Difference between revisions of "Download Nyquist Plug-ins"
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==Audio Generators== | ==Audio Generators== | ||
− | '''Binaural Tones with Surf 2''' (bitone2.ny) {{external|[http://audacity.sourceforge.net/nyquist/bitone2.ny View]}} | {{external|[http://audacity.sourceforge.net/nyquist/bitone2.zip Download]}} | {{external|[http://audacity.sourceforge.net/nyquist/bitone21.mp3 MP3 example clip]}} | + | '''<font color="#FF6600">Binaural Tones with Surf 2''' </font>(bitone2.ny) {{external|[http://audacity.sourceforge.net/nyquist/bitone2.ny View]}} | {{external|[http://audacity.sourceforge.net/nyquist/bitone2.zip Download]}} | {{external|[http://audacity.sourceforge.net/nyquist/bitone21.mp3 MP3 example clip]}} |
A sinewave tone of one constant frequency is generated in the left channel of a stereo track, and a series of changing tones of slightly different frequencies are generated in the right. In addition, a stereo "surf" noise is generated. The differences between the left- and right-channel frequencies are termed "beat frequencies". | A sinewave tone of one constant frequency is generated in the left channel of a stereo track, and a series of changing tones of slightly different frequencies are generated in the right. In addition, a stereo "surf" noise is generated. The differences between the left- and right-channel frequencies are termed "beat frequencies". | ||
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Variables: | Variables: | ||
− | #''Left channel tone frequency'' from 50 Hz to 1000 Hz (default 100 Hz). | + | #'''Left channel tone frequency:''' from 50 Hz to 1000 Hz (default 100 Hz). |
− | #''Beat frequency [Hz], duration [minutes], time to change to next beat frequency [minutes]'' | + | #'''Beat frequency [Hz], duration [minutes], time to change to next beat frequency [minutes]:''' There are six of these edit fields in which you may enter up to three indicated values, separated by a space. The first of these edit fields has default values of 17.5 0.25 0.25. In the sixth field you may enter a final beat frequency and duration of that frequency. If you enter a only a single value into any of these fields, the duration of that beat frequency will be zero. If you leave any of these edit fields blank they will be ignored. |
− | #''Adjust total time'' between 1 and 60 minutes [0=no adjustment]. | + | #'''Adjust total time:''' between 1 and 60 minutes [0=no adjustment]. |
− | #''Fade-in and fade-out times'' [seconds]. Sets the time for fading in and fading out the volume at the start and end of the generated audio. | + | #'''Fade-in and fade-out times:''' [seconds]. Sets the time for fading in and fading out the volume at the start and end of the generated audio. |
− | #''Stereo surf frequency'' from 0 Hz to 2 Hz (default 0.1 Hz). If this setting is above zero, the surf sound will be panned back and forth somewhere between the left and right audio channels at the specified frequency, how far depending on the sixth variable: | + | #'''Stereo surf frequency:''' from 0 Hz to 2 Hz (default 0.1 Hz). If this setting is above zero, the surf sound will be panned back and forth somewhere between the left and right audio channels at the specified frequency, how far depending on the sixth variable: |
− | #''Stereo surf spread'' [percent] between 0 and 100 percent (default 80%). The larger this number, the further the surf sound will move away from the center pan position (0% results in the surf sound remaining in the center). | + | #'''Stereo surf spread:''' [percent] between 0 and 100 percent (default 80%). The larger this number, the further the surf sound will move away from the center pan position (0% results in the surf sound remaining in the center). |
− | #''Tone to surf volume ratio'' [percent] from 0 to 100 percent (default 70%). Adjusts the relative volume of the tones and surf sound. | + | #'''Tone to surf volume ratio:''' [percent] from 0 to 100 percent (default 70%). Adjusts the relative volume of the tones and surf sound. |
In addition to the tones, you can also generate stereo surf based on {{external|[http://en.wikipedia.org/wiki/Pink_noise pink noise]}}. This is a lower-frequency "rushing" sound compared with "hissing" white noise. | In addition to the tones, you can also generate stereo surf based on {{external|[http://en.wikipedia.org/wiki/Pink_noise pink noise]}}. This is a lower-frequency "rushing" sound compared with "hissing" white noise. | ||
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− | ///////////// | + | '''<font color="#FF6600">Buzz tone generator''' </font>(buzz.ny) {{external|[http://audacity.sourceforge.net/nyquist/buzz.ny View]}} | {{external|[http://audacity.sourceforge.net/nyquist/buzz.zip Download]}} | |
+ | {{external|[http://audacity.sourceforge.net/nyquist/buzz1.mp3 MP3 Example Clip]}} | ||
− | + | Generates a nasal-sounding tone composed of the base frequency combined with a number of harmonics. If for example you choose a frequency of 100 Hz with four harmonics, this plug-in will generate a tone comprised of 100, 200, 300 and 400 Hz, of equal amplitude. (The more harmonics, the more nasal and high-pitched the tone sounds). Although the original Nyquist code asks for a MIDI note number rather than a frequency, this plug-in allows you to choose between the frequency (e.g. 440 Hz) and the MIDI note number (57). (0=frequency, 1=MIDI note number.) You may choose the number of harmonics, tone and appended silence duration. | |
− | + | <hr> | |
− | + | <font color="#FF6600">'''Harmonic Noise''' </font>(harmonicnoise.ny) {{external|[http://audacity.sourceforge.net/nyquist/harmonicnoise.ny View]}} | {{external|[http://audacity.sourceforge.net/nyquist/harmonicnoise.zip Download]}} | |
− | + | Author: Steven Jones. Generates sounds by mixing narrow bands of noise. The center frequency of each band is harmonically related to the fundamental and the amplitude decreases inversely with the harmonic number. Depending on the width of the band, the result can sound very noisy or distinctly tonal with a heavy chorusing effect. | |
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− | + | #'''MIDI Note List:''' A list of MIDI notes to be produced. Each note must be separated by at least one space and punctuation is not allowed. Notes are specified either by an integer or by a Nyquist mnemonic. | |
+ | #'''Number of Harmonics:''' An integer between 1 and 32 which sets the number of partials for each note generated. | ||
+ | #'''Duration:''' The tone's duration in seconds. | ||
+ | #'''Band Width:''' The noise band width in Hertz. Higher values result in a more noisy tone. | ||
+ | #'''Odd Harmonics Only:''' Choose between all harmonics or odd-numbered harmonics only. | ||
<hr> | <hr> |
Revision as of 09:47, 14 January 2008
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These are optional plug-in effects for Audacity. They are written in the Nyquist programming language . For installation instructions, see the main plug-ins page .
|
Instructions: View Download Example
Audio Generators
Binaural Tones with Surf 2 (bitone2.ny) View | Download | MP3 example clip
A sinewave tone of one constant frequency is generated in the left channel of a stereo track, and a series of changing tones of slightly different frequencies are generated in the right. In addition, a stereo "surf" noise is generated. The differences between the left- and right-channel frequencies are termed "beat frequencies".
- Beta (14-21 Hz and higher)
- Alpha (7-14 Hz)
- Theta (4-7 Hz)
- Delta (0-4 Hz).
Research suggests there are many variations on these bands as well as additional brainwave frequencies, and that different states may be associated each.

To use this plug-in, first open a new stereo track in Audacity (ALT, P, S in pre-1.3 versions or ALT, T, N, S in 1.3 and later versions). Open the Generate menu and select "Binaural Tones with Surf 2" from the drop-down menu.
Variables:
- Left channel tone frequency: from 50 Hz to 1000 Hz (default 100 Hz).
- Beat frequency [Hz], duration [minutes], time to change to next beat frequency [minutes]: There are six of these edit fields in which you may enter up to three indicated values, separated by a space. The first of these edit fields has default values of 17.5 0.25 0.25. In the sixth field you may enter a final beat frequency and duration of that frequency. If you enter a only a single value into any of these fields, the duration of that beat frequency will be zero. If you leave any of these edit fields blank they will be ignored.
- Adjust total time: between 1 and 60 minutes [0=no adjustment].
- Fade-in and fade-out times: [seconds]. Sets the time for fading in and fading out the volume at the start and end of the generated audio.
- Stereo surf frequency: from 0 Hz to 2 Hz (default 0.1 Hz). If this setting is above zero, the surf sound will be panned back and forth somewhere between the left and right audio channels at the specified frequency, how far depending on the sixth variable:
- Stereo surf spread: [percent] between 0 and 100 percent (default 80%). The larger this number, the further the surf sound will move away from the center pan position (0% results in the surf sound remaining in the center).
- Tone to surf volume ratio: [percent] from 0 to 100 percent (default 70%). Adjusts the relative volume of the tones and surf sound.
In addition to the tones, you can also generate stereo surf based on pink noise . This is a lower-frequency "rushing" sound compared with "hissing" white noise.
Buzz tone generator (buzz.ny) View | Download | MP3 Example Clip
Generates a nasal-sounding tone composed of the base frequency combined with a number of harmonics. If for example you choose a frequency of 100 Hz with four harmonics, this plug-in will generate a tone comprised of 100, 200, 300 and 400 Hz, of equal amplitude. (The more harmonics, the more nasal and high-pitched the tone sounds). Although the original Nyquist code asks for a MIDI note number rather than a frequency, this plug-in allows you to choose between the frequency (e.g. 440 Hz) and the MIDI note number (57). (0=frequency, 1=MIDI note number.) You may choose the number of harmonics, tone and appended silence duration.
Harmonic Noise (harmonicnoise.ny) View | Download
Author: Steven Jones. Generates sounds by mixing narrow bands of noise. The center frequency of each band is harmonically related to the fundamental and the amplitude decreases inversely with the harmonic number. Depending on the width of the band, the result can sound very noisy or distinctly tonal with a heavy chorusing effect.
Variables:
- MIDI Note List: A list of MIDI notes to be produced. Each note must be separated by at least one space and punctuation is not allowed. Notes are specified either by an integer or by a Nyquist mnemonic.
- Number of Harmonics: An integer between 1 and 32 which sets the number of partials for each note generated.
- Duration: The tone's duration in seconds.
- Band Width: The noise band width in Hertz. Higher values result in a more noisy tone.
- Odd Harmonics Only: Choose between all harmonics or odd-numbered harmonics only.
KLSTRBAS |
View klstrbas.ny | Download klstrbas.zip
klstrbas.ny
Author Steven Jones 02 October 2004
This software released under the terms of the GNU public license
KLSTRBAS (for "cluster bass") generates several signals with a fixed frequency
ratio between them. Apparently early Roland drum machines created
cymbal sound in part by combining multiple square waves with non-integral
frequency ratios. The combined signal was then high passed filtered to
produce a very dense cluster of high frequency harmonics. The genesis of
KLSTRBAS was a failed attempt to create cymbal sounds using this technique.
Parameters:
key - MIDI key number decay - Decay time in whole seconds fdecay - Fractional decay time in 1/100 second density - The number of component waveforms is 4 times the density Higher densities produce a deeper flange effect but can also cause the sound to go out of tune. detune - Effects the relative frequencies. See below flange - Effects relative frequencies. See below tab - Wave form used as a basic component. 0=sine, 1=tri, 2=square, 3=saw The wave tables are not band limited so aliasing may result if either key or n are too high.
The frequency of each component is determined by the key number, the detune
and flange parameters. Specifically the nth component has a frequency of
p * (1 + d/100 + g)^n where
p is the fundamental frequency determined by the key number
d is the detune amount 0 <= d <= 99 and g is derived by the flange parameter.
g = 1/(10^(4-f)) for flange value f
KLSTRBAS can also produce synth kick drum sounds by setting decay time to a
fraction of a second.
Noise Band |
View nseband.ny | Download nseband.zip
nseband
Author Steven Jones 03 October 04
This software is released under the terms of the GNU public license
nseband creates narrow band noise by ring modulating a sine wave with low-pass
filtered noise. The effect is similar to band-pass filtering noise.
PWM |
View pwm.ny | Download pwm.zip
PWM.ny
Author Steven Jones 07 October 2004
This software is released under the terms if the GNU public license
Generates a modulated pulse tone with the following parameters.
key - Tones frequency as MIDI key number 0...127
cents - Detune amount in cents 0...99
Duration - Tones duration in Milli seconds, 1...30000
Mod Rate - Number of modulation cycles, 1...100
Mod Depth - Modulation depth as percent, -100...+100
Mod Wave - 0 = tri, 1 = up sawtooth, 2 = down sawtooth
Width - Fixed pulse width as percent. 0...100
Amp - Amplitude as percent 0...100
If the sum of the fixed width and the instantaneous modulation amount is
outside the interval [0,99] The output will go to full off or full on.
Random Touch Tones |
View randomtouchtones.ny | Download randomtouchtones.zip
Example audio clips: [MP3 Clip 1]
Random Touch Tones (TM) mp3 samples:
1. 20 tones generated at default values;
2. 40 tones generated at default values, silence duration 0
seconds;
3. 5 tones generated at tone and silence duration 0.2 seconds,
twist value 0.
Risset Bell |
View rbell.ny | Download rbell.zip
Adapted by Steven Jones 03 October 2004
Simulates a realistic bell tone based on the pioneering work of Jean Claude
Risset. This plug in is an adaptation of a demonstration lisp file by Pedro
Jose Morales contained in the standard Nyquist distribution. The only
parameters are MIDI key number and decay time.
Risset Drum |
View risset.ny | Download risset.zip
Risset Drum generator
The Risset-drum is based on the pioneering work of Jean Claude Risset in
the mid 1960's. It produces a realistic drum sound consisting of three
components; a sine wave ring-modulated by narrow band noise, an
enharmonic tone, and a relatively strong sine wave at the fundamental.
by Steven Jones
RNDTONE |
View rndtone.ny | Download rndtone.zip
rndtone generates random sine waves.
Author Steven Jones, 01 October 2004
This software is released under the terms of the GNU public license.
The generated tones have random frequencies, attack and decay times.
Parameters:
dur - Total duration in seconds n - Number of tones generated. floor - Minimum frequency in Hertz ceiling - Maximum frequency in Hertz
SQ1 |
View sq1.ny | Download sq1.zip
sq1 "algorithmic sequencer number 1"
Author Steven Jones, 30 September 2004
This software is released under the terms of the GNU public license.
The sq1 sequencer generates complex sequences of tones by using the sum of
three square wave LFOs to frequency modulate two oscillators. The oscillators
output one of four waves (sine, tri, square and saw) and may be detuned
relative to each other. The wave tables are not band limited so aliasing
will result for sufficiently high frequencies. There is also an overall three
stage amplitude envelope.
Parameters: center - The unmodulated carrier frequency in Hertz. detune - The frequency of oscillator 2 relative to oscillator 1 wave - The wave selector. 0=sine, 1=tri, 2=square, 3=saw Both oscillators produce the same wave-shape. attack - Attack time in seconds. sustain - Sustain time in seconds decay - Decay time in seconds f1 - Frequency of LFO 1 in hertz a1 - Amplitude of LFO 1 f2 - Frequency of LFO 2 a2 - Amplitude of LFO 2 f3 - Frequency of LFO 3 a3 - Amplitude of LFO 3
The three LFOs are interchangeable. Note that the LFO amplitudes are
calibrated in Hertz indicating the corresponding frequency shift in the audio
oscillators.
Surf |
View surf-lfo.ny | Download surf-lfo.zip
Example audio clips: [MP3 Clip 1]
surf-lfo.ny: LFO Surf generator
LFO: low frequency oscillator. A signal whose frequency is
below the human ear's ability to hear as a tone,
generally below 20 cycles per second [Hertz or hz].
Generates mono or stereo surf which sweeps between a lower and
upper filter frequency. Stereo surf also sweeps back-and-forth
somewhere between the left and right audio channels.
To generate stereo surf, first open a blank stereo track [alt+p, s
in Audacity pre-1.3, alt+t, n, s in 1.3 and later]. Open the
generate menu. Click on 'Surf [LFO]'.
Variables with instructions:
1. Mono or stereo surf [1=mono 2=stereo]
Mono surf is heard only in the center between the two speakers, or
in the middle of your head when wearing headphones. Stereo surf
also sweeps back-and-forth somewhere between the two audio
channels, depending on the next setting, Stereo Spread.
2. Stereo spread [stereo only: percent]
The larger this value, the more widely the stereo surf will move
back-and-forth between the left and right audio channels.
When this value is above zero, the deeper section of the surf sweep
will be heard more in the left channel; below zero, the deeper
section of the surf sweep will be heard mor in the right channel.
3. Fade-in and fade-out times [seconds]
To smoothly fade in and fade out the volume at the start and end of
the surf.
4. Surf duration [minutes]
Up to sixty minutes of LFO surf.
5. Surf type [0=white noise 1=pink noise]
White noise is more of a 'hissing' sound, whereas pink noise is a
lower 'rushing' sound.
Technically, white noise is 'equal energy per frequency', whereas
pink noise is 'equal energy per octave'.
6. Surf sweep frequency [hz]
Sets how slow or fast the surf sweeps between the lower and upper
filter frequencies, and the left and right channels [for stereo
surf].
The next two variables
7. Lower filter frequency [hz]
8. Upper filter frequency [hz]
determine how low and how high the lowpass filter sweeps the surf
noise.
9. Bass frequency to boost [hz]
You can boost [increase the volume of] frequencies of the surf
sound below this setting, to get a deeper-sounding surf. Somewhat
analogous to the bass knob on your stereo.
10. Bass boost [db]
Sets how much to boost the above bass frequency. 0db means no
boost, 6db means double the amplitude of the bass frequency, and so
on.
Note
If you get an error message 'Nyquist returned too many audio
channels', this means you tried to generate stereo surf without
first having opened a blank stereo track in audacity. See
instructions at the start of this help file for instructions on how
to do this.
Written by David R. Sky, June 17, 2007.
[http://www.shellworld.net/%7Edavidsky/nyquist.htm
http://www.shellworld.net/~davidsky/nyquist.htm]
Thanks to Steven Jones for pink noise generator code.
Released under terms of the GNU Public License
[http://www.opensource.org/licenses/gpl-license.php
http://www.opensource.org/licenses/gpl-license.php]
Surf |
View surf-oxy.ny | Download surf-oxy.zip
Example audio clips: [MP3 Clip 1]
surf-oxy.ny: Jean-Michel Jarre's _Oxygene_ surf
Jean-Michel Jarre put out a hauntingly beautiful electronic album
in 1976, Oxygene. One section of this album had a repeating
surf sound: a sweep from the right to the left audio channel, a
pause, and then a deep crash in the right channel. After another
pause, this cycle repeated many times. Very relaxing to listen to.
This sound generator plug-in emulates that surf cycle, in either
mono or stereo.
To generate stereo surf, first open a blank stereo track [alt+p, s
in Audacity pre-1.3, alt+t, n, s in 1.3 and later]. Open the
generate menu. Click on
'Surf [Oxygene]'.
Variables:
1. Surf output [1=mono 2=stereo]
To generate mono or stereo Oxygene surf.
2. Stereo spread [stereo only - percent]
If you've chosen to generate stereo Oxygene surf, this setting will
determine how widely the surf sweeps away from the center pan
position. From +100 percent to -100 percent. Positive values make
the sweep section go from the right to the left, with the crash in
the right. Negative values reverse this pattern.
3. Fade-in and fade-out times [seconds]
Time to fade in and fade out the volume at the start and end of the
surf, if you wish.
4. Number of Oxygene surf cycles
How many Oxygene surf cycles to generate.
5. Surf type [0=white noise 1=pink noise]
White noise is a higher-frequency 'hissing', whereas pink noise is
a lower-frequency 'rushing' sound.
The following two settings
6. Sweep starting filter frequency [hz]
7. Sweep ending filter frequency [hz]
set the starting and ending frequencies for the lowpass filter to
sweep the sweep portion of Oxygene surf. [A lowpass filter allows
frequencies below a certain value to pass, while frequencies above
that value are attenuated, or reduced in volume.]
8. Sweep duration [seconds]
This sets how slow or fast the sweep portion of Oxygene surf takes.
9. Post-sweep silence duration [seconds]
Duration of the silence after the sweep.
10. Crash filter frequency [hz]
The lowpass filter frequency of the crash.
11. Crash bass frequency boost [db]
How much to increase the volume of the above filter frequency and
below. 0db means no boost, 6db means double the amplitude of this
bass frequency, and so on.
12. Crash duration [seconds]
Duration of the crash.
12. Post-crash silence duration [seconds]
how much silence before the Oxygene surf cycle repeats.
Notes
1. If you get an error message 'Nyquist returned too many audio
channels', this means you tried to generate stereo surf without
having first opened a blank stereo track in Audacity. See
instructions at the top of this help file on how to do this.
2. In the original _Oxygene_, reverb was applied to the surf,
giving it a more expansive sound and feeling. If you want to have
reverb added to Oxygene surf, you need to apply it yourself after
the surf sound has been generated. There's 'Gverb' already in the
Audacity effects menu, and many people use Anwida's free VST reverb
plug-in.
3. Seagulls not included.
Written by David R. Sky, June 20, 2007.
[http://www.shellworld.net/%7Edavidsky/nyquist.htm
http://www.shellworld.net/~davidsky/nyquist.htm]
Thanks to Steven Jones for pink noise Nyquist code.
Released under terms of the GNU Public License
[http://www.opensource.org/licenses/gpl-license.php
http://www.opensource.org/licenses/gpl-license.php ]
Touch Tones |
View touchtone.ny | Download touchtone.zip
Touch Tones (TM) generator - produce telephone tones in Audacity.
Length of tone and appended silence are definable, plus 'twist' - the
ratio of the high to low tones (in db). Twisted telephony jargon.
Written by David R. Sky
Tuning Fork |
View tuningfork.ny | Download tuningfork.zip
Example audio clips: [MP3 Clip 1]
- tuningfork.ny: Tuning Fork
In preparing to do some vocal work with Audacity, I realized I
wanted a software tuning fork.
After you have copied tuningfork.ny into your Audacity plug-ins
folder and restarted Audacity, Tuning Fork is in the generate menu.
After you click on it, a brief table outlining C notes and their
equivalent MIDI note numbers appears near the top of the screen.
C0=0 C1=12 C2=24 and so on.
Variables:
Tone duration: up to 120 seconds.
Constant or fade out (0=constant 1=fade out, default=0 constant):
Choose whether you want the tone to remain at constant volume or
fade out during the duration.
MIDI or frequency (0=MIDI 1=frequency, default=0 MIDI):
Choose whether to generate a MIDI note number or frequency.
A440=MIDI note 69, middle C=60.
MIDI note: if you have chosen to generate a MIDI note, enter the
note number here. Note that you can use non-integers here (such as
60.75)
Frequency: If you have chosen to generate a frequency, enter the
frequency here.
Written by David R. Sky
Variable Duration Silence generator |
View varsilence.ny | Download varsilence.zip
Variable Duration Silence Generator
This plug-in will generate any length of silence from 0.000 to
60.999 seconds. I wanted it so that I could add a specific length
of time after a sound (such as Steven Jones' Risset Drum), in order
to repeat the audio for a rhythmic effect.
there are two edit fields: one for thousandths (1/1000) of a second (0.000
to 0.999 seconds), and the other for whole seconds (0 to 60), for 0.000 to
60.999 seconds.
Copy the file into your audacity plug-ins folder. Next time you
open Audacity, Variable Duration Silence Generator will be in the
generate menu.
Written by David R. Sky
Ten band E Q |
View 10bandeq.ny | Download 10bandeq.zip
Ten-band EQ. Select the band number (1 to 10) and select gain (-24 to +24 db).
Written by David R. Sky
Classic EQ |
View 15bandEQ.ny | Download 15bandEQ.zip
Example audio clips: [MP3 Clip 1] [MP3 Clip 2]
15 Band Equalizer:
This effect can equalize more than one band at the same time
You have 15, to choose and you can manipulate whole of them moving their sliders
There are also two clips:
The first one is a phrase two times, with no changes and with the five lowest bands raised 10 db
The second one is the same but here the bands wich are raised 10 db are the highest five ones
by Josu Etxeberria and David Sky
Amplify left or right channel |
View amplr.ny | Download amplr.zip
- Amplify left or right channel
If you have digitized a cassette and want to amplify or attenuate
one channel only, this plug-in will do it.
Variables
1. Channel selector: 0=left channel, 1=right channel (default
0).
2. Volume to amplify or attenuate the channel (default 0 db, no
change in volume).
Written specifically for people who use a screen reader, and people
who prefer to use the keyboard over a mouse.
Written by David R. Sky
Bouncing ball delay with panning |
View bbdelay.ny | Download bbdelay.zip
Bouncing ball delay with panning by David Sky
Bouncing ball delay with Tone Shift |
View bbdtone.ny | Download bbdtone.zip
Example audio clips: [MP3 Clip 1] [MP3 Clip 2]
- Bouncing Ball Delay with Tone Shift
Combines the bouncing Ball Delay and Delay with Tone Shift plug-
ins. A delay effect in which the echos get faster, like a bouncing
ball. And each echo is shifted in pitch by the designated amount
(semitone plus cents (hundredths of a semitone)).
The value for the decay amount (in db) for an increasing pitch can
be left at the default 0. However, with decreasing pitch, the
lengths of the delays increase over time, overlapping with each
other more and more. In this case, clipping can occur if the decay
value is left at 0.
Written by David R. Sky
Bouncing Ball Delay |
View bouncingball.ny | Download bouncingball.zip
Example audio clips: [MP3 Clip 1] Bouncing
ball delay - Just like it sounds. Like a bouncing ball, the bounces get
faster and faster. Based on a delay plug-in. You can set time that the
bounces increase in sdpeed with each delay, the number of bounces, and
how much in db the sound decreases with each bounce.
Written by David R. Sky
Stereo Butterfly |
View butterflyramp.ny | Download butterflyramp.zip
Example audio clips: [MP3 Clip 1]
Third in the series of Stereo Butterfly plug-ins. As with the
previous two, 0 setting sounds like mono, +1 is regular stereo, -1
is left and right channels flipped with each other.
Select which value to start at and which value to finish at. The
default is from 0 to 1, which creates the effect of your stereo
audio starting out sounding mono, then gradually "widening" to full
stereo as the selection progresses.
Start and finish values may lie anywhere between -1 and +1.
Written by David R. Sky
Stereo Butterfly |
View butterflystatic.ny | Download butterflystatic.zip
Stereo Butterfly (static)
The name comes from a butterfly's wings, which can be spread wide
(1, full stereo), closed (0, sounding mono), or somewhere in-
between. Stereo butterfly can even mirror the left and right
channels (-1... the butterfly's flipped!). And also anywhere
between the extremes from -1 to 1.
Written by David R. Sky
Center Pan Remover |
View centerpanremover.ny | Download centerpanremover.zip
- Center pan Remover (often called Vocal Remover)
For removing vocals which are panned to center, you can invert one
channel then pan both channels to center. Audio which is common to
both channels then disappears. this often removes other audio such
as drums, which are also often panned to center.
With centerpanremover.ny, you can select whether to invert one
channel, or to invert a range of frequencies in one channel, before
panning both channels to center.
Variables
1. Select whether to invert one channel or a band of frequencies.
2. If frequency band is selected, "remove frequencies above..." is
default 500Hz.
3. If frequency band is selected, "remove frequencies below..." is
default 2000Hz.
In other words, with the default settings, a two-octave range from
500Hz to 2000Hz (2 kilohertz) is removed.
Audio that isn't exactly panned to center is louder in one channel
than in the other. Thus, the further from the center pan position,
the louder that audio will be in the result.
Written by David R. Sky November 12, 2004
Released under terms of the GNU Public License
[http://www.opensource.org/licenses/gpl-license.php
http://www.opensource.org/licenses/gpl-license.php]
Chimes delay |
View chimesdelay.ny | Download chimesdelay.zip
Example audio clips: [MP3 Clip 1] [MP3 Clip 2] [MP3 Clip 3] [MP3 Clip 4] [MP3 Clip 5] [MP3 Clip 6]
- chimesdelay.ny: Chimes Delay
(Version 2 plug-in, works in Audacity 1.2.3 and later)
1. Adds random delay to your audio. You can specify the maximum
delay of the random delays (default 10 seconds, maximum 120
seconds) and how many random delays within that max delay time
(default 20 delays, maximum 100).
2. Using the minimum volume field, you can specify the lowest
random volume that each random delay can have (default is 50
percent, range between 0 and 100 percent).
3. This plug-in also randomly changes the pitch of each random
delay. You can specify a note list (which is where the name 'Chimes
Delay' comes from). the following is the default note list:
-24 -12 -5 0 4 7 12 14 19
Each number indicates how many semitones your audio could be pitch
shifted (along with matching tempo shift). For example, 0 indicates
no pitch shift, 12 indicates rise of 12 semitones (one octave), -5
indicates drop of 5 semitones (like going from C down to G below
that C note).
If the audio you have loaded into Audacity is C3, the above note
list would produce the following notes randomly:
C1 C2 G2 C3 E3 G3 C4 D4 G4
A distinctly major-sounding scale.
If you have a note list specified, this plug-in will randomly
choose from that list of notes for pitch and tempo shifting of each
random delay.
4. However, you can delete this note list, in which case a list of
notes will be generated between a lower and upper number. The
default values of these two numbers are -12 semitones (decrease of
1 octave) and +24 semitones (increase of 2 octaves), respectively.
5. If your audio is in stereo, each random delay with random volume
and random pitch change will also be randomly panned anywhere
between left and right. (It is best that your audio is first panned
to center before applying Chimes Delay.)
6. Additional notes:
Adding a bit of regular delay and/or other effects before applying
Chimes Delay results in a richer sound.
If you want a particular note (from the note list) to be repeated
more often, you can enter it more than once in the list.
If you simply want your audio randomly delayed with no multiple
pitch changes, either enter just one number into the note list, or
enter the same number into the min and max notes fields.
If you want no random amplitude changes, make the amplitude field
100 percent.
It is possible that total length of your resulting audio will be
max delay _plus_ duration of your original audio. This may be still
longer if the final delay(s) is/are decreased in pitch (resulting
in a reduced tempo).
If your original audio is non-musical, chimesdelay.ny will not make
it musical.
Many thanks to Steven Jones! - his cool 'Harmonic Noise' generator
plug-in is the source for Nyquist code to handle a string-input
note list. Thanks Steven for additional list help as well.
Written by David R. Sky, January 30, 2005
Released under terms of the GNU Public License
[http://www.opensource.org/licenses/gpl-license.php
http://www.opensource.org/licenses/gpl-license.php]
Comb filter |
View comb.ny | Download comb.zip
Example audio clips: [MP3 Clip 1]
The name 'comb' filter comes from how it acts on the audio spectrum
of what it's applied to: it looks like a comb with the teeth
pointing up. For example, if you set the comb frequency at 1000 Hz,
the comb filter emphasizes 1000 Hz as well as 2000, 3000, 4000 and
so on Hz. This particular plug-in produces an 'airy' effect, which
is more pronounced the higher the comb decay value is set, and
resonance is increasingly produced as well.
A comb filter can be produced using flanger-like settings on a
delay effect, but this filter does not use a delay to get the
result, so it does sound somewhat different.
Written by David R. Sky
Customizable EQ |
View customeq.ny | Download customeq.zip
Custom EQ: Select center frequency of band, width of band in octaves (0.1 to 5.0), and apply gain (-24 to +24 db).
Written by David R. Sky
Dual Tape Decks |
View dualtapedecks.ny | Download dualtapedecks.zip
Dual Tape Decks effect
This plug-in was written to duplicate an effect I heard in the late
1970s: I recorded then played identical audio on two mono tape
decks. There was an amazing "whooshing" effect as one tape deck
"caught up" with and passed what the other tape deck was playing.
This plug-in allows the "whooshing" to go back and forth. Different
effects are made using mono-sounding vs. "true" stereo audio.
You can adjust the phase difference between the two decks (180
degrees is the default), the starting phase of the effect (0
degrees is the default), LFO frequency and depth of the effect. The
larger depth is, the more pronounced the pitch and tempo shift
become until there is a noticeable warble.
Neat stereo flanger-like effects can be made by (for example)
applying dualtapedecks.ny to audio, applying Stereo Butterfly (static)
with a spread value of zero (sounds mono after applying), then
applying dualtapedecks.ny a second time with the same settings as the
first time.
This plug-in will work on mono audio as well, but the only effect
will be rising and falling changes in pitch and tempo.
Written by David R. Sky
Selection Duration |
View duration.ny | Download duration.zip
- Selection Duration (Version 2 plug-in)
Once copied into your Audacity plug-ins folder, this plug-in
utility appears in the analyze menu. It gives the duration of audio
you have selected. A lot easier to use for people using a screen
reader (rather than trying to decipher those numbers on the
screen).
If you have opened or imported more than one track and have not yet
done a Quick Mix, this plug-in sequentially gives the duration of
each track you loaded into Audacity. Simply press <enter>
after each track duration is given. The final screen gives
information from Nyquist, which you can ignore. Simply press
<enter> to get to the regular Audacity screen.
This is a Nyquist Version 2 plug-in which works in Audacity 1.2.3
and later.
Written by David R. Sky with Nyquist pointers from Steven Jones and
Dominic Mazzoni.
Regular interval labels |
View equalabl.ny | Download equalabl.zip
equalabl.ny: Regular interval labels [analyze menu plug-in]
thanks to Sami Jumppanen from the Audacity users group for
suggesting this plug-in: adding labels to the label track at
regular intervals. Thanks to leland Lucius from the Audacity
development list for code feedback which helped wake me from late
night programming! And thanks to Gale Andrews from the Audacity
development list for suggesting improvements.
Warning! Using this plug-in will delete any label track you had
in your loaded audio. If you accidentally use this plug-in and want
to restore your previous label track, simply press control+z
once.
After you've copied equalabl.ny into your Audacity plug-ins folder,
start a new session of audacity. Load audio you want to add
regularly-spaced labels to. Select audio [control+a]. Open analyze
menu [alt+a]. Click on 'Regular interval labels'. use or change the
following five default variablees:
1. Label interval [seconds]
Default sixty seconds between labels, from one second to six
hundred seconds [ten minutes].
2. Label text
The text that will appear in each label, default is "label".
3. Prepend numbers to label text [0=no 1=yes]
Default is yes, so your labels would sequentially be "0label
1label..." and so on, using the default text.
4. Final label [0=exclude 1=include]
For example, if your selection is sixty seconds long, and your
label interval is ten seconds, the final label would be at the
exact end of your selection. By default, the above variable is to
exclude [not set] the final label.
5. Final audio segment duration equal with others [0=no 1=yes]
Your label interval setting may result in the final segment of
audio being unequal with the preceding ones. By default, the above
variable is set to make the final audio segment equal in duration
with the preceding ones. This might make the label interval
slightly different from your chosen one, depending on the size of
the final audio segment.
Note
People using a screen reader can view the label track in Audacity
1.3.3 beta by opening the track menu [alt+t] and clicking on 'edit
labels'. The labels and their time positions can be read by
cursoring up and down and left-right. press alt+f4 to return to the
main Audacity screen.
Written by David R. Sky.
[http://www.shellworld.net/%7Edavidsky/nyquist.htm
http://www.shellworld.net/~davidsky/nyquist.htm]
released to the Audacity community June 25, 2007.
Thanks to Alex S. Brown for example code from his silencemarker.ny
plug-in for placing labels on the label track.
Released under terms of the GNU Public License
[http://www.opensource.org/licenses/gpl-license.php
http://www.opensource.org/licenses/gpl-license.php]
Extract Audio |
View extractp.ny | Download extractp.zip
- Extract Audio (percent)
Select all your audio, then use this plug-in to set the start and
end margins using percentages. Start and end margins are anywhere
from 0 to 100 percent.
If for example your audio is ten seconds long, you set the start at
0 percent and the end at 50 percent, the result will be the first
5 seconds of your selected audio. If you re-apply the effect with
the same settings, the result will be the first 2.5 seconds of your
original selected audio.
If you do not like the extracted audio margins, you can undo the
result and apply with slightly different settings. Or you can re-
apply for shrinking the audio further. Great for people who do not
want to fiddle around with the cursor keys for selecting audio.
Beware! Audio outside the start and end percent margins will be
discarded! A plug-in for re-inserting the (altered) extracted audio
into the discarded audio is in the works.
Written by David R. Sky
Fade In and Out |
View fade-io.ny | Download fade-io.zip
- Fade In and Out (Version 2 plug-in)
Define fade-in and fade-out times, in seconds. Saves fiddling
around with the mouse or keyboard.
Written by David R. Sky
Delay with Stereo Flip |
View flipdelay.ny | Download flipdelay.zip
- Delay with Stereo Flip
This is a stereo delay effect: with each delay, the stereo channels
are flipped left-right and vice versa. Inspired by a sound effect
heard in the opening track of Mike Oldfield's "songs From Distant
Earth."
Variables:
decay: the drop in volume (db) with each delay (default 3.0);
delay time: default 0.5 seconds;
number of delays, including the original audio: default 10.
Thanks to Steven Jones for illustrating how to check for even/odd
numbers.
Written by David R. Sky December 2, 2004
Released under terms of the GNU Public License
[http://www.opensource.org/licenses/gpl-license.php
http://www.opensource.org/licenses/gpl-license.php]
High Pass Filter with Q |
View highpass2.ny | Download highpass2.zip
High Pass Filter with Q
A high pass filter with q, or resonance. the higher q is, the more
the cutoff frequency will resonate (produce a tone). (High pass
filter allows frequencies above the cutoff frequency to pass, cuts
off the lower frequencies.) works on mono and stereo audio.
Applied to white noise, this filter and the low pass filter with Q
can be used to
produce wind-like sounds, but only at a constant frequency.
By David R. Sky
Delay with high pass filter |
View hpdelay.ny | Download hpdelay.zip
Example audio clips: [MP3 Clip 1]
High-pass Delay
You can define delay time, how much each delay
decreases in db (although this does not seem to be necessary, so
the default value is 0 db), number of delays, the starting cutoff
frequency of the high pass filter, and how much to increase the
cutoff frequency (in octaves) from there for each delay.
Applied to a voice, each delay sounds like it's increasingly coming
from a telephone.
Written by David R. Sky
Hyperexp |
View hyperexp.ny | Download hyperexp.zip
The hyperexp effect is a type of compression. Signal amplitudes of approximately
unity are relatively unchanged. Low amplitude sections are greatly
amplified. The effect is a partial nullification of the amplitude envelope.
(c) Steven Jones 27 September 2004
This software is released under the terms of the GNU public license.
LFO High Pass Filter |
View lfohp.ny | Download lfohp.zip
Example audio clips: [MP3 Clip 1] [MP3 Clip 2]
- LFO High Pass Filter
Similar to the LFO Low pass Filter, except a low frequency
oscillator moves the cutoff frequency of a highpass filter up and
down.
Variables
1. Center cutoff frequency, in Hz (default 640Hz).
2. LFO depth: how far above and below the center frequency the LFO
sweeps the filter (in octaves, default 1.0).
3. LFO frequency: speed of up and down sweeping (default
0.2Hz).
4. LFO starting phase: in degrees, from -180 to +180 (default 0).
Audio examples
lfolp1.mp3: center frequency 640Hz, 5 octave depth, LFO 1.0Hz,
applied to 110Hz square wave.
lfolp2.mp3: 640Hz center frequency, 5 octave depth, 5Hz LFO,
applied 3 times to voice.
written by David R. Sky
LFO Low Pass Filter |
View lfolp.ny | Download lfolp.zip
Example audio clips: [MP3 Clip 1] [MP3 Clip 2] [MP3 Clip 3] [MP3 Clip 4]
LFO Low Pass Filter
Like on an electronic music synthesizer, a low frequency oscillator
moves the cutoff frequency of a lowpass filter up and down.
Variables
1. Center cutoff frequency, in Hz (default 640Hz).
2. LFO depth: how far above and below the center frequency the LFO
sweeps the filter (in octaves, default 1.0).
3. LFO frequency: speed of up and down sweeping (default
0.2Hz).
4. LFO starting phase: in degrees, from -180 to +180 (default 0).
Audio examples
lfolp1.mp3, lfolp2.mp3, lfolp3.mp3: default settings, applied to
white noise 1, 2 and 3 times, respectively.
lfolp4.mp3: center frequency 640Hz, 2 octave depth, LFO 1.0Hz,
applied to 640Hz square wave.
written by David R. Sky
LFO panning |
View lfopan.ny | Download lfopan.zip
LFO panning - Panning controlled by a low frequency oscillator.
Frequency of the oscillator and width of the stereo spread are
controllable by you. You must have the audio in stereo first,
preferably in the center for best results.
Written by David R. Sky
Low Pass Filter with Q |
View lowpass2.ny | Download lowpass2.zip
Low Pass Filter with Q
A low pass filter with q, or resonance. the higher q is, the more
the cutoff frequency will resonate (produce a tone). (Low pass
filter allows frequencies below the cutoff frequency to pass, cuts
off the higher frequencies.) Works on mono and stereo audio.
By David R. Sky
Delay with low pass filter |
View lpdelay.ny | Download lpdelay.zip
Example audio clips: [MP3 Clip 1]
Low-pass delay
You can define delay time, how much each delay
decreases in db (although this does not seem to be necessary, so
the default value is 0 db), number of delays, the starting cutoff
frequency of the low pass filter, and how much to decrease the
cutoff frequency (in octaves) from there for each delay.
To me, this has the psychoacoustic effect of each delay sounding
further and further away.
Based on an effect heard in a popular Cher tune in the late 1990s
or later. Thanks for the idea Cher!
Written by David R. Sky
Multiband EQ |
View multibandeq.ny | Download multibandeq.zip
Multi-band EQ: Select total number of bands (T, from 2 to 30),
band number (1 to 30, depending on how many total bands T you chose),
and apply gain (-24 to +24 db). Determines width of band depending on
total band number T you chose.
Written by David R. Sky
Mutron |
View mutron.ny | Download mutron.zip
Loosely based on the Mutron stomp box from the late 70's. Basically it is an
envelope follower controlled filter.
Center - sets un-modulated filter frequency.
Depth - sets filter modulation depth either negative or positive.
Band Width - Resonance control, lower values are more resonant.
Mode - There are four filters:
0 - Low pass
1 - High pass
2 - Band Reject
3 - Band Pass
By Steven Jones
Notch Filter |
View notch.ny | Download notch.zip
Notch filter
Like its name suggests, a notch filter cuts out a "notch" in the
spectrum of your audio. The default frequency is 60Hz, great for
removing 60Hz electrical hum, in case your recording equipment has
picked this up.
The second control, q, determines the width of the notch cut from
your audio. Default q is 1, below 1 creates a wider notch, above 1
creates a narrower notch.
Works on mono and stereo audio.
By David R. Sky
Panning |
View pan.ny | Download pan.zip
Example audio clips: [MP3 Clip 1] Pan
- If you prefer to use the keyboard over the mouse, pan will statically
pan your audio anywhere between left and right channels. You must have
the audio in stereo first, preferably in the center for best results.
Delay with Pitch Shift |
View pitchdelay.ny | Download pitchdelay.zip
Example audio clips: [MP3 Clip 1] [MP3 Clip 2]
- Delay with Pitch Shift
A delay plug-in except each delay is pitch shifted. Standard delay
variables: decay (in db), delay time (in seconds), and number of
delays. Also includes amount of pitch shift (from 0.1 to 10.0) and
selection of whether the pitch is to be multiplied (0) or divided
(1) by the shift number.
- tonedelay.ny: Delay with Tone Shift
Similar to pitchshift.ny except you can define in semitones how
much each delay is to be pitch shifted. A shift of 1 semitone means
each delay is increased in pitch by 1 semitone, a shift of -1 means
a decrease of 1 semitone. Includes whole semitones plus semitone
cents (hundredths of a semitone).
Warning! Both plug-ins are best applied to relatively short
duration audio, or few number of delays for longer audio. Otherwise
Audacity will be working a _long_ time. Same thing seems to happen
if there is already pitch shifting within the audio. (This all may
be simply my computer, which runs at 233MHz.)
Written by David R. Sky
ramp panning |
View ramppan.ny | Download ramppan.zip
Ramp panning - evenly pan your audio from anywhere to anywhere
else in the stereo field. Left to right, center to in-between left and
center, where-ever you wish.
Written by David R. Sky
Random Amplitude Modulation |
View randomamp.ny | Download randomamp.zip
Random amplitude modulation
Similar to Random Panning, this time playing around with the volume
knob. You can set the maximum speed (maxspeed, in hz) of the random
changes. The other setting, factor, determines how deeply the
random amp effect is. Because of the way the random signal is
generated, the lower maxspeed is, the higher factor must be in
order to produce the same amp result. (Factor can be used to
increase or decrease the amp effect.)
By David R. Sky
Random low pass filter |
View randomlp.ny | Download randomlp.zip
Random Low pass Filter
Like someone is playing around with the cutoff frequency knob of
your low pass filter. As with Random Panning and Random Amp
Modulation, there are maxspeed and factor controls, plus a third
control, the maximum cutoff frequency of the low pass filter.
Once again, the lower you set maxspeed, the higher you must set
factor to get similar depth of random filtering result. You can use
factor to decrease or increase the amount of randomness modulation.
If you generate white noise then apply this effect, you can
_partially_ simulate wind sound. the rising and falling tones of
wind are for another plug-in...
By David R. Sky
Random panning |
View randompan.ny | Download randompan.zip
Random Panning
Like someone is playing around with the panning knob. You can set
the maximum speed (maxspeed, in hz) of the random changes. The
other setting, factor, determines how deeply the random panning
effect is. Because of the way the random signal is generated, the
lower maxspeed is, the higher factor must be in order to produce
the same panning result. (Factor can be used to increase or
decrease the panning effect.)
By David R. Sky
Random Pitch Modulation |
View randompitch.ny | Download randompitch.zip
Random pitch Modulation
Randomly modulates the pitch of your audio. As with previous
randomly-controlled effects, the maximum speed (maxspeed) of the
random changes is adjustable (in Hz). Again, the lower maxspeed is,
the higher you must set the factor control to get the same depth of
randomness as with higher maxspeed settings.
Increasing warping depth gives you increasing pitch changes. A
fourth control, max pitch depth, is included. If the warping depth
is high enough, max depth should be made higher otherwise there
will be momentary periods of no pitch changes. With lower warping
settings, this does not happen, and the effect can be re-applied
repeatedly for further random pitch changes.
It's difficult to explain the workings of the controls, it's easier
to experiment and find out what happens.
This effect works on mono and stereo audio. In stereo, each channel
has different random pitch modulation applied.
By David R. Sky
Reverse bouncing ball delay with Tone Shift |
View rbbdtone.ny | Download rbbdtone.zip
Example audio clips: [MP3 Clip 1] [MP3 Clip 2]
- Reverse Bouncing Ball Delay with Tone Shift
The fast bounces come first, and each bounce is tone shifted.
Written by David R. Sky
Reverse bouncing ball delay |
View reversebouncing.ny | Download reversebouncing.zip
Example audio clips: [MP3 Clip 1]
Reverse bouncing ball delay
The fast bounces come first, gradually slowing down. Reverse of the
bouncing ball delay effect.
by David R. Sky
Audio Sample Sequencer 1 |
View sequencer1b.ny | Download sequencer1b.zip
Example audio clips: [MP3 Clip 1] [MP3 Clip 2] [MP3 Clip 3]
- sequencer1b.ny: Audio Sample Sequencer 1.b
Improved over sequencer 1.a: simplified code makes this a version
1 plug-in. Sequences mono and stereo audio already loaded into
Audacity, whether a note, vocal sound, series of notes and/or vocal
sounds, etc.
Select all audio first, then click on Audio Sample Sequencer 1.b
from the effects menu.
Variables:
tempo (steady, in beats per minute);
beats per measure (one to sixteen). This plug-in "assumes" that one
sequence of notes is one measure;
sequences to generate: how many of these sequences do you want to
generate (from one to a thousand);
16 pairs of tone shift and volume level (tone shift is measured in
semitones - 0 means no tone shift, negative numbers indicate tone
drop, positive numbers indicate tone increase; volume level 1.0
means regular volume, 0.0 means you do not hear a note generated
for that beat. volume can range from 0.0 to 1.0.)
When you have a tone loaded into Audacity, you can create a major
scale of 8 notes by 1. selecting desired tempo; 2. selecting 8
beats per measure; 3. selecting how many sets
(sequences) of these 8 notes you want to generate (default is 16).
These would be the settings for the 8 tone shift edit fields:
0, 2, 4, 5, 7, 9, 11, 12.
If you start with a C note, the above settings will generate a C
major scale. If your starting note is a B flat note, the above will
create a B flat major scale.
Rhythms can also be created by playing with the volume edit fields.
For example, in a series of eighth notes, notes which fall on only
the first, fourth and seventh beats can be made by making beats per
measure 8, and making volume on #2, 3, 5, 6 and 8 to 0.0.
It is possible to generate a short sequence using this plug-in,
then apply it again to that new audio with different settings.
Copy sequencer1b.ny into your Audacity plug-ins folder. The next
time you open Audacity, you'll find audio Sample Sequencer 1.b in
the effects menu.
Written by David R. Sky December 17, 2004
Released under terms of the GNU Public License
[http://www.opensource.org/licenses/gpl-license.php
http://www.opensource.org/licenses/gpl-license.php]
Tempo Change |
View tempo.ny | Download tempo.zip
Example audio clips: [MP3 Clip 1]
- Tempo Change
I want to convert old tapes recorded at half speed to normal speed
using Audacity, and wanted a simple plug-in to change tempo.
Variables
1. Tempo change: default is 0.5.
2. Multiply or divide selector: 0=multiply, 1=divide; default is 0.
If multiply is selected, tempo of the audio is multiplied by the
tempo change number. Selecting divide divides the tempo by the
tempo change number.
Applying this effect to selected audio (using default settings)
halves the tempo (and therefore the pitch).
Multiplying by 0.5 is the same as dividing by 2.
Audio examples
tempo1.mp3: "Hello" at half, normal and twice normal speeds.
Written by David R. Sky
Time Shift tool |
View timeshift.ny | Download timeshift.zip
- Time Shift Tool
A plug-in for performing the same task as the time shift tool in
Audacity.
Choose the track you want to time shift (0=left, 1=right), and the
amount to shift (range from 0.0 to 100.0 milliseconds). If you
choose the left channel shifted by 10.0ms, it will start 10.0ms
later than previously.
Only works on stereo audio. Useful for aligning tracks due to
recording latency. Also can be used for stereo effects.
Written by David R. Sky
Delay with tone Shift |
View tonedelay.ny | Download tonedelay.zip
Turntable Warping |
View turntablewarp-ms.ny | Download turntablewarp-ms.zip
- turntablewarp-ms.ny: Turntable Warping (mono/stereo)
(version 2 plug-in)
Improved over the previous turntablewarp.ny plug-in - you can warp
both mono and stereo audio.
You can simulate unplugging your turntable while it's playing,
plugging it in, and related effects. A version 2 Nyquist plug-in
which works on Audacity 1.2.3 and later.
Variables with explanations:
"step" indicates how many semitones above or below pitch of
original audio (1 step = 1 semitone, 12 steps = 1 octave, etc.);
"amplitude" indicates volume level (in percent).
There are 5 values you can change:
1. start step (default 0): how many semitones above or below
original audio at start of selection;
2. start amplitude (default 100%): volume at start of selection;
3. change time (default 50%): point in time of original selection
at which warping values can change between first and second part of
selection. This point has an internal step value of 0 semitones
change, and internal volume of 100%;
4. end step (default -12 semitones, 1 octave drop): at end of
warped selection, how many semitones audio has been warped;
5. end volume (default 40%): volume at end of warped audio.
Applying this plug-in with the default settings results in the
audio starting to slow down halfway through the original selection,
and dropping to 40% of original volume. Like unplugging a turntable
while it's playing.
Since the "change time" is internally set at step 0 semitones and
volume 100%, different warping effects can be created, depending on
the start and end step and volume values:
slowing down then speeding up;
speeding up then slowing down;
speeding up to a particular step value then remaining at that
value;
slowly speeding up then quickly speeding up;
etc.
Thanks to Roger B. Dannenberg for the warp tutorial on which this
plug-in is based. Thanks Sebastien Chopin for the request for the
stereo version of this plug-in.
Copy turntablewarp-ms.ny into the Audacity plug-ins folder. next
time you (re)start Audacity, Turntable Warping (mono/stereo) (V2)
will appear in the effects menu.
Written by David R. Sky December 26, 2004
Released under terms of the GNU Public license
[http://www.opensource.org/licenses/gpl-license.php
http://www.opensource.org/licenses/gpl-license.php]
Turntable Warping |
View turntablewarp.ny | Download turntablewarp.zip
- turntablewarp.ny: Turntable Warping (version 2 plug-in)
You can simulate unplugging your turntable while it's playing,
plugging it in, and related effects. At this point only works on
mono audio. A version 2 Nyquist plug-in which works on Audacity
1.2.3 and later.
Variables with explanations:
"step" indicates how many semitones above or below pitch
of original audio (6 steps = half an octave, 12 steps = 1 octave,
etc.);
"amplitude" indicates volume level (in percent).
There are 5 values you can change:
1. start step (default 0): how many semitones above or below
original audio at start of selection;
2. start amplitude (default 100%): volume at start of selection;
3. change time (default 50%): point in time of original selection
at which warping values can change between first and second part of
selection. This point has an internal step value of 0 and internal
volume of 100%;
4. end step (default -12): at end of warped selection, how many
semitones audio has been warped;
5. end volume (default 40%): volume at end of warped audio.
Applying this plug-in with the default settings results in the
audio starting to slow down halfway through the original selection,
and dropping to 40% of original volume. Like unplugging a turntable
while it's playing.
Since the "change time" is internally set at step 0 and
volume 100%, different warping effects can be created, depending on
the start and end step and volume values:
slowing down then speeding up;
speeding up then slowing down;
speeding up then remaining normal;
slowly speeding up then quickly speeding up;
etc.
Thanks to Roger B. Dannenberg for the warp tutorial on which this
plug-in is based.
Written by David R. Sky December 13, 2004
Released under terms of the GNU Public License
[http://www.opensource.org/licenses/gpl-license.php
http://www.opensource.org/licenses/gpl-license.php]
Stereo Widener |
View widener.ny | Download widener.zip
- Stereo Widener
This plug-in gives the effect of widening stereo audio, using a
method David Walsh outlined on the Audacity-users list (thanks
David).
There is one variable, Inverted Signal Volume (vol, default is -20
db).
Each channel is inverted, attenuated by (vol), then added to the
opposite channel. The greater (vol) is, the wider the stereo effect.
Written by David R. Sky