
# Basic usage (HRF and PSP only, no F0 or GCI)
vq <- lst_covarep_vq("vowel.wav")
print(vq$HRF)
print(vq$PSP)

# With F0 for H1-H2 computation
vq <- lst_covarep_vq("vowel.wav", f0 = 150)
print(vq$H1_H2)

# Batch processing
files <- c("a.wav", "e.wav", "i.wav", "o.wav", "u.wav")
vq_all <- lst_covarep_vq(files)
} # }