Compander

Compander is a classic compressor/expander. What is different here is the use of a peaks response file. The peaks response file is a frequency response, analyzed from a segment of the sound, that is taken to represent the peak bin amplitudes for the sound. Each frequency bin of the peaks frequency response functions as the 0 dB reference point for that frequency bin. The amplitude of the frequency bin is companded relative to this reference.

See also: Freqresponse

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Amplitude Reports Print Mode
Analysis Frames Per Second
Begin Time
Compression Threshold in dB
Decibels of Compression
Decibels of Expansion
End Time
Envelope Attack in Seconds
Envelope Release in Seconds
EQ - Low Shelf Gain
EQ - High Shelf Gain
EQ - Low Shelf Frequency
EQ - High Shelf Frequency
FFT Length
Freqresponse File
Frequency Shift Factor
Gain
High Cutoff Frequency
Low Cutoff Frequency
Octaves of Rolloff from Frequency Band
Oscillator Resynthesis Threshold in Decibels
Output Format
Peak Rescale Level
Peaks Frequency Response Printout: High Cutoff Frequency in Hz
Pitch Transposition in Semitones
Resynthesis Channel
Smoothing Bandwidth in Octaves
Time Expansion/Contraction Factor
Time Interval Between Reports
Window Size in Samples
Window Type

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

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.


Begin Time

The time, in seconds, at which to begin processing the soundfile.


Compression Threshold in dB

Determines the threshold for compression. Any frequency louder than this parameter will be compressed.


Decibels of Compression

Determines how much to reduce frequencies louder than the compression threshold by.


Decibels of Expansion

Determines how much louder to make sounds which are quieter than the expansion threshold.


End Time

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.


Expansion Threshold in dB

Determines the threshold of expansion. Any frequencies quieter than this will be increased by the dB of the expansion gain parameter.


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.


Freqresponse File

If the button to use an existing freqresponse file is active, this path indicates its path.


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

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.


High Cutoff Frequency

The top barrier for companding. Frequencies above this point will not be affected during compression/expansion.


Low Cutoff Frequency

The bottom barrier for companding. Frequencies below this point will not be affected during compression/expansion.


Octaves of Rolloff from Frequency Band


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. The channels are processed one at a time beginning with the first channel. The first pass writes zeros in the channels yet to be processed, replacing them when processing proceeds to those 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.


Peaks Frequency Response Printout: High Cutoff Frequency in Hz

The cutoff frequency for data printed out by the instrument. No data will be printed about frequencies higher than this parameter.


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.


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.


Smoothing Bandwidth in Octaves

This mechanism allows routines that use frequency response files to smooth the responses by replacing the magnitude of a frequency bin with an average taken from a band centered around that bin.

The degree of smoothing is controlled through the bandwidth specified in octaves. The larger the bandwidth, the more smoothing.


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.