modulationSpectrum {soundgen}R Documentation

Modulation spectrum

Description

Produces a modulation spectrum of waveform(s) or audio file(s), with temporal modulation along the X axis (Hz) and spectral modulation (1/KHz) along the Y axis. A good visual analogy is decomposing the spectrogram into a sum of ripples of various frequencies and directions. Roughness is calculated as the proportion of energy / amplitude of the modulation spectrum within roughRange of temporal modulation frequencies. The frequency of amplitude modulation (amMsFreq, Hz) is calculated as the highest peak in the smoothed AM function, and its purity (amMsPurity, dB) as the ratio of this peak to the median AM over amRange. For relatively short and steady sounds, set amRes = NULL and analyze the entire sound. For longer sounds and when roughness or AM vary over time, set amRes to get multiple measurements over time (see examples).

Usage

modulationSpectrum(
  x,
  samplingRate = NULL,
  scale = NULL,
  from = NULL,
  to = NULL,
  amRes = 5,
  maxDur = 5,
  logSpec = FALSE,
  windowLength = 15,
  step = NULL,
  overlap = 80,
  wn = "hanning",
  zp = 0,
  power = 1,
  normalize = TRUE,
  roughRange = c(30, 150),
  amRange = c(10, 200),
  returnMS = TRUE,
  returnComplex = FALSE,
  summaryFun = c("mean", "median", "sd"),
  averageMS = FALSE,
  reportEvery = NULL,
  cores = 1,
  plot = TRUE,
  savePlots = NULL,
  logWarpX = NULL,
  logWarpY = NULL,
  quantiles = c(0.5, 0.8, 0.9),
  kernelSize = 5,
  kernelSD = 0.5,
  colorTheme = c("bw", "seewave", "heat.colors", "...")[1],
  col = NULL,
  main = NULL,
  xlab = "Hz",
  ylab = "1/KHz",
  xlim = NULL,
  ylim = NULL,
  width = 900,
  height = 500,
  units = "px",
  res = NA,
  ...
)

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)

amRes

target resolution of amplitude modulation, Hz. If NULL, the entire sound is analyzed at once, resulting in a single roughness value (unless it is longer than maxDur, in which case it is analyzed in chunks maxDur s long). If amRes is set, roughness is calculated for windows ~1000/amRes ms long (but at least 3 STFT frames). amRes also affects the amount of smoothing when calculating amMsFreq and amMsPurity

maxDur

sounds longer than maxDur s are split into fragments, and the modulation spectra of all fragments are averaged

logSpec

if TRUE, the spectrogram is log-transformed prior to taking 2D FFT

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, %

wn

window type accepted by ftwindow, currently gaussian, hanning, hamming, bartlett, rectangular, blackman, flattop

zp

window length after zero padding, points

power

raise modulation spectrum to this power (eg power = 2 for ^2, or "power spectrum")

normalize

if TRUE, the modulation spectrum of each analyzed fragment maxDur in duration is separately normalized to have max = 1

roughRange

the range of temporal modulation frequencies that constitute the "roughness" zone, Hz

amRange

the range of temporal modulation frequencies that we are interested in as "amplitude modulation" (AM), Hz

returnMS

if FALSE, only roughness is returned (much faster)

returnComplex

if TRUE, returns a complex modulation spectrum (without normalization and warping)

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

averageMS

if TRUE, the modulation spectra of all inputs are averaged into a single output; if FALSE, a separate MS is returned for each input

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

if TRUE, plots the modulation spectrum of each sound

savePlots

if a valid path is specified, a plot is saved in this folder (defaults to NA)

logWarpX, logWarpY

numeric vector of length 2: c(sigma, base) of pseudolog-warping the modulation spectrum, as in function pseudo_log_trans() from the "scales" package

quantiles

labeled contour values, % (e.g., "50" marks regions that contain 50% of the sum total of the entire modulation spectrum)

kernelSize

the size of Gaussian kernel used for smoothing (1 = no smoothing)

kernelSD

the SD of Gaussian kernel used for smoothing, relative to its size

colorTheme

black and white ('bw'), as in seewave package ('seewave'), or any palette from palette such as 'heat.colors', 'cm.colors', etc

col

actual colors, eg rev(rainbow(100)) - see ?hcl.colors for colors in base R (overrides colorTheme)

xlab, ylab, main, xlim, ylim

graphical parameters

width, height, units, res

parameters passed to png if the plot is saved

...

other graphical parameters passed on to filled.contour.mod and contour (see spectrogram)

Details

Algorithm: prepare a spectrogram, take its logarithm (if logSpec = TRUE), center, perform a 2D Fourier transform (see also spectral::spec.fft()), take the upper half of the resulting symmetric matrix, and raise it to power. The result is returned as $original. For plotting purposes, the modulation matrix can be smoothed with Gaussian blur (see gaussianSmooth2D) and log-warped (if logWarp is a positive number). This processed modulation spectrum is returned as $processed. If the audio is long enough, multiple windows are analyzed, resulting in a vector of roughness values. For multiple inputs, such as a list of waveforms or path to a folder with audio files, the ensemble of modulation spectra can be interpolated to the same spectral and temporal resolution and averaged (if averageMS).

Value

Returns a list with the following components:

References

See Also

plotMS spectrogram analyze

Examples

# White noise
ms = modulationSpectrum(runif(16000), samplingRate = 16000,
  logSpec = FALSE, power = TRUE,
  amRes = NULL)  # analyze the entire sound, giving a single roughness value
str(ms)

# Harmonic sound
s = soundgen(amFreq = 25, amDep = 50)
ms = modulationSpectrum(s, samplingRate = 16000, amRes = NULL)
ms[c('roughness', 'amMsFreq', 'amMsPurity')]  # a single value for each
ms1 = modulationSpectrum(s, samplingRate = 16000, amRes = 5)
ms1[c('roughness', 'amMsFreq', 'amMsPurity')]
# measured over time (low values of amRes mean more precision, so we analyze
# longer segments and get fewer values per sound)

# Embellish
ms = modulationSpectrum(s, samplingRate = 16000,
  xlab = 'Temporal modulation, Hz', ylab = 'Spectral modulation, 1/KHz',
  colorTheme = 'heat.colors', main = 'Modulation spectrum', lty = 3)

## Not run: 
# A long sound with varying AM and a bit of chaos at the end
s_long = soundgen(sylLen = 3500, pitch = c(250, 320, 280),
                  amFreq = c(30, 55), amDep = c(20, 60, 40),
                  jitterDep = c(0, 0, 2))
playme(s_long)
ms = modulationSpectrum(s_long, 16000)
# plot AM over time
plot(x = seq(1, 1500, length.out = length(ms$amMsFreq)), y = ms$amMsFreq,
     cex = 10^(ms$amMsPurity/20) * 10, xlab = 'Time, ms', ylab = 'AM frequency, Hz')
# plot roughness over time
spectrogram(s_long, 16000, ylim = c(0, 4),
  extraContour = list(ms$roughness / max(ms$roughness) * 4000, col = 'blue'))

# As with spectrograms, there is a tradeoff in time-frequency resolution
s = soundgen(pitch = 500, amFreq = 50, amDep = 100,
             samplingRate = 44100, plot = TRUE)
# playme(s, samplingRate = 44100)
ms = modulationSpectrum(s, samplingRate = 44100,
  windowLength = 50, step = 50, amRes = NULL)  # poor temporal resolution
ms = modulationSpectrum(s, samplingRate = 44100,
  windowLength = 5, step = 1, amRes = NULL)  # poor frequency resolution
ms = modulationSpectrum(s, samplingRate = 44100,
  windowLength = 15, step = 3, amRes = NULL)  # a reasonable compromise

# customize the plot
ms = modulationSpectrum(s, samplingRate = 44100,
  windowLength = 15, overlap = 80, amRes = NULL,
  kernelSize = 17,  # more smoothing
  xlim = c(-70, 70), ylim = c(0, 4),  # zoom in on the central region
  quantiles = c(.25, .5, .8),  # customize contour lines
  col = rev(rainbow(100)),  # alternative palette
  power = 2)                   # ^2
# Note the peaks at FM = 2/KHz (from "pitch = 500") and AM = 50 Hz (from
# "amFreq = 50")

# Input can be a wav/mp3 file
ms = modulationSpectrum('~/Downloads/temp/16002_Faking_It_Large_clear.wav')

# Input can be path to folder with audio files. Each file is processed
# separately, and the output can contain an MS per file...
ms1 = modulationSpectrum('~/Downloads/temp', kernelSize = 11,
                         plot = FALSE, averageMS = FALSE)
ms1$summary
names(ms1$original)  # a separate MS per file
# ...or a single MS can be calculated:
ms2 = modulationSpectrum('~/Downloads/temp', kernelSize = 11,
                         plot = FALSE, averageMS = TRUE)
plotMS(ms2$original)
ms2$summary

# Input can also be a list of waveforms (numeric vectors)
ss = vector('list', 10)
for (i in 1:length(ss)) {
  ss[[i]] = soundgen(sylLen = runif(1, 100, 1000), temperature = .4,
    pitch = runif(3, 400, 600))
}
# lapply(ss, playme)
# MS of the first sound
ms1 = modulationSpectrum(ss[[1]], samplingRate = 16000, scale = 1)
# average MS of all 10 sounds
ms2 = modulationSpectrum(ss, samplingRate = 16000, scale = 1, averageMS = TRUE, plot = FALSE)
plotMS(ms2$original)

# A sound with ~3 syllables per second and only downsweeps in F0 contour
s = soundgen(nSyl = 8, sylLen = 200, pauseLen = 100, pitch = c(300, 200))
# playme(s)
ms = modulationSpectrum(s, samplingRate = 16000, maxDur = .5,
  xlim = c(-25, 25), colorTheme = 'seewave',
  power = 2)
# note the asymmetry b/c of downsweeps

# "power = 2" returns squared modulation spectrum - note that this affects
# the roughness measure!
ms$roughness
# compare:
modulationSpectrum(s, samplingRate = 16000, maxDur = .5,
  xlim = c(-25, 25), colorTheme = 'seewave',
  power = 1)$roughness  # much higher roughness

# Plotting with or without log-warping the modulation spectrum:
ms = modulationSpectrum(soundgen(), samplingRate = 16000, plot = TRUE)
ms = modulationSpectrum(soundgen(), samplingRate = 16000,
  logWarpX = c(2, 2), plot = TRUE)

# logWarp and kernelSize have no effect on roughness
# because it is calculated before these transforms:
modulationSpectrum(s, samplingRate = 16000, logWarpX = c(1, 10))$roughness
modulationSpectrum(s, samplingRate = 16000, logWarpX = NA)$roughness
modulationSpectrum(s, samplingRate = 16000, kernelSize = 17)$roughness

# Log-transform the spectrogram prior to 2D FFT (affects roughness):
modulationSpectrum(s, samplingRate = 16000, logSpec = FALSE)$roughness
modulationSpectrum(s, samplingRate = 16000, logSpec = TRUE)$roughness

# Complex modulation spectrum with phase preserved
ms = modulationSpectrum(soundgen(), samplingRate = 16000,
                        returnComplex = TRUE)
plotMS(abs(ms$complex))  # note the symmetry
# compare:
plotMS(ms$original)

## End(Not run)

[Package soundgen version 2.6.3 Index]