(Names in brackets mean that these are separate programs. The others are sub-modules of SPECNU.)
The SPECNU Group includes command line programs to perform various audio cleaning procedures. The Sound Loom GUI contains a comprehensive Cleaning Kit that employs these programs as well as others, conveniently assembling a whole range of functions for audio cleaning in one place. This is described in the Technical section below.
Release 7 has added two other functions to the group: RAND and SQUEEZE.
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infile – input analysis file[Query: is there a dur parameter as in WARP?]
outfile – output analysis file
shape – breakpoint textfile containing time-semitone transposition pairs, defining the contour of the largest fractal shape.
Times must start at zero and increase, with the final time defining the duration of the pattern.
Pitch range: -12 to 12 semitones; the value at the final time is ignored.
-mmaxfrac – maximum degree of fractalisation (time-variable). If not set (or zero), fractalisation proceeds until the minimum possible wavelength is reached. (Range: 0 to 1000)
-tstr – time stretch of fractal pattern (Range: 0, or 1 to 2; time-variable). If set to zero, no timestretching is done.
-iwarp – interval warping of fractal pattern (Range: 0 to 8; time-variable). If set to zero, no warping is done.
-s – shrink pitch-intervals as fractal time-scales shrink
-n – transposition only (Default: retain formant envelope)
FRACTAL SPECTRUM is a spectral version of FRACTAL WAVE, distorting the spectrum fractally by transposition. The breakpoint file shape defines a time and transposition contour; transposition happens over the total duration of this and is then repeated over every resulting sub-unit of the pattern, over every sub-sub-unit, etc. until the smallest time-unit is reached.
The other parameters are optional. Maxfrac sets the degree of fractalisation, which otherwise proceeds until the minimum possible wavelength is reached; str time-stretches the fractal pattern, but not the sound. warp distorts the intervals defined in the shape pattern.
The process may best be understood by experimenting with a steady-pitched source and selecting the tranposition-only flag (-n). A burst of tones is produced at high speed, as defined by the shape datafile. This is not unlike the sister process FRACTAL WAVE (but faster). If the (-n) flag is then unset, the source's formant envelope is retained. With a speech source, the voice retains its overall integrity, but with pitch fluctations caused by the fractal transpositions.
SEE ALSO: the soundfile version FRACTAL WAVE.
End of FRACTAL SPECTRUM
Note: The CDP Release 8 set includes a program called SPECLEAN, which appears to be identical to this function.
sigfile – analysis file of the source sound to be cleaned
noisfile – analysis file cut from the source sound, containing a sample of the noise (only) to be eliminated. Try to find a section containing only that noise, e.g., at the beginning or end of the sound.
outanalfile – resulting cleaned sound
persist – minimum time (in ms.) for which the clean signal must persist above the threshold level if it is to be retained (Range 0 to 1000 ms.)
noisgain – multiplies the threshold noise-levels found in the noisfile by an amount you specify, before they are used for comparison with the source file. (Range: 1 to 40)
Both input files are analysis files. Noisfile is a sample of noise (only) from the sigfile.
This process enables sources (once analysed) to be cleaned, and is similar to the earlier version of SPEC CLEAN. It differs from SPEC CLEAN by having a persist parameter.
Thus with SPECNU CLEAN you can specify how long a signal must persist before you will accept it as valid data (rather than simply noise to be eliminated).
This avoids the 'bubbling' problem of the original SPEC CLEAN. With SPEC CLEAN, when a signal was near the noise threshold, the spectrum might be judged as acceptable signal, or as noise, from one window to another, possibly changing rapidly between (accepted) signal and (eliminated) noise. The result of this switching on and off of the signal in the output, is a 'bubbling' effect in the sound. The persist parameter ensures that the signal does not bubble in and out in this way.
This process is intended for cleaning up recordings made in noisy environments. It may be used alongside various editing and filtering processes, to clean a source sound bit by bit.
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1 Generates matrix, and transformed sound.
2 Use input matrix data to transform sound.
3 Exchange real and imaginary values in FFT output.
4 Invert phase in each channel, octave to octave
infile – input soundfile (MONO)
outfile(s) – output soundfile (adds extension) and (Mode 1) outname.txt matrix file
analchans – number of analysis channels, as used in PVOC (Power-of-2: 4-16384)
winoverlap – overlap of analysis windows, as used in PVOC (Range: 1-4)
inmatrixfile – text datafile generated by Mode 1
-c – Apply matrix-transform cyclically:
window1 = spec * matrix
window2 = spec * matrix * matrix
window3 = spec * matrix * matrix * matrix
and so on.
...
Mode 1 tends to produce extreme distortion, even with very low-level infile (t.b.c).
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Example command line to randomise the order of spectral windows:
specnu rand inanlfile outanalfile 0.5 50
inanalfile – input analysis file
outanalfile – output analysis file
-ttimescale – determines the number of windows locally-randomised. Default: all the windows are randomised
-ggrouping – consecutive windows are grouped into sets of size grouping. These sets are randomised. The default grouping is one window per set.Timescale may vary over time.
The degree of randomisation is controlled by the user. By default a set contains only one window. If all the windows are randomised, this means that the maximum randomisation is achieved. The amount of randomisation can be reduced in two ways using the optional parameters.
- Setting a timescale focuses the randomisation into specific time areas. Thus, as you move forward through the input file, time-specific areas will be jumbled.
- Giving a value for grouping > 1 keeps this number of consecutive windows the same, but then jumbles the sets of windows.
SPECNU RAND provides a way of texturing a sound. Furthermore, as it is operating in the Spectral Domain, the process will create timbral variations.
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1 Removes pitch and its harmonics up to a specified frequency limit
2 Removes everything besides the specified pitched material
inanalfile – input analysis file with pitched components
outanalfile – resulting analysis file with pitched component removed
midimin – minimum pitch to remove (MIDI pitch values 1-127)
midimax – maximum pitch to remove (MIDI pitch values 1-127) (all pitches between midimin and midimax are removed)
rangetop – frequency (hz) at which the search for associated harmonics (TO BE ELIMINATED) stops
atten – degree of attenuation of the suppressed components Range: 0 to 1 (1 = maximum attenuation, 0 = no attenuation)
The MIDI range between minimin and midimax should be small.
If the range is an octave or more, the whole spectrum between midimin and rangetop will be removed.
SPECNU REMOVE will only work with stable, clearly-pitched components in the spectrum. It allows unwanted signals with steady pitch to be removed from a source. Deciding how many harmonics (and how large a pitch-range) should be eliminated is a matter of judgement based on listening to the results. Removing more is not necessarily better as the resulting hole in the spectrum can appear to emphasize the very pitch(band) you wanted to eliminate.
This process is intended for cleaning up recordings made in noisy environments. It is particularly useful for removing pitched components of speech from unwanted speaking voices behind a recording. (Filtering processes can be used to remove unwanted sibillants). It may be used alongside various editing and filtering processes, to clean a source sound bit by bit. These various functions are integrated in the Cleaning Kit option on the Music Testbed of the Sound Loom.
Applied to a vocal sound, Mode 1 leaves a modest 'hole' in the sound. Trevor Wishart writes: "There are lots of complications in removing a pitch from a sound, which the user will discover. If you remove too many harmonics, the hole itself seems to produce the very pitch you were trying to remove! I have no idea why this is. The high frequency limit should therefore be kept as low as possible if you are seriously attempting to remove an existing pitch element from a sound."
Mode 2, in removing everything else, can leave a rather pinched voice, which may have compositional uses.
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Mode 1 Moirée slice – alternate chanwidth-wide (groups of) channels are sent to the different output files.
For example: with bandcnt = 3 and chanwidth = 1, groups of just 1 channel width are selected cyclically, to make 3 output files:Mode 2 Frequency band slice – equivalent to Mode 1. The chanwidth is defined as frequency (in Hz) rather than as a count of analysis channels (Range: 43.066406 to 11025.0).With bandcnt = 2 and chanwidth = 2, groups of 2-channel width are selected cyclically, to make 2 output files:
- File 1 gets channels 0 : 3 : 6 : 9 : 12 : 15 : 18 ... etc.
- File 2 gets channels 1 : 4 : 7 : 10 : 13 : 16 : 19 ... etc.
- File 3 gets channels 2 : 5 : 8 : 11 : 14 : 17 : 20 ... etc.
With bandcnt = 5 and chanwidth = 4, groups of 4-channel width are selected cyclically, to make 5 output files:
- File 1 gets channels 0+1 : 4+5 : 8+9 : 12+13 : 16+17 ... etc.
- File 2 gets channels 2+3 : 6+7 : 10+11 : 14+15 : 18+19 ... etc.
- File 1 gets channels 0+1+2+3 : 20+21+22+23 ... etc.
- File 2 gets channels 4+5+6+7 : 24+25+26+27 ... etc.
- File 3 gets channels 8+9+10+11 : 28+29+30+31 ... etc.
- File 4 gets channels 12+13+14+15 : 32+33+34+35 ... etc.
- File 5 gets channels 16+17+18+19 : 36+37+38+39 ... etc.
Mode 3 Pitch band slice – similar to Mode 2, but the chanwidth is now defined in semitones, and the slices are of equal pitch width (rather than of equal frequency width, as in Modes 1 and 2.
Mode 4 Harmonics slice – similar to Mode 2, but each band is centered on a harmonic of the sound. The position of the harmonics is determined from the pitchdata textfile (possibly extracted using REPITCH GETPITCH – using Mode 2 to produce a breakpoint text file).
Mode 5 Invert spectrum – spectral frequencies are inverted, around a given frequency value.
inanalfile – input analysis file
outanalfile – multiple output analysis files: the first file is as named, the second and subsequent files replace the last character of the first file's name with '1', '2', etc.
bandcnt – the number of output files (and the channel separation of the bands in the output sounds)
chanwidth – values differ in the various modes:pitchdata – breakpoint text file containing time pitch value pairs
- Mode 1: chanwidth is the number of channels in each band in each output sound
- Mode 2: chanwidth is the width in Hz of each band in each output sound
- Mode 3: chanwidth is the width in semitones of each band in each output sound
freq (Mode 5) – the frequency around which to invert all spectral frequencies.
This process slices a signal into a number of spectral components, that can then be reassembled into the original sound, or used in some other way. The output sounds will contain a number of bands (bandcnt) and channel widths (chanwidth), depending on the user's selections, so each sound will be an amalgam of different parts of the original source.
It is not possible to predict which sound will appear higher or lower, as each sound is made up of slices taken from the whole range of the source. Thus it all depends on what the source contains. In the various band slices, there could be:
- nothing at all (don't be surprised if you get a silent outfile)
- something, but a very low level (i.e., amplitude)
- something very prominent
- something that combines the contents of another band (in the stack in question) to produce a prominent output.
In Mode 4, the lower the pitch of the source, the more outputs are produced because there are more potential harmonics before you reach the maximum frequency limit. In fact, there can be a very large number of output files, many of which will be silent. This process does not know if there is any energy at any particular harmonic. It just tracks where the harmonics of the sound would be, and reports what it finds. A pure sine tone for example is going to produce many silent outputs. The number of silent outputs will be reduced if the sound has a very rich or bright spectrum, the program is of more use with this kind of input file.
Mode 5 does not produce multiple outputs and is, in effect, a separate function. The frequencies that make up the spectrum of a sound are inverted (turned upside-down) above and below the given frequency value, so that those that went upwards from this frequency now go down, and vice versa. A pitched sound will normally be turned into one with inharmonic partials. This is a powerful technique to create entirely new timbres from familiar sounds, and may be compared with STRANGE SHIFT, which also creates inharmonic sounds.
At the moment it appears to be a way to get a number of tonally different versions of the input analysis file rather quickly. The potential for composition lies in possible recombinations of these rather unpredictable slices, especially if they are processed separately before being recombined.
To give some idea of what might be done, consider that interesting IRCAM experiment where different rates of vibrato are applied to the odd and to the even harmonics of a vocal sound. You start with the odd harmonics in the left channel and the even harmonics in the right channel. With no vibrato, and you hear a (mono) voice at stage centre. But when you apply a differentiated vibrato you hear a 'clarinet' in one channel and a 'cello' in the other. This suggests the idea of generalising this approach, applying different processes to the different harmonics, and recombining the results, to see what happens. Do you get an elaborate variation of the original sound or does the single original image split into several different images, and if so might this be musically useful?
The potentially large number of outputs suggests that it would be practical to approach using this program by means of batch processing, perhaps controlled by scripting with a batch file or higher level scripting language. Scripting multi-sound playback, for example, would enable all the outputs to be heard in sequence, and this could then be extended to eliminating silent files or processing e.g., alternate files in different ways.
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Example command line to partition the spectrum:
specgrids specgrids mysnd.ana split.ana 8 8
(produces multiple output files: split0.ana, split1.ana, etc.)
inanalfile1 – input (mono) analysis file
outanalfilename – root name for a series of output analysis files
outfilecnt – the number of output spectral files to create
chan-grouping – the number of adjacent channels per group in the output spectraNB: The product of outfilecnt and chan-grouping must be an (even) divisor of the spectral channel-count, e.g., 512.
This program enables you to separate out different parts of a sounds spectrum and place these different parts over a multi-channel rig. First it divides the spectrum of a set of disjunct spectra. This means that the input analysis file needs to have distinctly different spectral features – heard as timbral colours.
Each output spectrum will contain some selection of the spectral channels of the source, while setting the other channels to zero amplitude. Between them, the complete set of output spectra contains all the channel info from the source.
Now you are in a position to resynthesise the different segment-streams separately and distribute the resulting soundfiles over a multichannel output. Thus the spectrum of the original sound is distributed around the multichannel space.
Let's look at this more closely. If the input file has 16 channels "abcdefghijklmnop":
- With 2 outfiles and chan-grouping = 1,
the outfiles contain "a-c-e-g-i-k-m-o" and "b-d-f-h-j-l-n-p" respectively.- With 2 outfiles and chan-grouping = 2,
the outfiles contain "ab-ef-ij-mn" and "cd-gh-kl-op".- With 2 outfiles and chan-grouping = 4,
the outfiles contain "abcd-ijkl" and "efgh-mnop".- With 2 outfiles and chan-grouping = 8,
the outfiles contain "abcdefgh" and "mnopijkl".- With 4 outfiles and chan-grouping = 1,
the outfiles contain "a-e-i-m" "b-f-j-n" "c-g-k-o" and "d-h-l-p".- With 4 outfiles and chan-grouping = 2,
the outfiles contain "ab-ij" "cd-kl" "ef-mn" and "gh-op".- With 4 outfiles and chan-grouping = 4,
the outfiles contain "abcd" "efgh" "ijkl" and "mnop".
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inanalfile – input frequency-analysis file outname – generic name for the (numbered) output files to generate. Files are named <outname>_0, <outname>_1,... etc.
minfrq – lowest frequency used for channel-permutations. (Range: 5-10000 Hz)
maxfrq – highest frequency used for channel-permutations. (Range: 150-10000 Hz)
NB: maxfrq must be at least 129.199 Hz more than minfrq.
-r – remove any other components.
SPECULATE progressively swaps channel data, following a permutation sequence like "Plain-Bob" bell-ringing. It produces an output file for each swap, so these typically number several hundred files, unless the frequency range is restricted (see minfreq and maxfreq).
The audio outputs start with the original sound and progressively become shriller (like successive high-pass filtering); once the sequence has passed the half-way point, this process is reversed and they become progressively more like the original again.
Setting the -r flag to "Remove other components" may reduce the number of output files. Every ten files or so will sound a little different, but it's hard to know how to handle such a vast amount of timbral variation.
SPECULATE is a powerful way of producing scores of timbral variants of the source sound, if you can handle the sheer number of outputs. You might for example use SFEDIT SPHINX to switch between some of them at various times.
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Example command line to create the squeeze effect:
specnu squeeze inanalfile outanalfile 341 0.4
inanalfile – input analysis file
outanalfile – output analysis file
centrefrq –frequency in Hz around which frequency data is squeezed
squeeze – the amount the spectrum is squeezed (< 1)All parameters may vary over time:
breakpoint files containing time centrefrq and time squeeze data columns separated by a space or a tab.
Note that the frequency at which the squeezing is to take place can be specified (centrefrq). The result will be some form of intensification (aural density) in that part of the sound. The degree of density will depend on the squeeze factor, higher factors yielding complex 'tones' in the midst of the overall sound.
At the time of writing, the program is producing output centred around 341Hz, only. However, combined with suitable transposition(s), e.g. using REPITCH TRANSPOSE, it might still be a useful way to colour a sound.
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sigfile – input analysis file with noise to be removed
noisfile – a sample of noise, cut from the original signal, containing only (a sample of) the noise you wish to remove from the sigfile outanalfile – output analysis file with noise removed
persist – minimum time (in ms.) for which the clean signal must persist above the threshold level if it is to be retained. (Range 0 to 1000 ms.)
noisgain – multiplies the threshold noise-levels found in the noisfile by an amount you specify, before they are used for comparison with the source file. (Range: 1 to 40).NB: Both input files are analysis files.
This process is similar to SPECNU CLEAN. SPECNU CLEAN assumes that, once your desired signal has reached an appropriate level it will mask any persisting background noise. With very noisy environments, this may not be sufficient, and the entire noise signal will need to be subtracted from the source. This process is only useful where the noise in the signal is persistent (e.g. the noise of an air-conditioning unit.)
This process is intended for cleaning up recordings made in noisy environments. It may be used alongside various editing and filtering processes, to clean a source sound bit by bit. These various functions are integrated in the Cleaning Kit option on the Music Testbed of the Sound Loom.
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The four SPECNU cleaning functions and several others are integrated in the
Cleaning Kit option on the Music Testbed of the
Sound Loom. The following screen indicates what is available:
The way these functions relate to the SPECNU programs is hidden to the user, as the Sound Loom Cleaning Kit builds the SPECNU functionality into a much broader context. The input to the Cleaning Kit is a normal soundfile, not an analysis file, and when a selection is made and the CLEAN button is activated, new dialogues appear which enable you to enter data and complete the processing.
More information is included in a page from Trevor Wishart's website: SoundLoom Cleaning Kit. This document contains a great deal of useful information about potential problem areas in sounds and how one might deal with them.