bandpass: Bandpass/stop filters

In soundgen: Sound Synthesis and Acoustic Analysis

Bandpass/stop filters

Description

Filtering in the frequency domain with FFT-iFFT: low-pass, high-pass, bandpass, and bandstop filters with flexible filter depth (dB) and adjustable bandwidths at filter boundaries. Similar to the "ffilter" function from the seewave package, but here we use FFT instead of STFT - that is, the entire sound is processed at once. This works best for relatively short sounds (seconds), but gives us maximum precision (e.g., for precise notch filtering) and doesn't affect the attack and decay. NAs are accepted and can be interpolated or preserved in the output. Because we don't do STFT, arbitrarily short vectors are also fine as input - for example, we can apply a low-pass filter prior to decimation when changing the sampling rate without aliasing. Note that, unlike pitchSmoothPraat, bandpass by default applies an abrupt cutoff instead of a smooth gaussian filter, but this behavior can be adjusted with the bw argument.

Usage

bandpass(
  x,
  samplingRate = NULL,
  lwr = NULL,
  upr = NULL,
  action = c("pass", "stop"),
  dB = Inf,
  bw = 0,
  na.rm = TRUE,
  from = NULL,
  to = NULL,
  normalize = FALSE,
  reportEvery = NULL,
  cores = 1,
  saveAudio = FALSE,
  plot = FALSE,
  savePlots = FALSE,
  embed = FALSE,
  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)
lwr, upr	cutoff frequencies, Hz. Specifying just lwr gives a high-pass filter, just upr low-pass filter with action = 'pass' (or vice versa with action = 'stop'). Specifying both lwr and upr a bandpass/bandstop filter, depending on 'action'
action	"pass" = preserve the selected frequency range (bandpass), "stop" = remove the selected frequency range (bandstop)
dB	a positive number giving the strength of effect in dB (defaults to Inf - complete removal of selected frequencies)
bw	bandwidth of the filter cutoffs, Hz. Defaults to 0 (abrupt, step function - avoid!), a positive number corresponds to the standard deviation of a Gaussian curve, and two numbers set different bandwidths for the lower and upper cutoff points
na.rm	if TRUE, NAs are interpolated; otherwise, they are preserved in the output
from, to	if NULL (default), analyzes the whole sound, otherwise from...to (s)
normalize	if TRUE, resets the output to the original scale (otherwise filtering often reduces the amplitude)
reportEvery	when processing multiple inputs, report estimated time left every reportEvery iterations (NULL = default, NA = don't report); see reportTime
cores	number of cores for parallel processing
saveAudio	if TRUE, saves the processed audio in a subdirectory named after the function and created in the input directory (if input is a file or folder) or in the working directory
plot	if TRUE, produces a plot of the results
savePlots	if TRUE, creates a subdirectory in the input directory (if input is a file or folder) or in the working directory (if input is a vector etc), named after the function (eg "spectrogram/"). All plots and audio files (if any) are saved in this new directory. If there are multiple inputs, an html notebook is also created for easy viewing and listening
embed	if TRUE and savePlots is set and there are multiple inputs, all saved images and audio (if any) are embedded in the exported html notebook for easy sharing; if FALSE, the html file links to separate images and audio files (but separate files are still saved). NB: for this to work, package "base64enc" must be installed
width, height, units, res	graphical parameters for saving plots passed to png
...	other graphical parameters passed to plot() and meanSpectrum

Details

Algorithm: fill in NAs with constant interpolation at the edges and linear interpolation in the middle; perform FFT; set the frequency ranges to be filtered out to 0; perform inverse FFT; set to the original scale; put the NAs back in.

Value

Numeric vector of the same length as input.

Examples

# Filter white noise
s1 = fade(c(rnorm(2000, 0, 1), NA), samplingRate = 16000)

# low-pass
s1_lp = bandpass(s1, 16000, upr = 2000, bw = 100, plot = TRUE)
meanSpectrum(s1_lp, 16000)

# high-pass by 40 dB
s1_hp = bandpass(s1, 16000, lwr = 2000, dB = 40, bw = 50,
  plot = TRUE, windowLength = 5)
# windowLength is passed to meanSpectrum for plotting (ignore the warnings)

# bandstop
bandpass(s1, 16000, lwr = 1000, upr = 1800, bw = 50, action = 'stop', plot = TRUE)

# bandpass (turns noise into a whistle)
s1_bp = bandpass(s1, 16000, lwr = 2000, upr = 2100, bw = 25, plot = TRUE)
# playme(rep(s1_bp, 5))
# spectrogram(s1_bp, 16000)

# low-pass and interpolate a short vector with some NAs
x = rnorm(150, 10) + 3 * sin((1:50) / 5)
x[sample(seq_along(x), 50)] = NA
plot(x, type = 'l')
x_bandp = bandpass(x, samplingRate = 100, upr = 10, bw = 1)
points(x_bandp, type = 'l', col = 'blue')

# boost or attenuate a particular frequency band by 20 dB
s3 = bandpass(s1, 16000, lwr = 1700, upr = 2100, bw = 200,
  dB = 20, plot = TRUE)
# spectrogram(s3, 16000)
s4 = bandpass(s1, 16000, lwr = 2000, upr = 4300, bw = c(100, 500),
  dB = 60, action = 'stop', plot = TRUE)
# spectrogram(s4, 16000)

# precise notch filtering is possible, even in low frequencies
whiteNoise = rnorm(16000)
s5 = bandpass(whiteNoise, 16000, lwr = 30, upr = 40, normalize = TRUE,
              plot = TRUE, xlim = c(0, 0.5))
spectrum(s5, 16000, xlim = c(0, 0.1))
# playme(s5)
# spectrogram(s5, 16000, windowLength = 150, yScale = 'log')

## Not run: 
# high-pass all audio files in a folder
bandpass('~/Downloads/temp', lwr = 2000, saveAudio = TRUE, savePlots = TRUE)

## End(Not run)

soundgen documentation built on Sept. 20, 2026, 5:07 p.m.