The idea behind filtdeviator is to use a frequency response function to not only filter a sound (as with filter), but to create a topology of frequency deviation working in correlation with the filter. Consequently, filtdeviator is filter with added parameters for specifying how the filter frequency response function will be mapped into the deviation of frequency. The added parameters set the base and peak deviation for how the response will be mapped into both pitch transposition and frequency shift, and how the function will be warped within the range set by these limits. There is also a master (0-1) deviation control for globally controlling the deviation.
All the controls of filtdeviator allow you to dynamically vary the presence and effect of amplitude filtering and frequency deviation, making filtdeviator an interesting routine for exploring the way filters can be used to impede/transform the resonant signature of a sound. Using small amounts of frequency deviation with no amplitude filtering and a sweeping transposition of the filter will produce an effect something akin to the commercial guitar phase shifter; larger amounts of deviation take it into another place entirely. Adding the correlated amplitude filtering conceals the deviation more (positioning it more at the edges of formants), producing a sound something like the floppy resonant behavior of slide whistles.
See also: Freqresponse, Filtresponsemaker, Chordresponsemaker
Amplitude Deviation Response Time in Seconds
Use this parameter to control the rate at which amplitude changes occur. The number of seconds given dictate the number of seconds it takes for a sound to decay to -60 dB of its former state.
Amplitude Reports Print Mode
Two flags are provided for controlling the output amplitude statistics; one turns the statistics on or off, and the other sets how often they will be reported. The statistics provide the peak output level in amplitude and decibels. With integer format output files, output values exceeding the normalized peak amplitude of 1. (0 dB) are clipped to a value of 1.0, and the statistics placed in clip mode; in clip mode reports are made only for frames where clipping occurs. The peak amplitude, its time, and the number of clipped samples are reported at the end of processing. With floating-point format output files, output values exceeding the normalized peak amplitude of 1. are not clipped since they will be rescaled in the second pass; output statistics proceed normally throughout. The levels before and after rescaling are reported at the end of processing.
0 turns amplitude reports off, 1 turns them on.
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.
Deviation Mode
0 = Response mode: In response mode, the filter response is used to determine frequency deviation between boundaries.
1 = Random mode: In random mode the base and peak form boundaries for a random function. The function, smoothed by the response time, is fitted into the boundaries to produce the resulting deviation from the bin's original frequency. The deviation is then attenuated by using the normalized frequency response for the bin as a gain control.
2 = File mode: In file mode, a function file, smoothed by the response time, is used in place of a random function. Give a file in place of a mode code number.
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.
Low/High Shelf Equalization
Equalization has been provided at various points in routines to allow for the needed adjustment of spectra. The EQ consists of low and hi shelf segments, whose width is adjusted through control of the shelf breakpoint frequency. The region between the shelf segments is represented by a linear decibel gradient between the decibel levels of the two shelves. Some routines implement the EQ before pitch changes, others after. EQ placed before pitch changes (pre-transpose/shift) will cause the EQ to be transposed with the pitch changes, whereas afterwards (post-transpose/shift) will keep them fixed as shifts and transpositions occur.
Low Shelf Gain
Determines how the amplitude of sounds below the low shelf frequency will be affected.
High Shelf Gain
Determines how the amplitude of sounds above the high shelf frequency will be affected.
Low Shelf Frequency
Determines the frequency below which the low shelf gain will be used.
High Shelf Frequency
Determines the frequency above which the high shelf gain will be used.
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.
Frequency Response Printout: High Cutoff Frequency in Hz
During execution, frequency response data is printed. Frequencies over this cutoff will not be printed. Set to 0 to turn all printing off.
Frequency Response Shift
Same as the frequency shift parameter, but operates on the frequency response file. See Frequency Shift Factor for more information.
Frequency Response Transposition in Semitones
Controls pitch transposition in semitones for the frequency response file. See Pitch Transposition in Semitones for more information.
Frequency Shift Factor
With the frequency shift control, a constant or function value is added to all the bin frequencies to produce a nonlinear pitch domain translation of the spectrum. Frequency shift is related to things like ring modulation and their similarly nonlinear shifts of pitch characteristics. Use this to create small distortions of the harmonic integrity of a sound.
Gain in Decibels
The output and other components can be gained. 0 dB represents unity gain, no change. A change of +/- 6 dB represents a doubling or halving of the amplitude. Increments of 10 dB are loosely associated with one change in dynamic level.
Oscillator Resynthesis Threshold in Decibels
The phase vocoder resynthesizes the signal using one of two methods, depending on the type of changes made to the FFT. If the changes are only to the magnitudes (amplitudes), then the faster overlap/add method is used. If however changes in frequency are made, then the FFT integrity is compromised, necessitating use of the oscillator bank method in which each bin is synthesized as a sine wave changing in frequency and amplitude. This method is slower, although a resynthesis threshold is available that can be used to increase the computation speed by turning off bins whose amplitude falls below the threshold. A threshold of -60dB is appropriate, although safety warrants using a lower threshold if the spectrum is thin and its decays exposed; use your ear.
Pitch Transposition in Semitones
With the pitch transposition control, a constant or function value is multiplied against all bin frequncies. This is classic transposition, here specified in semitones of transposition (12 semitones equals an octave). Conversion is made to produce the appropriate frequency multiplier.
Random Amplitude Deviation
Controls random amplitude deviation. 0 turns this feature off, 1 turns it on.
Randomize Mode
Passband randomize mode adds a random modulation to the amplitude passing at the level between the floor and the filter response level for the bin.
Stopband mode adds random amplitudes in the region above the response, returning in randomized form the amp that the response cuts away.
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.
Source Decibels Floor
Controls the mix of the source and filtered sounds. The filtered signal level is equal to 1 - source floor, which ranges from -96 to 0 dB.
Consequently, the source level functions as a floor over which lies the filtered signal. A source floor of 0 dB would neutralize filtering since there would be no filter range above the floor, a floor of -96 dB would produce the full effect of the filter.
Time Expansion/Contraction Factor
Once the spectral modifications are made to the FFT analysis, an inverse FFT is invoked to produce the samples of a time-domain signal. The classic phase vocoder paradigm controls the number of samples through the interpolation value and its relation to the decimation. The arcane relationship of decimation and interpolation is here translated into the parameter of time expansion/contraction, allowing for the direct scaling of time. Use values greater than 1 to expand time, less than 1 contract it.
Time Interval Between Reports
Determines the interval in seconds of the soundfile between amplitude reports. See Amplitude Reports Print Mode for a further explaination.
Transposition/Shift Flag
Determines whether to apply pitch transposition and frequency shift factor before or after filtering. 0 applies them before filtering, 1 applies them after.
Warp Index
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.
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.