Envelope is a routine for tracking the amplitude envelope of a sound. Output can be ASCII, floats or a soundfile. Selecting floats or ASCII will produce a file suitable for use in the control of a parameter.
Amplitude Envelope Warp
Many of the routines employ the principle of warping in which a distribution of values is transformed by an identity function. In these places an exponential function is employed to remap a 0-1 range of values into a new orientation that preserves the minima (0) and maxima (1) while bringing the distribution closer to either extreme as a result of the curvature of the exponential function selected. The curvature of the exponential function is selected through a warp index. Specifically, warp index w will reorient the input x through the function below (^ = exponentiation).
y = (1. - (e^(x * w))) / (1. - (e^w))
In this function, the warp index of 0 produces a linear function and an untransformed output. Positive warp index values of increasing magnitude produce curves of increasing concavity (increasing slope) that draw values towards the 0-valued minima, and reduce the function integral. Negative values do the opposite, drawing values towards the maxima of 1, increasing the integral.
The practical use of this mechanism is found in various places. One such place is the reshaping of the frequency response distribution characteristics. In this, positive warp indeces cause the peaks of the response to be accentuated while the weaker frequencies are expanded out (i.e. pushed towards 0). Negative values have the opposite effect as they compress the dynamic range of the response and raise the relative level of the weaker noise components. Another place where warp applies is in the remapping of FFT amplitudes through the spectrum warpshape. In this, the sucessive FFT frames have their amplitudes remapped by the identity function, similiarly expanding or compressing the dynamic range depending upon the warp specified; 0 (linear warp function) leaves the amplitudes unchanged.
Analysis Frames per Second
This controls how often the phase vocoder will perform an analysis on the signal. It is a translation of the classic decimation control that specifies how many samples to skip between analysis frames. More frames increases the resolution of time but decrease speed. 200 frames per second is a good reference point. If you expand time you should increase this proportionately to maintain about 200 or more frames per second.
Compression Threshold in Decibels
Determines the threshold for compression. Any frequency louder than this parameter will be compressed.
Data Type
Determines how the instrument will read the values in the fields which set the upper and lower detection boundaries. 0 means the values will be read as frequencies, 1 as being in the form octave.pitchclass.
End Time in Seconds
The time, in seconds, at which to stop processing the soundfile. 0 or less is equivalent to the duration of the soundfile.
Envelope Modifications
The rate at which amplitude changes are allowed to occur effects how smooth spectral evolutions will be. To control this, many routines contain attack and decay response times controls: once translated these controls manipulate the coefficients of the following filter.
y(n) = (1. - A) * x(n) + A * y(n)
The filter is a lowpass designed to increasingly smooth the sudden changes in a signal as the value of the coefficient, A, is increased. Its control is through the response time parameter which is the time in seconds it takes a signal, shifting from one state to another, to decay to -60 dB of its former state. Response times are transformed to create the necessary coefficients for the selected frame rate. The response time is separated into attack and decay; this allows seperate control of the smoothing of the signal depending upon whether it is increasing or decreasing in amplitude. Short attack/decay response times can be used in places where dynamic processing induces garble or even pops. You can use longer response times to generally smooth or blur the onset/offset of sound components, particularly if the response controls are being applied to a time-varying filter. When applied to amplitudes, longer decay respsonse-times do not sound good, for in their delay of the decay, they end up amplifying te residual noise of a sound.
Envelope Attack Time in Seconds
Envelope attack time affects the speed at which the amplitude of a sound changes. Large values blur the sound's attack, smaller values sharpen it.
Envelope Release Time in Seconds
Envelope release time affects the speed at which the amplitude of a sound changes. Large values cause the sound to fade for a longer period, smaller values cause the sound to cut off more suddenly.
FFT Length
The FFT size must be a power of 2. Larger FFT sizes resolve frequencies better but transient behavior more poorly. Choose your FFT size according to the sound you are working with. A size of 1024 or 2048 works well in most cases.
High Frequency/Pitch Boundary
The upper boundary used when analyzed the input soundfile. Frequencies/pitches above this will be ignored.
Low Frequency/Pitch Boundary
The lower boundary used when analyzed the input soundfile. Frequencies/pitches above this will be ignored.
Multiple Channel Method
Determines whether the output file will contain data on the peak or average amplitude for each frame. 0 indicates peak amplitude, 1 indicates average.
Output Data Format
Determines what kind of data is output by Centroid:
0 = Frequency Units
1 = Pitch In Octave Units
2 = Octave.pitchclass Code
3 = Semitones of Deviation from Reference Pitch
4 = Inverted Semitones of Deviation from Reference Pitch
Output Data Type
Determines how the data will be saved. 0 indicates an ASCII file, 1 indicates 32-bit floats.
Print Elapsed Time
Prints out the time index where the process currently is in the soundfile while it is being analyzed.\n\n0 turns this off, 1 turns it on.
Resynthesis Channel
All routines allow both monophonic and multi-channel input files to be processed. With multi-channelled files, you can either select one channel and produce a monophonic output file, or process all the channels. Channels are numbered beginning with 1. Processing of multi-channelled files is done one channel at a time beginning with channel 1, with zeros written to channels which have yet to be processed. Processing one channel at a time requires less memory and allows you to audition the output sooner than if you did all channels at once.
Use 0 to process all channels.
Window Size in Samples
The window size is a less opaque parameter; like the FFT, it must be a power of 2. Windows twice the size of the FFT work well. Larger window sizes may resolve frequencies better. Specifying 0 for the window size will automatically set the window to twice the FFT size.
Window Type
The FFT and inverse FFT are computed using a window. Like the FFT size, the shape of the window used can effect the quality of the analysis and resynthesis. (See F.R.Moore, Stieglitz, or Roads for further explanation.) A variety of windows are available including: Hamming, Rectangular, Blackman, Triangular, and Kaiser (in 8 different forms as related to 8 different alpha values). Blackman (-w2) or Kaiser (-w8) are recommended for most applications. In some unusual cases where transient behavior is being lost, consider using other windows such as the Rectangular, although take care to assure that it is not producing pops or a buzzy sound.