Calculates root mean square (RMS) amplitude in overlapping windows, providing an envelope of RMS amplitude - a measure of sound intensity. Longer windows provide smoother, more robust estimates; shorter windows and more overlap improve temporal resolution, but they also increase processing time and make the contour less smooth.
getRMS(
x,
samplingRate = NULL,
scale = NULL,
from = NULL,
to = NULL,
windowLength = 50,
step = NULL,
overlap = 70,
stereo = c("left", "right", "average", "both")[1],
killDC = FALSE,
normalize = TRUE,
windowDC = 200,
summaryFun = "mean",
reportEvery = NULL,
cores = 1,
plot = FALSE,
savePlots = NULL,
main = NULL,
xlab = "",
ylab = "",
type = "b",
col = "green",
lwd = 2,
width = 900,
height = 500,
units = "px",
res = NA,
...
)Returns a list containing:
a list of RMS amplitudes per frame for each sound, on the scale of input; names give time stamps for the center of each frame, in ms.
a dataframe with summary measures, one row per sound
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
sampling rate of x (only needed if x is a
numeric vector)
maximum possible amplitude of input used for normalization of
input vector (only needed if x is a numeric vector)
if NULL (default), analyzes the whole sound, otherwise from...to (s)
length of FFT window, ms
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 between successive FFT frames, %
'left' = only left channel, 'right' = only right channel, 'average' = take the mean of the two channels, 'both' = return RMS for both channels separately
if TRUE, removed DC offset (see also flatEnv)
if TRUE, the RMS amplitude is returned as proportion of
the maximum possible amplitude as given by scale
the window for calculating DC offset, ms
functions used to summarize each acoustic characteristic;
see analyze
when processing multiple inputs, report estimated time left every ... iterations (NULL = default, NA = don't report)
number of cores for parallel processing
if TRUE, plot a contour of RMS amplitude
full path to the folder in which to save the plots (NULL = don't save, '' = same folder as audio)
general graphical parameters
graphical parameters pertaining to the RMS envelope
graphical parameters for saving plots passed to
png
other graphical parameters
Note that you can also get similar estimates per frame from
analyze on a normalized scale of 0 to 1, but getRMS is
much faster, operates on the original scale, and plots the amplitude contour.
If you need RMS for the entire sound instead of per frame, you can simply
calculate it as sqrt(mean(x^2)), where x is your waveform.
Having RMS estimates per frame gives more flexibility: RMS per sound can be
calculated as the mean / median / max of RMS values per frame.
analyze getLoudness