Ring uses the phase vocoder to create an all-pass resonator. It works by structuring the FFT resynthesis as a bank of feedback filters that feed back the sinusoid of each bin in a strength proportional to the amplitude of that bin (after adjustment by global feedback controls). This allows the sound to "ring" in a way something like reverb or comb filter resonance. The difference from comb filtering is that with ring spectral resonance is created not through a collection of comb filters selected for their ability to resonate various pulse wave spectra, but rather, through an array of feedback filters (sized by the FFT) that resonate a sine wave spectrum while dynamically tuning their feedback frequencies to the frequencies of the input sound. In short, it creates a kind of "self resonance".
Ring is a nice way of increasing the resonant pitch characteristics of a sound, although it has its weaknesses. Ring works best with larger FFT sizes as it is attempting to synthesize or accentuate the more pitched/harmonic characteristics of the sound; this is something larger FFTs, with their increased frequency resolution, handle better. Use of the Kaiser window, with its low sidelobe amplitudes, helps as well. In adition, there is a threshold for preventing the noise features of a sound from being resonated, plus an EQ that can be positioned to filter either the source input to the feedback loop, or the feedback return.
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.
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 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.
Output Format
The output sound file is written as a NeXT/Sun format sound file in either 16-bit short or 32-bit floating point format, of one or more channels. 0 tells PVCX to use the format of the input file, 1 equals integer format, and 2 equals rescaled floats.
Peak Rescale Level
Selection of the floating point output file format invokes an amplitude rescaling feature. Once processing is complete, a second pass through the sound file is made to rescale the values to the decibel level specified. A dB rescale level of 1 causes rescaling to the level of the original input file.
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 Floor in dB
The amount of random amplitude deviation to be used. 0 = none, -96 = full range.
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.
Threshold in dB
The source bins are input into the reverb path depending on whether they lie above or below the threshold. The passmode determines whether to pass bins above or below the threshold (0 = below, 1 = above). For harmonic sounds, passing above -20 works well. For some noisy sounds, passing stuff below -20 can be interesting.
Threshold Pass Mode
The source bins are input into the reverb path depending on whether they lie above or below the threshold. The passmode determines whether to pass bins above or below the threshold (0 = below, 1 = above). For harmonic sounds, passing above -20 works well. For some noisy sounds, passing stuff below -20 can be interesting.
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.
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.