PVC
[Image]
INDEX
* INTRODUCTION
* INSTALLATION
UNPACKING
SETTING THE DESTINATION DIRECTORY
COMPILING THE BINARIES
* UNIX COMMAND
INFORMATION PAGE
SETTING FLAG VALUES
CONTROLLING PARAMETERS WITH FUNCTIONS
RUNNING COMMANDS WITH SHELL SCRIPTS
* ROUTINES: SHORT DESCRIPTIONS
BASIC ROUTINES
PLAINPV
TWARP
AMPLITUDE WARPING
NOISEFILTER
COMPANDER
SPECTWARPER
ADDITIVE SYNTHESIS
HARMONIZER
INHARMONATOR
CHORDMAPPER
SUBTRACTIVE SYNTHESIS
FILTER
FREQRESPONSE
CHORDRESPONSEMAKER
FILTRESPONSEMAKER
PVANALYSIS
TVFILTER
CONVOLVER
RESONANCE/REVERB
RING
RINGFILTER
RINGTVFILTER
NONLINEAR FREQUENCY DEVIATION
FILTDEVIATOR
TVFILTDEVIATOR
FEATURE EXTRACTION
ENVELOPE
CENTROID
FLUXOID
PITCHTRACKER
CONTROL FUNCTION PROCESSING
SHAPE
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* TERMS AND COMMON FEATURES
OVERLAP/ADD METHOD VS. OSCILLATOR BANK METHOD AND RESYNTHESIS
THRESHOLDS
SOURCE
MULTIPLE CHANNELS
PLAYBACK DURING PROCESSING
INPUT SOUND FILE
OUTPUT SOUND FILE
FLOATING-POINT AMPLITUDE RESCALING
OUTPUT STATISTICS
FREQIUENCY RESPONSE TERMINAL OUTPUT
ANALYSIS FILES
DECIBELS
LOW/HI SHELF EQUALIZATION
WARP INDEX
PITCH TRANSPOSITION
FREQUENCY SHIFT
ENVELOPE RESPONSE TIME
RING DECAY TIME
FFT SIZE
WINDOW SIZE
WINDOW TYPE
FRAMES PER SECOND
TIME EXPANSION/CONTRACTION
BEGIN/END TIMES
GAIN
FILTERING: SOURCE SIGNAL LEVEL
TRANSPOSITION/SHIFT APPLICATION FLAG
FILTER TYPES: PASS OR REJECT
RESPONSE FUNCTION SMOOTHING
ANALYSIS DATA ACCESS MODE
CONVOLVER PANPOT
FREQUENCY RESPONSE ACCUMULATION METHOD
RING ROUTINES: FILTER PLACEMENT
COMPRESSION AND EXPANSION
UTILITIES
FILE CONVERSION: aiffs, aiffd, nexts, nextd, nextfloats
FUNCTION VIEWING: showme, showspect
USING THE SHELL SCRIPTS
SAMPLE SCRIPT: S.PLAINPV
SHELL SCRIPT OUTPUT SOUND FILE CONVERSION
GEN FUNCTION CONTROL OF PARAMETERS
SAMPLE OUTPUT: S.PLAINPV
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* INTRODUCTION
PVC is a collection of phase vocoder signal processing routines and
accompanying shell scripts for use in the transformation and manipulation of
sounds. It is written in C and designed to be used in a UNIX environment. It
has come about as a result of my path of education and research into phase
vocoder technology. It follows in the spirit of the work by Eric Lyon (out
of which PVC is built) and Chris Penrose whose particular dsp research
springs from the coding and tutorial work of F.R. Moore and Marl Dolson.
Moore's book, Elements of Computer Music, published by Prentice Hall, is
therefore a great resource for making sense of the phase vocoder engine
which I am unable to go into here. Curtis Road's book, The Computer Music
Tutorial, published by MIT Press, has sections on the phase vocoder as well;
these may better introduce the beginner to the practical concerns of this
technology. Short of the explanations these sources provide, I have
attempted to offer below some explanations, particularly as needed for
control of the parameters in these routines. A manual and tutorial would be
great to have; unfortunately time has not yet made it so.
These routines reflect my need for tools which can perform different
spectral resynthesis tasks; both simple and experimental. Their refinement
has advanced with my growing skills and curiosity, which I expect will
continue as long as I have questions about sound. Most of these routines can
be viewed in terms of traditional additive or subtractive synthesis tasks,
coming about as they did from the desire for greater finesse and control of
these two basic types of synthesis. While the speculative nature of some
give them an idiosyncractic character, most should, with practice, reveal
the transparency of their names if not the role they can play in the shaping
of sound. All require a good ear tuned towards sound and idea as none of
these routines are automatic, although many hold great potential for the
diligent.
This 3.0 release contains only those routines which I think are stable,
useful and moderately transparent. Some earlier versions have been omitted,
replaced or consolidated into newer routines. For example, compander
remains, but the ideas behind bandamp have ripened into spectwarper, a
remarkable "super companding" tool for windowing amplitude, and balancing
the resonance/noise-residues of a sound. The harmonic tone reorganizer,
chordmapper, has continued to grow in its controls (however arcane),
offering increasingly subtle ways to reorganize harmonic spectra. The
noisefilter routine is now very good, having become a PVC first encounter
routine for many whose noisy lives cross my path. Tvfiltdeviator now joins
the arcane but novel filtdeviator routine. In addition, I have added a set
of feature analysis routines (pitchtracker, centroid, envelope, fluxoid);
which should be useful in generating function files to control different
synthesis strategies. There are other, more experimental routines (some
actually appeared in 2.0) which are still proving themselves; in time they
will appear or reappear. As with 2.0, floating-point files (combined with a
rescale feature) continue to be readable and writable. Someday I will deal
with AIFF headers (although they do not offer floating-point values), but
not for now.
Paul Koonce
koonce@music.princeton.edu
RETURN TO INDEX
----------------------------------------------------------------------------
INSTALLATION
The PVC package contains both my PVC routines, and the CMUSIC gen functions
written by F.R. Moore (each in a separate directory). I have included some
SGI shell script utilities as well, for easily changing sound file formats.
Moore's standalone gen functions are useful tools for creating function
files that can then be used in the time-varying control of parameters. The
gen functions included are cspline, gen0, gen1, gen2, gen3, gen4, gen5,
gen6, and genraw; a one-line summary can be obtained by running the command
without any arguments. A detailed explanation of each can be found in the
appendix of Moore's Elements of Computer Music.
You can compile and install the PVC and gen function routines separately or
together following unpacking and setting of the destination directory.
1) Unpack:
First move PVC.tar.gz to the directory of your choice. Unzip it with gunzip.
gunzip PVC.tar.gz
Then, unarchive it with tar.
tar xvf PVC.tar
This will produce a PVC directory in which you will find several other
directories.
2) Set Destination Directory:
Next you will need to set the destination directory in the Makefile located
in the /PVC directory. This is the master makefile for both the PVC and gen
routines. Change the directory specified by:
DESTDIR = /You_must_set_the_desination_directory
to the directory in which you want the routines installed.
3) Compile:
To compile and install the PVC and gen function routines both, type:
make
which if successful should be followed with:
make install
To compile and install only the PVC routines, type the following.
make PVC
make install
And to compile and install only the gen routines, type the following.
make GEN
make install
In all cases the make install moves the compiled routines from the /PVC/bin
directory to your specified destination directory. (The handful of utilities
in the UTILITIES directory are copied as well.) If the destination directory
is in your .cshrc path, and you have sourced your shell file, as in:
source .cshrc
you should be able to type any of the routines and see their flag
information page. Try typing:
plainpv
for example.
RETURN TO INDEX
----------------------------------------------------------------------------
UNIX COMMAND-LINE FORMAT
The routines are UNIX, command-line routines in the form of:
routine [flags] input_soundfile output_soundfile
At present the only soundfile formats excepted (both input and output) are
NEXT/SUN formant files in either 16-bit short samples, or 32-bit
floating-point samples. In PVC, the float form has a required rescale
function, which is the whole reason for using floats in this case. All
processing is done in floats. Control of each routine's parameters is done
through flags, as in:
plainpv -N1024 -p12 input.snd output.snd
In most cases, the parameter flag inputs allow you to specify either a
constant (i.e. -p12 ) or a function file (i.e. -p/tmp/pitch_change );
function files give you time-varying control over a parameter (see below).
In most cases, parameters are initialized to default values.
RETURN TO INDEX
INFORMATION PAGE
Information about any routine can be seen by typing the name of the routine
without any arguments/files. Typing:
plainpv
produces the following information about plainpv.
plainpv: generic phase vocoder with dynamic controls
plainpv [flags] [input file (16-bit shorts)] [output file (optional)]
(values in brackets denote defaults)
N: FFT length (must be a power of 2) [1024]
M: window size in samples (must be a power of 2) [2*FFT]
(0 will automatically set window to 2*FFT size or larger)
w: window type: 0 = hamming, 1 = rectangular
2 = Blackman, 3 = Bartlett triangular [0.]
4-12 = Kaiser windows for alpha = 4-12, respectively
(representative sidelobe levels for alpha:
4 = -30dB, 8 = -58 dB, 12 = -90 dB)
D: analysis frames per second [200]
I: time expansion/contraction factor [1.]
(duration = duration * factor, 1. = original time)
P: pitch transposition in semitones (func) [0]
a: frequency shift factor
(bin frequency adder, before -P )(func) [0.]
b: begin time in seconds [0.]
e: end time in seconds ( 0. = end of file) [0.]
C: resynthesis channel (1 -> ?) (0 = all) [0]
SHELF EQ:(post transpose/shift)
H: SHELF EQ: Low shelf gain in dB (func) [0.]
X: SHELF EQ: High shelf gain in dB (func) [0.]
m: SHELF EQ: Low shelf frequency in Hz (func) [200.]
R: SHELF EQ: High shelf frequency in Hz (func) [2000.]
W: warp index for reshaping magnitude response (func) [0.]
Values > 0 expand the dynamic range,
values < 0 compress the dynamic range.
A: gain in decibels (func) [0.]
l: envelope attack time (func) [0.]
L: envelope release time (func) [0.]
T: BRICKWALL FILTER TYPE: 0 = bandpass, not 0 = band reject [0]
f: frequency window: low boundary
(before -P and -a) (in Hz) [0.]
F: frequency window: high boundary
(before -P and -a)(in Hz) [Nyquist frequency]
p: amplitude reports print mode: 0 = off, 1 = on [0]
i: time interval between amplitude reports [.25]
_: OUTPUT FORMAT: 0 = taken from input file
1 = 16-bit integer, 2 = 32-bit floats [0]
=: PEAK RESCALE LEVEL (float output only) 0 to -96 dB
Set to 1 to rescale to level of input file. [ 1 ]
TERMINAL DISPLAY AND GRAPH FILE OUTPUT
n: number of frames [0]
u: low bin frequency [-1]
U: high bin frequency
(-1 = nyquist) [Nyquist frequency]
S: TERMINAL DISPLAY: display option [0]
(0 = off, 1 = phase data, 2 = amp data, 3 = both)
c: GRAPH FILE: WRITE ascii to FILE
0 = off, 1 = freq, 2 = decibels [0]
3 = decibels - waterfall plot
(When on, this flag writes ascii point pairs
(with time frame on x axis) for plotting
with gnuplot.)
d: TERMINAL DISPLAY FILE NAME for -c [./ascii.out]
t: oscillator resynthesis threshold in decibels [ -96 ]
RETURN TO INDEX
SETTING FLAG VALUES
If no output file is specifed, the name pv.out.snd will be used in the local
directory. The bracketed values at the end of each parameter represent the
default value; it can be changed by specifying the flag letter preceeded by
a minus sign and followed by the new value, with no spaces on either side.
For example, the following:
plainpv -N2048 inputfile outputfile
would change the FFT size to 2048. Some flags require files rather than
constants. For these, simply supply the full pathname of the needed file as
in:
twarp -F/here/there/everywhere/analysis_file
which supplies twarp with the necessary analysis file.
RETURN TO INDEX
CONTROLLING PARAMETERS WITH FUNCTIONS:
Parameters which have the word (func) on the info page just before the
default as in:
W: warp index for reshaping magnitude response (func) [0.]
can be controlled dynamically. This is done by providing a full pathname
file in place of the constant. The file is assumed to be a headerless series
of values representing how the parameter will evolve as a function of time.
The values may be either 32-bit floating-point values, or ASCII numbers,
arranged one-per-line (the routine deciphers which it is). The function file
can have any number of values as the series is fitted to the specified
duration, linearly interpolated to produce the values inbetween. Function
files in 32-bit floating-point form can be created with the CMUSIC gen
routines provided with this package. (See INSTALLATION.)
RETURN TO INDEX
RUNNING THE COMMANDS WITH SHELL SCRIPTS:
While all routines can be run at the commandline, they are most easily run
using the shell scripts found in the SCRIPTS directory. These scripts are
useful for saving and managing the parameters; in many ways they are a
poor-man's GUI. All scripts contain a top section for setting variables, and
a bottom section where those variables are placed into the commandline flag
structure and run. Some scripts perform two routines such as a short
analysis routine followed by the main synthesis routine, while others run
just one routine. The variables for the routines in a shell are set in the
top section. Take note that shell script variable assignments do not allow
for spaces. The numerous parameters, which in some routines run as high as
53, make these scripts a necessity. They will be your friend if you take
care to leave the bottom part alone, and don't corrupt your variable names.
Someday I will make a better way to interface with the routines; for now
this is the way it is.
To run the scripts, simply type the name of the file (or appropriate
pathname when running from outside the directory in which it resides). For
example:
S.plainpv
(If this does not work, check to make sure the script is executable, and
that the first line contains #!/bin/sh).
If things are working correctly, the resynthesis should begin which you will
know from the output streaming to the terminal.
(See the explanation below about using shell scripts. )
RETURN TO INDEX
----------------------------------------------------------------------------
ROUTINES: SHORT DESCRIPTIONS
Below is a listing of the routines contained in this release along with a
short description of what each does.
BASIC ROUTINES
PLAINPV
Plainpv is a basic phase vocoder with control of pitch transposition,
frequency shift, time scale, amplitude warp and low/high shelf equalization.
It also has some nice controls for looking at the data produced by the phase
vocoder. Run this routine with S.plainpv. If you are interested in looking
and/or graphing segments of the data, run the
S.plainpv_with_printout_and_graph_files script instead and use the showspect
utility. You will need to have gnuplot installed.
RETURN TO INDEX
TWARP
Twarp is like plainpv except that it works from an analysis file rather than
a soundfile. This allows you to move forwards/backwards through time
according to a time function file. Use pvanalysis through the script
S.pvanalysis to make the analysis file; then run the script S.twarp.
RETURN TO INDEX
NOISEFILTER
Noisefilter filters out the noise in a sound by subtracting out a frequency
response. The frequency response is analyzed from a short segment in the
file where noise alone is found. For sounds that do not have segments of
isolated noise, there is a threshold mode. Run with S.noisefilter.
RETURN TO INDEX
AMPLITUDE WARPING
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. The entire analysis/companding process (including the analysis
segment using freqresponse) can be run using the script S.compander.
RETURN TO INDEX
SPECTWARPER
Spectwarper uses an expanded compansion scheme to highlight either a sound's
stronger, resonant components or its weaker noise/residual components.
Spectwarper is fairly similiar to compander; however, unlike compander which
compands bins against the constant peak of an input response file,
spectwarper compands bins using a peak drawn (in the current frame) from a
narrow frequency band centered around the value being processed. This causes
the compansion or "warping' of the amplitudes to accentuate(expansion) or
mask(compression) formants located within the frequency bands; the result
being the noise/pitch highlighting mentioned earlier. Part of this comes
from the treatment of compression in Spectwaper. Unlike compander which only
reduces the amplitude above the threshold when compressing, spectwarper
reduces the amplitude of the entire range, becoming, in effect, an expander
of the strongest amplitudes that expands them (when the compression level is
severe) out of the picture. Spectwarper is one of my favorite routines of
late simply because it provides such a simple and powerful control over the
noise and pitch characteristics of a sound. I love it, and use it often. Run
this routine with S.spectwarper.
RETURN TO INDEX
ADDITIVE SYNTHESIS -- HARMONIZER, CHORDMAPPER, AND INHARMONATOR:
These routines all allow for a kind additive synthesis based on the
remapping of phase vocoder data according to some model. Each requires an
ascii data file specifying how phase vocoder information will be replicated
or mapped. This mapping is constant for the run of the routine.
HARMONIZER
Harmonizer works much like a commercial harmonizer in that it allows you to
create harmony against the source by adding a transposed copy of it. Here
the concept is extended by allowing for multiple harmonizations, each taken
from a different band of frequencies, output with seperate gain. Run this
using the script S.harmonizer.
CHORDMAPPER
Chordmapper lets you specify how harmonically related groups of partials
will be replicated or mapped to produce chords. An input data file organizes
the remapping into tone groups, and includes ways to tune or neutralize the
frequency deviations of partials. Time-varying control of these features is
available as well. You can use this routine to build up thick chords from
single tones, or to delicately reorganize a harmonic spectrum. Run this
using the script S.chordmapper.
INHARMONATOR
Inharmonator lets you specify how the partials of one fundamental will be
remapped or deviated. While the more recent and developed routine
chordmapper is probably better for this task, I have decided to leave this
routine in for now. (Think chordmapper.)
RETURN TO INDEX
SUBTRACTIVE SYNTHESIS
FILTER
Filter is a very useful routine for filtering a sound by a frequency
response. Filtering is achieved by first creating the frequency response
through either synthesis or analysis, followed by filtering with filter.
Synthestic responses are created using either chordresponsemaker (which
synthesizes a spectrum as a collection of harmonic tones), or
filtresponsemaker (which synthesizes a frequency response using lines and
breakpoints). Analyzed responses can be made with freqresponse (which
analyzes a sound file segment and constructs a response representing the
peak or average amplitudes). Once made, the magnitudes of the FFT response
are multiplied against the time varying magnitudes of the input sound's FFT.
Filter allows time-varying control of the response shape (warp),
transposition/shift, compansion, smoothing, and source/filter mix, making
this a very useful tool for quickly manipulating the spectral
characteristics of a sound according to your synthetic or analytic goals.
The synthetic forms can be run with the scripts
S.filter_with_chord_synthesis or S.filter_with_breakpoint_synthesis; the
analysis-based form with S.filter_with_analysis. The analytic form is a
powerful tool for bringing the color of one sound into the realm of another.
RETURN TO INDEX
FREQRESPONSE
Freqresponse is a routine used by several others to prepare a spectrum for
use with routines that filter, compress or limit. The response can be
normalized or not depending on the needs of the routine which will use the
response.
CHORDRESPONSEMAKER
Chordresponsemaker is a routine that uses a collection of harmonic tones,
variable in size, to create a synthetic frequency response. It is found in
various filtering scripts.
FILTRESPONSEMAKER
Filtresponsemaker is a routine that uses breakpoints and straight lines to
create a synthetic frequency response. It is found in various filtering
scripts.
PVANALYSIS
Pvanalysis is the time varying form of freqresponse that creates a phase
vocoder analysis for use by other routines. The routines which require
pvanalysis files are twarp, convolver, tvfilter, ringtvfilter, and
tvfiltdeviator. Run this using the script S.pvanalysis.
TVFILTER
Tvfilter is the time-varying (tv) form of filter. Tvfilter uses a pvanalysis
file to change the magnitudes of the input sound file. As it is with filter,
tvfilter multiplies the magnitudes of the analysis FFT against the
magnitudes of the input sound's FFT, while preserving the frequency/phase
characteristics of the input sound. Preserving the phase of the input sound
file results in a cross-synthesis which sounds like the input sound file
covered or suppressed by the shadow of the analysis file. Like filter,
tvfilter offers a variety of controls for manipulating the filter
characteristic. The use of a phase vocoder analysis to represent the filter
characteristic also makes possible the temporal control of the filter file
(i.e. backwards/forwards control) as found with twarp. Run this using the
script S.tvfilter.
RETURN TO INDEX
CONVOLVER
In its setup and controll, convolver is the same as tvfilter. It's
processing, however, is different. In tvfilter filtering is produced by
multiplying the magnitudes from the polar form of the two analyses; leaving
the phases (or frequencies) of the source intact while modifying the
amplitudes of those frequencies. Convolver goes a bit further by multiplying
the two analyses in their Cartesian forms. This produces an intersection of
the two spectra. Unlike tvfilter which produces a shadowlike intersection,
shadowing the analysis file characteristic onto the input sound file,
convolver creates a true spectral intersection, allowing only that which is
common to both sounds to be heard. The effect is a sound which is somewhat
garbled as it outputs the more intermittently common spectral components of
the two. The form of the multiplication in convolver does not allow some of
the filter transposition controls associated with tvfilter. There is however
a convolution panpot which offers control of the mix between the convolution
and source sounds. Run this using the script S.convolver.
RETURN TO INDEX
RESONANCE/REVERB
RING
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 which can be positioned to filter either the source input to the
feedback loop, or the feedback return. Run this using the script S.ring.
RINGFILTER
Ringfilter marries filter with ring by allowing a frequency response to be
imposed on the resonance created with ring. Ringfilter begins to look more
like multiple-delay, comb filter resonance since the static frequency
response selects which frequencies will feed back. What is unique here is
that the frequency response can come from an analysis, allowing the input
sound to be resonated by the average spectral characteristic of another
sound. A synthesized frequency response can be used as well. Like the EQ in
ring, the filter in ringfilter can be positioned to either filter the source
input to the feedback loop, or the feedback return where it will have the
effect of introducing the filter characteristic more slowly through the
resulting variable rates of decay. Run ringfilter with
S.ringfilter_with_chord_synthesis to create a synthetic frequency repsonse,
and with S.ringfilter_with_analysis for an analyzed frequency response.
RINGTVFILTER
Ringtvfilter is to ringfilter what tvfilter is to filter; that is, it makes
the filter in ringfilter time-varying. This is a sophisticated idea, that
is, time-varying filtering of the resonance of a time-varying sound. The
best characterization would be to say that Ringtvfilter imprints the shadow
of one sound onto the reverb of another. Ringtvfilter requires some thought
and finese in order to separate and articulate the evolutions of the source,
resonance, and filter. The best results are created using dynamic,
high-profiled source sounds, rich with transient noise; and more constant,
pitch/harmonic sounds for the time-varying filter. Like tvfilter,
ringtvfilter requires an analysis file. Run this routine using
S.ringtvfilter.
RETURN TO INDEX
NONLINEAR FREQUENCY DEVIATION
FILTDEVIATOR
The idea behind filtdeviator is to use a frequency response function to not
only filter a sound (as with filter), but to 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. Their 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. The
scripts to run filtdeviator -- S.filtdeviator_with_ chord_synthesis and
S.filtdeviator_with_analysis -- are designed with frequency response
synthesis/analysis sections like those for filter and ringfilter. Run this
routine using either S.filtdeviator_with_analysis or
S.filtdeviator_with_chord_synthesis.
TVFILTDEVIATOR
Tvfiltdeviator is to filtdeviator what tvfilter is to filter; i.e. it uses a
time-varying filter response in place of the constant one. This routine
blows the lid off of what was unusual about tvfiltdeviator. It's great for
making wacky sounds out of ones with nice, fixed harmonies. The best use is
to use it to deviate itself. Try taking something like a harpsichord or
guitar (pitched stuff with decay) and do an analysis of the sound with
pvanalysis. Then use the analysis to deviate the same sound. What happens is
the strength of each of the sound's components becomes a control over the
frequency deviation of that component, one that causes the sound to go
"sproing" whenever it has any amplitude. Makes tonal music sound really
broken. Run this routine with tvfiltdeviator.
RETURN TO INDEX
FEATURE EXTRACTION
ENVELOPE
Envelope is a routine for tracking the amplitude envelope of a sound. Output
can be ASCII, floats or a NeXT soundfile. Selecting floats or ASCII will
produce a file suitable for use in the control of a parameter. Run this
routine with S.envelope.
CENTROID
Centroid is a routine for tracking the centroid of a sound. The centroid is
the average of all the frequencies weighted by their amplitudes. It
essentially gives you a kind of center frequency value for your spectrum.
The analysis can be restricted to a band of frequencies, allowing the
centroid to track a particular frequency component (although pitchtracker
can do this as well). Selecting floats or ASCII will produce a file suitable
for use in the control of a parameter. Run this routine with S.centroid.
FLUXOID
Fluxoid is a routine for tracking the average frequency change of a sound.
The average can be weighted (best) or not by the amplitudes. Selecting
floats or ASCII will produce a file suitable for use in the control of a
parameter. Run this routine with S.fluxoid.
PITCHTRACKER
Pitchtracker is a routine for tracking the fundamental pitch trajectory of a
sound. It is an experimental routine that works, I believe, but forever has
its quirks. Three detection methods are available for following the 1)
fundamental of the harmonic collection, 2) the strongest formant, or 3) a
band-limited centroid. Different output formats let you see, hear and
eventually use the fruits of your pitch tracking. Run this routine with
S.pitchtracker.
RETURN TO INDEX
CONTROL FUNCTION PROCESSING
RESHAPE
Reshape is a routine for transforming function streams to meet the needs of
different parameters. It takes a headerless float or ASCII function file as
input and outputs a headerless stream of float or ASCII values. With the
appropriate flags, it can be used to limit, resample, translate, warp,
expand, shrink, invert, quantize, and lowpass filter the input values. The
output can be translated into different amp or pitch units depending on your
needs. Run reshape at the command line.
RETURN TO INDEX
----------------------------------------------------------------------------
TERMS AND COMMON FEATURES
Below are various terms, parameters, or ways of doing things which are
common to many of the routines.
OVERLAP/ADD VS. OSCILLATOR BANK METHODS AND RESYNTHESIS THRESHOLDS:
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 which 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.
SOURCE
The source sound is the original input sound. Some routines allow for the
mix of the processed sound with the original source sound.
MULTIPLE CHANNELS
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. Prcessing one channel at a time
requires less memory and allows you to audition the output sooner than if
you did all channels at once.
INPUT SOUND FILE
The input sound file must be a NeXT/Sun format sound file in either 16-bit
short or 32-bit floating point format. It may have one or more channels.
OUTPUT SOUND FILE
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.
RETURN TO INDEX
FLOATING-POINT AMPLITUDE RESCALING
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.
PLAYBACK DURING PROCESSING
The header of the output soundfile is updated often, so if your peak
amplitude has not exceeded the 16-bit limit of the converters, you may play
the float or integer output file before processing has completed.
OUTPUT STATISTICS
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. Wth integer format ouput files, ouput 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, ouput 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.
RETURN TO INDEX
FREQUENCY RESPONSE TERMINAL OUTPUT
In many filtering or companding routines, a crude terminal print of the
frequency response is a available. A flag sets the high cutoff frequency for
this output; a value of 0 (0 Hz) turns printing off.
ANALYSIS FILES
Analysis files are binary, 32-bit floating-point files written by
pvanalysis, containing frames of FFT analysis data for one or more channels.
Analysis file data is preceeded by a header containing information about the
analysis. Analysis files are much larger than the sound files they
represent, and increase in proportion to the FFT size used. As such, files
can become very large, so it is advisable to only make them when needed
unless you have disk space to spare.
RETURN TO INDEX
DECIBELS
Amplitude is always handled in decibel units. The greatest magnitude of the
16-bit short integer is equated with an amplitude of 1.0 or 0 dB. 0 dB
functions as unity gain, and the peak amplitude in issues of compression,
expansion, and amplitude windowing. 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. 16-bit shorts allow for a 96 dB
dynamic range. Take care not to loose signal level as a consequence of
processing since quantization noise will emerge when you attempt to regain
your signal level by amplifying the integer sound file.
LOW/HI 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.
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.
RETURN TO INDEX
PITCH TRANSPOSITION
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.
FREQUENCY SHIFT
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.
RETURN TO INDEX
ENVELOPE RESPONSE TIME
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 the residual noise of a sound.
RING DECAY TIME
Decay time is an issue in the feedback of the ring routines. Like response
time, it is the time it takes the signal to decay to -60dB of its former
state, or better, the time it takes the reverb to decay to -60dB.
RETURN TO INDEX
FFT SIZE:
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.
WINDOW SIZE
The window size is a less opaque parameter; like the FFT, it must be a power
of 2. Windows which are 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, a feature I have always
used.
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 reccomended 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.
RETURN TO INDEX
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 which
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.
TIME EXPANSION/CONTRACTION
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.
RETURN TO INDEX
BEGIN/END TIMES
Processing may be performed on an entire file or a segment of it by
specifying begin and end times. End times less than or equal to 0 default to
the end of the input file.
GAIN:
The output and other components can be gained. 0 dB represents unity gain,
no change. See decibels.
FILTERING: SOURCE SIGNAL LEVEL
The mix of source and filtered sounds in the filter routines can be
controlled by the source decibels floor. This value, taken from the -96 to 0
dB range, specifies the level of the source signal. The filtered signal
level is equal to (1 - source amplitude floor). 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.
RETURN TO INDEX
TRANSPOSITION/SHIFT APPLICATION FLAG
Filter routines which allow for transposition and frequency shifting of both
filter and source have a flag which specifies whether transposition/shift
should be applied before or after filtering. If it is applied before, the
pitch transposition trajectory will evolve independent of the filter's
trajectory of transposition. If it is applied after, then the pitch
transposition trajectory will be added to the filter transposition
trajectory, causing the filter to move in parallel with the pitch
transposition movements plus any movements the filter transposition
parameter adds.
FILTER TYPES: PASS OR REJECT
Filters can be toggled to use frequency responses in pass or rejection mode.
In pass mode, the response's stronger magnitudes are used to pass source
through the filter; in rejection mode, to impede or reject components. In
rejection mode, the response is created by inversion in the decibel range,
not amplitude. In time-varying filtering (tvfilter), rejection can be in
mode 1 in which the response is inverted against a constant 0 dB peak, or in
mode 2 in which the response is inverted against the current analysis
frame's peak amp. Spectral warping is always applied after the response has
been transformed by rejection.
RETURN TO INDEX
RESPONSE FUNCTION SMOOTHING
Many routines which use frequency response files to filter or warp
amplitudes have a control which allows the response to be smoothed. The
smoothing is produced 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 manipulation of a bandwidth value, specified in octave
units. Larger bandwidths produce greater degrees of smoothness, 0 turns
smoothing off.
ANALYSIS DATA: ACCESS MODES
Routines which use analysis data made with pvanalysis -- twarp, convolver,
tvfilter, ringtvfilter, and tvfiltdeviator) -- access data the same; using
the time-point, rate, and data window boundary parameters, set to function
in either rate or explicit mode. In rate mode, the rate determines the speed
of movement through a data file; the time-point sets the starting position.
The rate may be positive (forward in time) or negative (backwards in time),
and vary according to a function. Explicit mode uses the time point
parameter to specify exactly where the analysis data should come from (units
here are in the time of the analyzed sound). (Explicit mode does not use the
rate control, and makes sense only if the time-point is controlled with a
function.) Both rate and explicit modes abide by the upper and lower data
window boundaries which delimit the data range. When the time-pointer moves
beyond the specified upper and lower time boundaries, it re-enters the
window from the other end, making the window into a circular/modular
structure. The boundaries can be controlled with functions as well, giving
this mode an expressive dimension far surpassing the time
expansion/contraction parameter. There is also an auto-stop feature that,
when turned on, causes processing to stop when it reaches the end of the
analysis.
RETURN TO INDEX
CONVOLVER PANPOT
The convolver routine has a unique panpot mechanism for controlling the mix
of input sounds (A and B) with their convolution. The panpot is a crossfade
mechanism that uses a -1 to 1 control range to accentaute either sound A, B
or their convolution. A value of -1 produces an output consisting entirely
of sound A, a value of 1, sound B. The 0 between these extremes produces the
convolution of A and B. Values between these points produce a crossfade mix
of either A or B and the convolution. For example, a trajectory from -1 to 1
would crossfade from sound A into the convolution, and on to sound B.
Separate gain controls for A, B and the convolution make it possible to tune
the continuity of this trajectory. In addition, the presence or spread of
the convolution into the crossfade range can be tuned with the domain warp
controls. The domain warp reshapes the movement through the crossfade range,
allowing you to create a more gradual approach from A or B into the
convolution center. This is achieved through a simple nonlinearizing of the
crossfade domain in warp index style. Increasingly positive domain warp
values (specified independently for each side) transform the linear
trajectory towards the convolution into a decellerating one, causing the
subtle mix area around 0 to be expanded. Therefore, if you want to hear more
convolution in your crossfade, increase the panpot domain warps.
FREQUENCY RESPONSE ACCUMULATION METHOD
Several of the response-producing routines have the option of accumulating
the response by either peak or average means. Whereas peak responses
represent the record of a sound's thresholds (or synthesis specification's
highest values), average responses represent the most common
characteristics. If the sound you are analyzing has intermittent moments of
sound whose peak characteristics you wish fully represented in the response,
use the peak mode; otherwise use the average.
RETURN TO INDEX
RING ROUTINES: FILTER PLACEMENT
Ringfilter and ringtvfilter use frequency response functions to filter the
reverb. Two filtering modes are available in which either the source input
to the feedback if filtered, or the feedback. When the response is used to
filter the source input, it filters the signal before it enters the feedback
mechanism, imposing its characteristic, from the start, on the feedback.
However, when positioned to filter the feedback component, the appearance of
the respsonse's spectral characteristic, in the reverb, appears gradually as
the signal decays. In this mode, the time it takes the signal to decay into
the response characteristic is controlled by an additional decay time
associated with the filter.
COMPRESSION AND EXPANSION
Spectral compression and expansion play a role in many routines. Its
implementation here is according to the traditional model that uses
thresholds and magnitudes of compression/expansion to reduce or enlarge the
dynamic range of a signal. With spectral compression, amplitudes that exceed
the specified compression threshold are reduced by an amount determined by
the decibels of compression (a multiplier of the bin's amplitude lying above
the threshold). Expansion works in a similar fashion, except that it changes
the amplitudes below, rather than above, the expansion threshold; this
results in an expansion of the dynamic range as the bins falling below the
threshold are made to cover a wider range.
The term companding or compander is a merging of the two names, useful in
situations where they are both available. While compander is the most
obvious example of a routine using companding, traditional compression can
be found in several other routines that involve filtering. It is not
uncommon, in those routines, to reduce the dynamic range of an analyzed
frequency response, particularly if it is time-varying, since the goal in
filtering is more about color than dynamic range.
In all routines that use some form of companding, the dynamic range of the
unprocessed signal/response is assumed to lie between 0 and -96 dB;
thresholds are chosen from within this range. The degree of compression or
expansion, expressed in decibels, represents how much the signal lying
beyond the threshold will be reduced. A value of -6 dB would halve the
dynamic range above the threshold in compression, or double the range below
the threshold in expansion.
Compander applies compression for each frequency bin separately rather than
as a macro gain change. It does this by using a frequency response file,
created with freqresponse, to establish a unique, 0 dB point of reference
for every bin; using its unique point of reference, every bin is compressed
or expanded.
RETURN TO INDEX
----------------------------------------------------------------------------
UTILITIES
FILE CONVERSION: aiffs, aiffd, nexts, nextd, nextfloats
The sound file conversion scripts: aiffs, aiffd, nexts, nextd, and
nextfloats are shell scripts available for converting sound files back and
forth between aiff and next formats, or from next to floats. They are all
effectively SGI scripts since they use the SGI sound file format conversion
utility, sfconvert. Aiffs and aiffd take next integer files and write new
aiff files, nexts and nextd the opposite; in addition aiffs and aiffd can be
used to write new aiff integer files converted from next float files.
Nextfloats writes a new float file from a next integer file.The s or d
following the aiff or next in the name stands for the action taken on the
original file once the new file is made; the s saves the original file (i.e.
does not delete it), the d causes it to be deleted. Multiple files may be
converted with the same run of the command. Running the command without any
input files will produce a description of the routine.
FUNCTION VIEWING: showme, showspect
Two graphing scripts are available for viewing functions and spectral data.
You must have gnuplot installed on your computer to use them (Type gnuplot
<CR> to see if you do). Showme is a simple script for viewing function
files. Run without an input file for a description. Showme takes headerless
floating-point or ASCII (give -a flag) function files and plots them.
Showspect plots the file of FFT amplitude or frequency data produced by the
plainpv script, S.plainpv_with_printout_and_graph_files. Showspect is useful
for seeing a graphic representation of a very particular part of an
analysis, it is not a substitute for a standard spectrogram application.
RETURN TO INDEX
----------------------------------------------------------------------------
USING THE SHELL SCRIPTS
SAMPLE SCRIPT: S.plainpv
Below is a copy of the complete script for running plainpv which you can
examine here to understand the basic structure of this shell script
mechanism. In it you find a top section for variables and a bottom section
for execution of the plainpv command. Set the variables of the top section
with the appropriate files and constants; do not use spaces. To run the
script, simply type it as a command as in the following.
S.plainpv
(See Running Commands with Shell Scripts)
When the shell runs the script, it copies the value of the assigned
variables into the respective flag positions and runs the command as if you
had typed it at the prompt in a shell window. Each shell script is set up to
print the command to the terminal just before running it. A sample of the
output follows the script below.
SHELL SCRIPT OUTPUT SOUND FILE CONVERSION
The conversion routine, aiffd, has been added at the end of every script
that writes an output sound file, as seen below.
# aiffd $output_file ;
To prevent it from unexpectedly converting the output of your sound file, it
has been commented out with the # sign. If you would like your next integer
or floating-point output files to be automatically converted to aiff sound
file format, simply remove the # sign. (See file conversion.)
GEN FUNCTION CONTROL OF PARAMETERS
Any parameter whose flag on the routine's information page has the word
(func) after it can be controlled by a function file. An easy way of doing
this is to generate the file when the script is run. To do this, simply
place the call to the gen function anywhere in the top section of the
script; the script will then run the command and make the file needed by the
shell variable. By the time the PVC routine is run in the bottom section,
the file will be ready.
The best place to locate your gen function calls is immediately following
the variable assignment, as in the example below which sets the variable
controlling the pitch transposition.
pitch_transposition_in_semitones=/tmp/ptrans
gen4 -L1000 0 -3 0 1 3 > /tmp/ptrans ;
In this example, the variable, pitch_transposition_in_semitones, is set with
the file name, /tmp/ptrans, which has in it the 1000 values output by the
gen4 command.
While the output of the gen4 command is in 32-bit floating point values, PVC
allows them to be ASCII as well. So, it would be possible to replace the
segment of text above with the following.
pitch_transposition_in_semitones=/tmp/ptrans
echo 0 1 2 2.5 5 10 > /tmp/ptrans ;
Here the gen4 call has been replaced with a simple call to echo which places
the ASCII values, 0 1 2 2.5 5 10, into the file /tmp/ptrans. When PVC looks
at the file, it figures out that it is text rather floats.
While the values in both cases are linearly interpolated by PVC, to create
the continuous function needed by the routine, the gen4 case would be
smoother since it has more values.
Lines in shells can be continued onto new lines with the backslash, which
comes in handy with gen functions. The above, for example, could be entered
as:
gen4 -L1000 \
\
0 -3 0 \
\
1 3 \
\
> /tmp/ptrans ;
which would simplify our parsing of it.
RETURN TO INDEX
#!/bin/sh
#******************************************************
#.................... PLAINPV .........................
#******************************************************
#******************** OUTPUT **************************
output_file=/S1/cm.mix.snd
#......................................................
output_data_format=1
#( 0: Same as input file )
#( 1: integers )
#( 2: rescaled floats )
#................ RESCALE .............................
rescale_level_in_decibels=1
#(-96 to 0 dB. Set to 1 to rescale to peak of input file.)
#******************** INPUT ***************************
input_file=/S1/t.snd
#........ BEGIN/END TIMES .............................
begintime=0
endtime=0
# (End time of 0 or less defaults to end of file.)
#======================================================
#*** ANALYSIS PARAMETERS ******************************
FFT_length=1024
window_type=2
windowsize=0
# (0 sets windowsize to 2 * FFT (or larger))
frames_per_second=200
#======================================================
#*** RESYNTHESIS PARAMETERS ***************************
#........... OUTPUT CHANNEL(S) .......................
output_channel=0
# (channels are numbered from 1 to the maximum.)
# (0 = all channels)
#.............OSCIL THRESHOLD ........................
oscillator_resynthesis_threshold_in_dB=-96
#( Try -60 to -70 unless dropouts become audible. )
#****************** MODIFICATIONS *********************
#.................. TIME ..............................
time_expansion_contraction_factor=1
# (Adjust frames_per_second in proportion to keep a
# constant rate.)
#.................. DECIBELS ..........................
gain_in_decibels=0
#.................. PITCH .............................
frequency_shift_in_Hz=-0
pitch_transposition_in_semitones=0
#............ AMPLITUDE RESPONSE ......................
release_time_in_seconds=0
attack_time_in_seconds=0
#............ SPECTRUM WARPSHAPE ......................
spectrum_warpshape_index=0
#............ BRICKWALL FILTER ........................
FILTER_TYPE=0
#( 0 = bandpass )
#( 1 = bandreject )
#......................................................
BRICKWALL_FILTER_window_low_frequency=-1
BRICKWALL_FILTER_window_high_frequency=-1
# (-1 selects respective lowest or highest frequency)
#======================================================
#*************** LOW/HIGH SHELF EQ *********************
LOW_SHELF_EQ_gain_in_decibels=0
LOW_SHELF_EQ_frequency=200
HIGH_SHELF_EQ_gain_in_decibels=0
HIGH_SHELF_EQ_frequency=2000
#********** AMPLITUDE STATISTICS **********************
print_amplitude_statistics_0_no__1_yes=1
amplitude_statistics_time_interval=.25
#======================================================
#========= SCRATCH SPACE ==============================
#======================================================
#====================================================
# COMMAND LINE SETUP -- OFFICE USE ONLY
# (DO NOT WRITE BELOW THIS LINE)
#====================================================
pvroutine=plainpv
PVFLAGS="\
\
-N$FFT_length \
-M$windowsize \
-w$window_type \
-D$frames_per_second \
-I$time_expansion_contraction_factor \
\
-a$frequency_shift_in_Hz \
-P$pitch_transposition_in_semitones \
-A$gain_in_decibels \
\
-C$output_channel \
-t$oscillator_resynthesis_threshold_in_dB \
\
-b$begintime \
-e$endtime \
\
-H$LOW_SHELF_EQ_gain_in_decibels \
-m$LOW_SHELF_EQ_frequency \
\
-X$HIGH_SHELF_EQ_gain_in_decibels \
-R$HIGH_SHELF_EQ_frequency \
\
-L$release_time_in_seconds \
-l$attack_time_in_seconds \
\
-W$spectrum_warpshape_index \
\
-T$FILTER_TYPE \
-f$BRICKWALL_FILTER_window_low_frequency \
-F$BRICKWALL_FILTER_window_high_frequency \
\
-_$output_data_format \
-=$rescale_level_in_decibels \
\
-p$print_amplitude_statistics_0_no__1_yes \
-i$amplitude_statistics_time_interval \
"
echo "\n\n$pvroutine $PVFLAGS $input_file $output_file "
$pvroutine $PVFLAGS $input_file $output_file
RETURN TO INDEX
----------------------------------------------------------------------------
SAMPLE OF OUTPUT FROM S.PLAINPV
Below is a sample of the output from S.plainpv.
plainpv -N1024 -M0 -w2 -D400 -I2 -a-0 -P2 -A0 -C0 -t-96 -b0 -e0 -H0 -m200
-X0 -R2000 -L0 -l0 -W0 -T0 -f-1 -F-1 -_1 -=1 -p1 -i.25 /S1/t.snd /S1/cm.mix.snd
/////////////////////////////////////////////////////////////////////
---------------------------------------------------------------------
============================== PLAINPV ==============================
---------------------------------------------------------------------
========================== INPUT SOUNDFILE ==========================
INPUT FILE: FILENAME = /S1/t.snd
INPUT FILE: SAMPLE RATE = 44100
INPUT FILE: NUMBER OF CHANNELS = 2
INPUT FILE: DURATION = 2.770386
INPUT FILE: BEGIN TIME = 0.000000
INPUT FILE: END TIME = 2.770386
INPUT FILE FORMAT: 16-BIT INTEGER
========================== OUTPUT SOUNDFILE =========================
OUTPUT FILE: FILENAME = /S1/cm.mix.snd
OUTPUT FILE: SAMPLE RATE = 44100
OUTPUT FILE: NUMBER OF CHANNELS = 2
OUTPUT FILE FORMAT: 16-BIT INTEGER
OUTPUT FILE: DURATION = 5.540771
======================== ANALYSIS PARAMETERS ========================
FFT SIZE = 1024
*
FUNDAMENTAL ANALYSIS FREQUENCY = 43.066406
*
WINDOW SIZE = 2048
FRAMES/SECOND = 400
DECIMATION SAMPLES (samples between analysis frames) = 110
======================= RESYNTHESIS PARAMETERS ======================
TIME EXPANSION/CONTRACTION FACTOR = 2
*
INTERPOLATION SAMPLES (samples between resynthesis frames) = 220
*
OSCILLATOR RESYNTHESIS THRESHOLD (in dB) = -96.000000
*
GAIN (in dB) = 0.000
PITCH TRANSPOSITION (in semitones) = 2.000
FREQUENCY SHIFT (in Hz) = 0.000
*
ENVELOPE ATTACK TIME (in seconds) = 0.000
ENVELOPE RELEASE TIME (in seconds) = 0.000
*
SPECTRUM WARPSHAPE INDEX = 0.000
*
FREQUENCY WINDOW: LOW BOUNDARY = 0.000000
FREQUENCY WINDOW: HIGH BOUNDARY = 22050.000000
*
*............. LOW/HIGH SHELF EQ............*
LOW SHELF FREQUENCY = 200.000
.......... LOW SHELF DECIBELS = 0.000
HIGH SHELF FREQUENCY = 2000.000
.......... HIGH SHELF DECIBELS = 0.000
*...........................................*
*
=====================================================================
ANALYSIS: CHANNEL = 1
..............USING BLACKMAN WINDOW
.....USING OSCILLATOR BANK RESYNTHESIS
*********************************************************************
** PEAK AMPLITUDE STATISTICS **
*********************************************************************
TIME PEAKAMP DECIBELS (LAST DECIBELS PEAK)
*********************************************************************
( 0.00 - 0.25) 0.0005 -66.295 -66.295
( 0.25 - 0.50) 0.2052 -13.754 -13.754
( 0.50 - 0.75) 0.3285 -9.668 -9.668
( 0.75 - 1.00) 0.3066 -10.269
( 1.00 - 1.25) 0.3176 -9.962
( 1.25 - 1.50) 0.2731 -11.275
( 1.50 - 1.75) 0.2655 -11.518
( 1.75 - 2.00) 0.2416 -12.337
( 2.00 - 2.25) 0.2930 -10.661
( 2.25 - 2.50) 0.2915 -10.707
( 2.50 - 2.75) 0.3067 -10.267
( 2.75 - 3.00) 0.4094 -7.757 -7.757
( 3.00 - 3.25) 0.3076 -10.241
( 3.25 - 3.50) 0.2841 -10.930
( 3.50 - 3.75) 0.2843 -10.924
( 3.75 - 4.00) 0.3241 -9.786
( 4.00 - 4.25) 0.3340 -9.524
( 4.25 - 4.50) 0.3612 -8.845
( 4.50 - 4.75) 0.3113 -10.136
( 4.75 - 5.00) 0.3094 -10.189
( 5.00 - 5.25) 0.3141 -10.058
( 5.25 - 5.50) 0.1142 -18.846
============= PEAK AMPLITUDE ========================================
CHANNEL TIME PEAKAMP DECIBELS (CLIPPED SAMPLES)
.....................................................................
1 2.898 0.4094 -7.757
*********************************************************************
=====================================================================
ANALYSIS: CHANNEL = 2
..............USING BLACKMAN WINDOW
*********************************************************************
** PEAK AMPLITUDE STATISTICS **
*********************************************************************
TIME PEAKAMP DECIBELS (LAST DECIBELS PEAK)
*********************************************************************
( 0.00 - 0.25) 0.0004 -67.948 -67.948
( 0.25 - 0.50) 0.2301 -12.763 -12.763
( 0.50 - 0.75) 0.2477 -12.122 -12.122
( 0.75 - 1.00) 0.1969 -14.115
( 1.00 - 1.25) 0.2631 -11.599 -11.599
( 1.25 - 1.50) 0.2086 -13.613
( 1.50 - 1.75) 0.2559 -11.840
( 1.75 - 2.00) 0.2671 -11.465 -11.465
( 2.00 - 2.25) 0.2768 -11.157 -11.157
( 2.25 - 2.50) 0.1762 -15.082
( 2.50 - 2.75) 0.2113 -13.502
( 2.75 - 3.00) 0.2549 -11.872
( 3.00 - 3.25) 0.2673 -11.460
( 3.25 - 3.50) 0.2869 -10.847 -10.847
( 3.50 - 3.75) 0.2841 -10.931
( 3.75 - 4.00) 0.1991 -14.019
( 4.00 - 4.25) 0.2131 -13.427
( 4.25 - 4.50) 0.2540 -11.904
( 4.50 - 4.75) 0.2235 -13.014
( 4.75 - 5.00) 0.2407 -12.369
( 5.00 - 5.25) 0.2941 -10.629 -10.629
( 5.25 - 5.50) 0.1166 -18.667
============= PEAK AMPLITUDE ========================================
CHANNEL TIME PEAKAMP DECIBELS (CLIPPED SAMPLES)
.....................................................................
2 5.103 0.2941 -10.629
*********************************************************************
=====================================================================
PEAK AMPLITUDES: ALL CHANNELS
---------------------------------------------------------------------
CHANNEL TIME PEAKAMP DECIBELS (CLIPPED SAMPLES)
.....................................................................
1 2.898 0.4094 -7.757
2 5.103 0.2941 -10.629
=====================================================================
PLAINPV: RESYNTHESIS COMPLETED
