getLoudness: Get loudness

Description

Estimates subjective loudness per frame, in sone. Based on EMBSD speech quality measure, particularly the matlab code in Yang (1999) and Timoney et al. (2004). Note that there are many ways to estimate loudness and many other factors, ignored by this model, that could influence subjectively experienced loudness. Please treat the output with a healthy dose of skepticism! Also note that the absolute value of calculated loudness critically depends on the chosen "measured" sound pressure level (SPL). getLoudness estimates how loud a sound will be experienced if it is played back at an SPL of SPL_measured dB. The most meaningful way to use the output is to compare the loudness of several sounds analyzed with identical settings or of different segments within the same recording.

Usage

getLoudness(
  x,
  samplingRate = NULL,
  scale = NULL,
  from = NULL,
  to = NULL,
  windowLength = 50,
  step = NULL,
  overlap = 50,
  SPL_measured = 70,
  Pref = 2e-05,
  spreadSpectrum = TRUE,
  summaryFun = c("mean", "median", "sd"),
  reportEvery = NULL,
  cores = 1,
  plot = TRUE,
  savePlots = NULL,
  main = NULL,
  ylim = NULL,
  width = 900,
  height = 500,
  units = "px",
  res = NA,
  mar = c(5.1, 4.1, 4.1, 4.1),
  ...
)

Value

Returns a list:

specSone: spectrum in bark-sone (one per file): a matrix of loudness values in sone, with frequency on the bark scale in rows and time (STFT frames) in columns
loudness: a vector of loudness in sone per STFT frame (one per file)
summary: a dataframe of summary loudness measures (one row per file)

Arguments

x: path to a folder, one or more wav or mp3 files c('file1.wav', 'file2.mp3'), Wave object, numeric vector, or a list of Wave objects or numeric vectors
samplingRate: sampling rate of x (only needed if x is a numeric vector)
scale: maximum possible amplitude of input used for normalization of input vector (only needed if x is a numeric vector)
from, to: if NULL (default), analyzes the whole sound, otherwise from...to (s)
windowLength: length of FFT window, ms
step: you can override overlap by specifying FFT step, ms (NB: because digital audio is sampled at discrete time intervals of 1/samplingRate, the actual step and thus the time stamps of STFT frames may be slightly different, eg 24.98866 instead of 25.0 ms)
overlap: overlap between successive FFT frames, %
SPL_measured: sound pressure level at which the sound is presented, dB
Pref: reference pressure, Pa (currently has no effect on the estimate)
spreadSpectrum: if TRUE, applies a spreading function to account for frequency masking
summaryFun: functions used to summarize each acoustic characteristic, eg "c('mean', 'sd')"; user-defined functions are fine (see examples); NAs are omitted automatically for mean/median/sd/min/max/range/sum, otherwise take care of NAs yourself
reportEvery: when processing multiple inputs, report estimated time left every ... iterations (NULL = default, NA = don't report)
cores: number of cores for parallel processing
plot: should a spectrogram be plotted? TRUE / FALSE
savePlots: full path to the folder in which to save the plots (NULL = don't save, '' = same folder as audio)
main: plot title
ylim: frequency range to plot, kHz (defaults to 0 to Nyquist frequency). NB: still in kHz, even if yScale = bark, mel, or ERB
width, height, units, res: graphical parameters for saving plots passed to png
mar: margins of the spectrogram
...: other plotting parameters passed to spectrogram

Details

Algorithm: calibrates the sound to the desired SPL (Timoney et al., 2004), extracts a spectrogram with powspec, converts to bark scale with (audspec), spreads the spectrum to account for frequency masking across the critical bands (Yang, 1999), converts dB to phon by using standard equal loudness curves (ISO 226), converts phon to sone (Timoney et al., 2004), sums across all critical bands, and applies a correction coefficient to standardize output. Calibrated so as to return a loudness of 1 sone for a 1 kHz pure tone with SPL of 40 dB.

References