Nyquist Effect Plug-ins

Amplify, Mix and Pan Effects
Amplify Left or Right Channel (amplr.ny) View | Download

Author: David R.Sky

If you have a stereo track and want to amplify or attenuate one channel only without using the mouse, this plug-in will do it.

Parameters:

Note: Audacity 2.x now has keyboard commands you could use instead: SHIFT + M to open the Track Drop-down menu on the the focused track, and then SHIFT + G to change the gain on the focused track, using up or down arrow to change focus.
 * 1) Channel 0=left channel, 1=right channel (default 0).
 * 2) Volume [dB]: amplify or attenuate the channel (default 0 dB, no change in volume).

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Bass to Center (BassToCenter.ny) [[Media:BassToCenter.ny|View]] | [[Media:BassToCenter.zip|Download]]

Author: Jvo Studer.

Frequency-selective filter to crossfeed (mix) bass frequencies to center (mono). Requires stereo tracks.

Simulates the "Elliptic EQ" filter found on vinyl mastering consoles by means of a first order highpass filter in the side-channel (L-R difference). This is useful to bring the bass drum (or beat in electronic music) to the center. If bass frequencies are partially out of phase there will be some bass loss which can be compensated for with the Bass Boost shelf filter (typically +0.5 dB to +2 dB will be sufficient).

Parameters:
 * 1) Crossover Frequency: [10 - 500 Hz, default = 150]
 * 2) Bass Feed Proportion: [20 - 100 %, default 95]
 * 3) Bass Boost: [0 - 6 dB, default 0.5]

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Center Pan Remover (centerpanremover.ny) View |  Download

Author: David R.Sky

Removes center-panned content in stereo tracks by inverting and making mono. Can be used to mitigate vocals in music tracks if the vocals are panned to centre. Optionally you can choose a band of frequencies to invert, rather than the whole channel. This may be less destructive of the content panned away from centre. The resulting audio retains two channels, but sounds mono because both channels are panned to centre.

Parameters:
 * 1) Invert band or channel: [0="band", 1="channel", default = channel]
 * 2) Remove frequencies above...: [20 - 20000 Hz, default 500]
 * 3) Remove frequencies below...: [20 - 20000 Hz, default 2000]

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Channel Mixer (channel-mixer.ny) [[Media:Channel-mixer.ny|View]] | [[Media:Channel-mixer.zip|Download]] | Type 3 plug-in (Requires Audacity 1.3.4 or later)

Author: Steve Daulton.

The Channel Mixer effect is a multi-purpose tool that can perform almost any type of channel mixing task.

Typical uses include:


 * Copying one channel of a stereo track to the other channel


 * Converting a stereo track into 2 channel mono


 * Stereo "widening" (or narrowing)


 * Swapping left and right channels of a stereo track


 * Vocal Removal.

The plug-in has 14 presets for the above and similar tasks. If the presets are not used, custom values for the mix of original left and right channel in the left and right outputs can be entered in the parameters as listed below. All values are percentages with a possible range of -100% to +100% and all default to zero value.

<-- LEFT CHANNEL OUTPUT -->  from original Left channel (%):  from original Right channel (%): <-- RIGHT CHANNEL OUTPUT --> from original Left channel (%): from original Right channel (%): 

The Zip download also includes a comprehensive 'FAQ' help file.

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Fade In and Out (fade-io.ny) View | Download

Author: David R.Sky

Define the length of fade-in and fade-out selections without using a mouse or cursor keys. Note the current Audacity 2.x has a Selection Toolbar providing a screen-reader friendly display of selection start time and duration which you could use for similar purpose.

Parameters:


 * 1) Fade in time: [seconds, maximum 30]
 * 2) Fade out time: [seconds, maximum 30]

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Panning (pan.ny) View | Download |  MP3 example clip

Authors: David R.Sky, Dominic Mazzoni

Audacity 2.x now lets you pan with the keyboard instead of the mouse, but if you are still using 1.2.x, or prefer the pan to modify the waveform immediately, this plug-in will statically pan your stereo audio anywhere between left and right channels. There is only one parameter:


 * 1) Pan position: [0=left, 1=right, default is 0.5 (center-panned)]

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Panning (LFO) (panlfo2.ny) Type 1 plug-in for Audacity 1.2.x and pre-1.3.4: View | Download

(panlfo2a.ny) Type 3 plug-in for Audacity 1.3.4 or later: (improved version) View | Download

Author: David R.Sky

Panning is controlled by a low frequency oscillator. Only works on unsplit stereo tracks. Pan the audio to center before use for best results.

Parameters:


 * 1) LFO frequency: [Hz, 0.02 - 20, default 0.1]
 * 2) LFO waveform: [sine, triangle, saw, inverted saw, pulse]
 * 3) Pulse waveform duty cycle: [percent, default 50]
 * 4) LFO starting phase: [degrees, -180 - +180, default 0]
 * 5) Leftmost pan position: [percent, default 5] - 0%=left channel, 50%=center, 100%=right channel
 * 6) Rightmost pan position: [percent, default 95] - 0%=left channel, 50%=center, 100%=right channel

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Panning (random) (panrand.ny) [[Media:panrand.ny|View]] |  [[Media:panrand.zip|Download]]

Author: David R.Sky

Randomly pans stereo audio from one side of the stereo field to the other - just like someone is playing around with the panning knob. Requires an unsplit stereo track.

Parameters:


 * 1) Maximum random panning speed: [Hz, 0.01 - 10.00, default 0.2] - how fast the random panning changes occur
 * 2) Maximum stereo width: [percent, 0 - 100, default 100] - how far away from center the signal is panned. 0% gives no panning, 100% results in the signal being randomly panned between hard left and hard right pan positions.

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Pseudo-Stereo (Pseudostereo.ny) [[Media:Pseudostereo.ny|View]] |  [[Media:Pseudostereo.zip|Download]] | Type 3 plug-in (Requires Audacity 1.3.4 or later)

Author: Steve Daulton

A stereo spatializer effect. Creates an artificial stereo effect that can be useful for giving some "depth" to mono recordings. Mono tracks must be converted to a 2 channel track before using the effect. To do so, click above the Mute and Solo buttons in the Track Control Panel, choose then click on the name of the upper track and select "Make Stereo Track" from the drop-down menu.

Parameters:


 * 1) Select source channel: [Left (upper) or Right (lower)]
 * 2) Delay factor (%): [0 to 100, default 30] Higher values will produce a wider stereo effect but may sound echoey.
 * 3) Effect mix (%): [0 to 100, default 80] 0% = Dry (original signal with no effect), 100% = Wet (effect and no original signal). Higher settings will produce a more pronounced stereo effect but may leave a "hole" in the center of the stereo field. Lower values produce a more subtle effect with the original signal centered mid-stage.

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Ramp Panning (panramp.ny) View | Download

Author: David R.Sky

Evenly pan your stereo audio, starting at one point in the stereo field and ending at another. -10 corresponds to 100% left, 0 to center and +10 to 100% right.

Parameters:


 * 1) Start position: from [where -10 - +10, default -10]
 * 2) End position: from [where -10 - +10, default +10]

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Repair Channel (repair_channel.ny) [[Media:Repair channel.ny|View]] | [[Media:Repair channel.zip|Download]] | Type 3 plug-in (Requires Audacity 1.3.4 or later)

Author: Steve Daulton

For repairing damage to one channel of a stereo track by overwriting the damaged region with audio from the other channel. Select the damaged audio and allow additional space for cross-fading, then apply the effect.

The plug-in includes an option for Stereo Simulation which will often make repairs to stereo recordings less noticeable. For tracks that have little or no stereo channel separation and for synthesized tones, best results will probably be achieved with Stereo Simulation disabled.

Parameters:


 * 1) Which Channel to Repair: [Left Channel or Right Channel] The upper channel of a stereo pair is the Left channel.
 * 2) Stereo Simulation: [Enabled or Disabled]
 * 3) Cross-fade Time: [0% to 50%] 0% will cut directly from one channel to the other without fading. At 50% the fades will occupy the entire selection. For the default 20% fade, the selection should be at least 40% longer than the actual damage to be repaired.

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Stereo Butterfly (static) (buttrfly.ny) View | Download

Author: David R.Sky

The original Stereo Butterfly plug-in, the name coming 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.

Parameters:


 * 1) Stereo width [width, between -1.0 and +1.0]

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Stereo Butterfly (LFO) (buttrlfo.ny) View | Download

Author: David R.Sky

Second in the Stereo Butterfly series. It takes stereo audio and makes it sound like the left and right channels are switching back and forth with each other. You can define the LFO (low frequency oscillator) rate. As in Stereo Butterfly (static), -1 is stereo channels fully flipped with each other, 0 sounds like mono, and 1 is full regular stereo. The difference here is that you can define two widths, so defining how you want the stereo to be manipulated over time. For instance, from -1 to +1 means Stereo Butterfly flips the left and right channels with each other at the frequency you set. If you set the two numbers at 0 and +1, the stereo audio will change between mono-sounding and regular stereo. Set at -1 and 0, the effect will be of fully flipped stereo changing to mono-sounding. Any other numbers you choose between -1 and +1 will give intermediate effects. Parameters:


 * 1) LFO frequency: [between 0.01 and 20 Hz]
 * 2) Width1: Stereo width from [range from -1.00 to +1.00]
 * 3) Width2: to Stereo width [range from -1.00 to +1.00]

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Stereo Butterfly (ramp) (buttrramp.ny) View | Download |  MP3 example clip

Author: David R.Sky

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.

Parameters:


 * 1) Spread stereo from... [range from -1.00 to +1.00]
 * 2) to: [range from -1.00 to +1.00]

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Stereo Widener (widener.ny) View | Download

Author: David R.Sky

Gives the illusion of widening stereo audio. The effect produces different results depending on whether you are listening to the audio through speakers or headphones, and the distance stereo speakers are apart. The widener works by inverting both left and right channels of stereo audio, then panning those inverted signals somewhere between the center pan position and the opposite channel.

Parameters:


 * 1) Inverted signal volume: [-48 dB - -6 dB, default -18 dB]
 * 2) Pan position: [0 (center) to -100 (opposite channel), default 0]
 * 3) Time offset: [0 - 20 ms, default 0] - applying an offset can enhance the illusion.

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Delay and Reverb
Bouncing Ball Delay (bouncingball.ny) View | Download |  MP3 example clip

Author: David R.Sky

Just like it sounds. A delay plug-in where, like a bouncing ball, the bounces get faster and faster. You can set time that the bounces increase in speed with each delay, the number of bounces, and how much in dB the sound decreases with each bounce.

Parameters:


 * 1) Decay amount [dB]: - how much quieter each bounce is.
 * 2) Delay time [seconds]:
 * 3) Number of bounces [times]:
 * 4) Tone shift (whole) [semitones]:
 * 5) Tone shift (cents) [cents]:

Note: The latest available delay plug-in 3 in Audacity 2.x includes bouncing ball delay and pitch shifting. but not panning.

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Bouncing Ball Delay with Panning (bbdelay.ny) [[Media:bbdelay.ny|View]] | [[Media:Bbdelay.zip|Download]]

Author: David R.Sky

Combines the Bouncing Ball Delay with a panning effect. A delay effect in which the echo get faster, like a bouncing ball. Each echo is panned further from center by the designated amount.

Parameters:


 * 1) Decay amount [dB]:
 * 2) Delay time [seconds]:
 * 3) Number of bounces [times]:
 * 4) Pan spread movement [move]: - defines the extent to which each bounce will be increasingly far from center

Note: The latest available delay plug-in 3 in Audacity 2.x includes bouncing ball delay and pitch shifting, but not panning.

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Bouncing Ball Delay with Tone Shift (bbdtone.ny) View | Download |  MP3 example clip 1 |  clip 2

Author: David R.Sky Combines the Bouncing Ball Delay and Delay with Tone Shift plug-ins. A delay effect in which the echoes get faster, like a bouncing ball. Each echo is shifted in pitch by the designated amount in semitones plus cents (hundredths of a semitone).

Parameters:


 * 1) Decay amount [dB]:
 * 2) Delay time [seconds]:
 * 3) Number of bounces [times]:
 * 4) Tone shift (whole) [semitones]:
 * 5) Tone shift (cents) [cents]:

Notes:
 * 1) 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. In this case, clipping can occur if the decay value is left at 0.
 * 2) Note: The latest available delay plug-in 3 in Audacity 2.x includes bouncing ball delay and pitch shifting.

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Reverse Bouncing Ball Delay (delayreb.ny) View | Download |  MP3 example clip

Author: David R.Sky

The fastest bounces come first, gradually slowing down - reverse of the bouncing ball delay effect. Includes normalisation. Note: The latest available delay plug-in 3 in Audacity 2.x includes reverse bouncing ball delay without normalisation, and also includes pitch shifting.

Parameters:


 * 1) Decay amount: [0.00 - 5.00 dB, default 0.50]
 * 2) Delay time: [0.01 - 1.00 seconds, default 0.05]
 * 3) Number of bounces: [1 - 100, default 15]

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Reverse Bouncing Ball Delay with Tone Shift (rbbdtone.ny) View | Download |  MP3 example clip 1 |  clip 2

Author: David R.Sky

The fastest bounces come first - reverse of the bouncing ball delay, and each bounce is tone shifted. Note: The latest available delay plug-in 3 in Audacity 2.x includes reverse bouncing ball delay and pitch shifting.

Parameters:


 * 1) Decay amount: [0.00 - 5.00 dB, default 0.05]
 * 2) Delay time: [0.01 - 1.00 seconds, default 0.02]
 * 3) Number of bounces: [1 - 100, default 15]
 * 4) Tone shift (whole): [-24 - +24 semitones, default -1]
 * 5) Tone shift (cents): [-100 - +100, default 0]

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Chimes delay (chimesdelay.ny) View | Download |  MP3 example clip 1 |  clip 2 |  clip 3 |  clip 4 |  clip 5 |  clip 6

Author: David R.Sky

Adds random delay to your audio, and randomly changes the pitch of each delay if you specify a note list (which is where the name 'Chimes Delay' comes from).

Each number in the note list indicates how many semitones your audio should be pitch shifted (along with matching tempo shift). For example, 0 indicates no pitch shift, 12 indicates a rise of 12 semitones (one octave), -5 indicates drop of 5 semitones (like going from C down to G below that C note).

Example: If the audio you have loaded into Audacity is C3, the above note list would randomly produce the following major-sounding notes:

C1 C2 G2 C3 E3 G3 C4 D4 G4

If you delete this note list, 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.

If your audio is stereo, each random delay with random volume and 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.)

Tips:
 * 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 minimum and maximum notes fields.
 * It is possible that total length of your resulting audio will be maximum delay time *plus* the 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).
 * Warning: If your original audio is non-musical, Chimes Delay will not make it musical!

Parameters:


 * 1) Chimes note list: [default list: -24 -12 -5 0 4 7 12 14 19]
 * 2) Minimum note:   [semitones from -12 to +48]
 * 3) Maximum note:  [semitones from -12 to +48]
 * 4) Maximum delay time: [seconds from 0.5 to 120] - the maximum delay of the random delays
 * 5) Minimum volume:  [percentage] -  the minimum random volume that each random delay can have. If you want no random amplitude changes, set this field to 100 percent.
 * 6) Number of chime delays: [from 1 to 100] - how many delays within the maximum delay time.

Acknowledgement due to Steven Jones whose "Harmonic Noise" generator plug-in is the source for Nyquist code to handle a string-input note list.

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Delay with High Pass Filter (hpdelay.ny) View | Download |  MP3 example clip

Author: David R.Sky

Applies a high pass filter to a delay so that with each subsequent delay, the filter's cut-off frequency is increased. A high pass filter attenuates sound below a given cut off frequency, therefore when this plug-in is applied, each delay sounds increasingly thin and lacking bass. Applied to a voice, it makes each delay sounds like it's increasingly coming from a telephone.

Parameters:


 * 1) Decay amount: [0 - 24 dB, default zero] - how much quieter each subsequent delay is
 * 2) Delay time: [0 - 5 seconds, default 0.5]
 * 3) Number of echoes: [1 - 30, default 10]
 * 4) Start cutoff frequency: [100 - 5000, default 1000] - the high pass cut-off frequency at the start of the delay.
 * 5) Cutoff increase: [octaves, 0.1 - 5.0, default 0.5] - how much to increase the filter cut-off point with each delay.

back to table of contents Delay with Low Pass filter (lpdelay.ny) View | Download |  MP3 example clip

Author: David R.Sky

Applies a low pass filter to a delay so that with each subsequent delay, the filter's cut-off frequency is reduced. A low pass filter attenuates sound above a given cut off frequency, therefore when this plug-in is applied, each delay sounds to have increasing bass. To the author, 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.

Parameters:


 * 1) Decay amount: [0 - 24 dB, default zero] - how much quieter each subsequent delay is
 * 2) Delay time: [0 - 5 seconds, default 0.5]
 * 3) Number of echoes: [1 - 30, default 10]
 * 4) Start cutoff frequency: [100 - 20000, default 1000] - the low pass cut-off frequency at the start of the delay.
 * 5) Cutoff reduction: [octaves, 0.1 - 5.0, default 0.5] - how much to reduce the filter cut-off point with each delay.
 * 6) Normalization level: [0 - 1.0, default 0.95]

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Delay with Pitch Change (delaypit.ny) View | Download |  MP3 example clip 1 |  clip 2

Author: David R.Sky

A delay plug-in except, each delay is pitch shifted. Note that pitch changes are accompanied by a corresponding change in duration of each delay.

Parameters:


 * 1) Decay amount: [0 - 24 dB, default 0]
 * 2) Delay time: [0 - 5 seconds, default 0.5]
 * 3) Number of echoes: [1 - 30, default 10]
 * 4) Pitch change factor: [1.001 - 3.0, default 1.1]
 * 5) Pitch: increase or decrease: [default = increase] - whether each delay is increased or decreased in pitch
 * 6) Normalization level: [0.0 - 1.0, default 0.95]

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Delay with Stereo Flip (delayfli.ny) View | Download

Author: David R.Sky

This is a stereo delay effect: with each delay, the stereo channels are flipped from left to right and vice versa. Inspired by a sound effect heard in the opening track of Mike Oldfield's "Songs From Distant Earth."

Parameters:


 * 1) Decay amount: [0 - 24 dB]
 * 2) Delay time: [0 - 5 seconds]
 * 3) number of delays: [1 - 100]
 * 4) Normalization level: [0 - 1, default 0.95]

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Delay with Tone Shift (tonedelay.ny) View | Download

Author: David R.Sky

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.)

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Distortion Effects
Harmonic Enhancer (enhancer.ny) [[media: Enhancer.ny | View]] | [[media: Enhancer.zip | Download]] | Type 3 plug-in

Author: Jvo Studer

Adds high frequency harmonics to brighten very dull recordings that don't respond to Equalization. On already good recordings, you can add a little extra "sparkle" or "air". The harmonics are generated by soft-clipping the high frequency band as in a diode limiter, then recombining this signal with the original.

Parameters:


 * 1) Enhancer Crossover Frequency: [2000 to 4500 Hz - default=3200] - lower this for very dull sources, or increase it to add only very slight or subtle high frequency harmonics.
 * 2) Enhancer Drive: [-10 to 10 dB - default = 0] - increase this to generate more harmonics and vice-versa.
 * 3) Harmonic Generator Mode: [Even order,Odd order - default = Even] - generates even harmonics or odd harmonics. Even harmonics tend to be less harsh.
 * 4) Enhancer Noise Gate Threshold: [-40 -16 dB - default = -28] - increase this to prevent adding un-necessary noise in quieter recordings.
 * 5) Enhancer Mix Level: [-26 to +6 dB - default = -10] - how much of the generated harmonics are mixed into the original audio.
 * 6) Output: [Mix (Normal),Effect Only,Effect Level - default = Mix] - the two "effect" modes let you see and hear the generated harmonics on their own. Use "Effect Level" specifically to test if the Enhancer Drive level is set correctly. and run Harmonic Enhancer in Mix (Normal) mode to apply the effect.

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Dynamics Processing
Broadcast Limiter II (RFT-Limiter-II.ny) View | Download

Author: Edgar-rft

Gives you the possibility to overdrive an Audacity track without introducing ugly digital distortion noise. The Limiter cuts all peaks above the given threshold, rounds the edges to avoid ugly distortion, while simultaneously amplifying the whole track to the maximum limit. This is a "soft clipping" effect.

Parameters:


 * 1) Threshold: sets the 'cutting edge' in a linear volume number from 0.0 to 1.0

The minimum threshold is -90dB, so you can set the threshold slider to 0.0 and listen to 1-bit of a 16-bit recording if you want. The plug-in has no memory limits, it can process audio tracks of several hours in length without problems.

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Broadcast Limiter III (RFT-Limiter-III.ny) View | Download

Author: Edgar-rft

Is in principle a similar "soft clipping" effect as 'Broadcast Limiter II', but adds an Exciter to control or intensify the high-range distortion. This function is often desired by musicians to make e.g. electric guitars or drum sets sound more aggressive.

Parameters:


 * 1) Exciter: controls the high-range distortion in linear numbers from 1 to 10
 * 2) Threshold: sets the 'cutting edge' in a linear volume number from 0.0 to 1.0

The plug-in has no memory limits, it can process audio tracks of several hours in length without problems.

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Hyperexp (hyperexp.ny) View | Download

Author: Steven Jones.

The Hyperexp effect is a type of compression. High amplitude sections of approximately unity are relatively unchanged. Low amplitude sections are greatly amplified. The effect is a partial nullification of the amplitude envelope. There is one parameter, which is to choose to normalize or not, the default choice being "yes".

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Limiter (Limiter.ny) [[media: Limiter.ny | View]] | [[media: Limiter.zip | Download]] | Type 1 plug-in for any version of Audacity

Author: Steve Daulton.

A "lookahead" dynamic range limiter to compress peaks that extend beyond the set threshold value. This is not a "wave shaper", it is a very fast compressor and is able to limit the maximum peaks with minimal harmonic distortion.

This limiter is an ideal choice for peak limiting live music recordings due to the exceptionally low harmonic distortion. For best results the audio should be normalized to 0 dB before applying this effect.

Parameters:
 * 1) Limit to (dB): (-10 dB to 0 dB) Sets the maximum peak level. As peaks in the original audio approach this level the gain is reduced so as to prevent the peaks exceeding the set level.
 * 2) Hold (ms): (1 to 50 ms Default = 10 ms) Holds the gain at the reduced level after a peak is detected so as to prevent the gain from "riding the waveform" which would cause harmonic distortion.
 * 3) Make-up Gain (0=No, 1=Yes): (Default = 1) When enabled (set to 1) the output is amplified by an amount equal to the "Limit" level. If the input audio has a peak level of 0 dB, the peak output level will also be 0 dB. When disabled the peaks are limited only.

Shorter Hold times allow the peaks to be tracked more accurately and the limiter will respond faster to the dynamics. If there are high levels of very low bass it will be necessary to increase the Hold time to avoid distortion. The default 10 ms hold time is sufficient for frequencies down to 100 Hz without distortion. To cleanly limit high amplitude, very low frequency bass (down to 50 Hz) the Hold should be increased to 20 ms. Setting the hold to 50 ms is sufficient right down to 20 Hz but the delay before the gain level "recovers" is likely to be too slow for most material.

Additional notes:


 * Threshold level: The Limit to (dB) control has a range of -10 dB to 0 dB. The effect is not designed to work beyond this range. When set to 0 dB there will be no change to the audio (though any over 0 dB audio will be clipped). If set below -10 dB the knee will be so soft that all of the audio will be compressed, not just the peaks.


 * Stereo Tracks: As is normal for this type of effect, the left/right channels of a stereo track are processed independently.


 * Lookahead: The limiter looks ahead for peaks and will begin to change the gain just before the peak occurs. This ensures that all peaks, no matter how fast they occur, will be caught. The lookahead time is roughly a quarter of the hold time.


 * Knee: The "hardness" of the knee depends on the threshold (Limit To) level. When the threshold is close to zero a hard knee is used but as the threshold is lowered the knee becomes softer so as to provide a smooth transition in gain level even with a very fast attack time. A typical threshold level of around -3 dB will have a relatively "soft knee" so as to avoid unnecessary distortion. That is, the amount of compression (compression ratio) progressively increases as the input gets louder. At the "Limit to" level the compression ratio is infinite (brick wall) which ensures that peaks will not exceed the limit.


 * Creative use: The limiter can be used on its own, or can be used to limit peaks after running a compressor that does not use lookahead (such as the SC4 LADSPA compressor). This can produce "crisper" compression than using a lookahead compressor such as the standard Audacity compressor or Chris's dynamics compressor.


 * Over 0 dB input: The input waveform should not exceed 0 dB. Over 0 dB input signals are treated as illegal and will be hard clipped to 0 dB before processing with the limiter. If necessary, the Amplify or Normalize effects should be run before applying this limiter to ensure that the input does not exceed 0 dB.

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Limiter (2) (limiter2.ny) [[media: limiter2.ny | View]] | [[media: limiter2.zip | Download]] | Type 3 plug-in for Audacity 1.3.4 or later.

Author: Steve Daulton.

The same as Limiter.ny except that the make up gain control is a multi-choice selection rather than a slider.

Parameters:
 * 1) Limit to (dB): (-10 dB to 0 dB)
 * 2) Hold (ms): (1 ms to 50 ms Default = 10 ms)
 * 3) Apply Make-up Gain: [No, Yes (default)]

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Noise Gate (noisegate.ny) [[media: noisegate.ny | View]] | [[media: NoiseGate.zip | Download]] | Type 3 plug-in

Author: Steve Daulton.

Noise Gates may be used to cut the level of noise between sections of a recording. While this is essentially a very simple effect, this Noise Gate has a number of features and settings that allow it to be both effective and unobtrusive and well suited to most types of audio.

Parameters:

For more detailed information and usage tips, read the help file included in the ZIP package, or the help screens included in the plug-in.
 * 1) Select Function: [Apply the Noise Gate effect | Test the noise level | View one of the Help screens].
 * 2) Stereo Linking: [Link Stereo Tracks (gate audio when both channels fall below the gate threshold)| Don't Link Stereo (gate channels independently)]
 * 3) Apply Low-Cut filter: [No (Do not apply filter) | 10Hz 6dB/octave | 20Hz 6dB/octave] Removes sub-sonic frequencies including DC offset.
 * 4) Gate frequencies above: [0 kHz to 10 kHz] Applies the gate only to frequencies above the set level which may be useful for reducing tape hiss, but will also introduce some 'phase shift'. Setting this below 0.1 kHz will switch this feature off.
 * 5) Level reduction: [-100 dB to 0 dB] How much the gated sections are reduced in volume. Values below -96 dB 'shut' the gate to produce absolute silence.
 * 6) Gate threshold: [-96 dB to -6 dB] When the audio level drops below this threshold the gate will 'close' and the output level will be reduced. When the audio level rises above this threshold the gate will 'open' and the output will return to the same level as the input.
 * 7) Attack/Decay: [10 to 1000 milliseconds] How quickly the gate opens and closes. At the minimum (10 ms) the gate will fully open and close almost instantly as the audio level crosses the threshold. At the maximum (1000 ms), the gate will begin to slowly open (fade-in) 1 second before the sound level exceeds the Threshold, and will gradually close (fade-out) after the sound level drops below the Threshold over a period of 1 second. Longer gate times (up to 10 seconds) may be achieved using text input rather than the slider.

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Pop Mute (popmute.ny) [[media: popmute.ny | View]] | [[media: PopMute.zip | Download]] | Type 3 plug-in

Author: Steve Daulton.

The effect is like an "upside-down" Noise Gate. Whereas a Noise Gate attenuates sounds that are below a specified threshold level, Pop Mute attenuates sounds that are above a specified threshold level. The effect can be used to heavily attenuate loud sounds. It may be useful for rescuing recordings that suffer from loud clicks or pops.

Sounds (such as 'pops') that have a peak level above the 'Threshold' level will be lowered to a 'residual' level set by the 'Mute Level'. Be aware that ALL sounds above the threshold will be affected. Take care to avoid selecting loud sounds that should not be muted.

The effect 'looks ahead' for peaks so that it can begin to lower the level of the sound smoothly a short time before the peak occurs. This is set by the 'Look ahead' time value. After the peak has passed, the level will smoothly return to normal over a period set by the 'Release time' setting.

To attenuate brief clicks, time values of around 5 ms are likely to work well. For larger pops, values of 10 ms or more may sound better. For reverberant sounds such as hand claps, the 'Release time' may be increased so as to catch some of the reverberation.

Parameters:


 * 1) View Help: [No | Yes] (default "No") View the built-in help screen.
 * 2) Threshold: [-24 dB to 0 dB] (default -6 dB) This is the level above which sounds are acted on (reduced in level)
 * 3) Mute Level: [-100 dB to 0 dB] (default -24 dB) How much to reduce the peak level by.
 * 4) Look ahead: [1 to 100 milliseconds] (default 10 millisecond) How far to look ahead for the next "pop" or "crackle".
 * 5) Release time: [1 to 1000 milliseconds] (default 10 millisecond) How rapidly to "release" the effect and return to normal volume after the pop has passed.

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Text Envelope (TextEnvelope.ny) [[media: TextEnvelope.ny | View]] | [[media: TextEnvelope.zip | Download]] | Type 3 plug-in (Requires Audacity 1.3.13 or later).

Author: Steve Daulton.

Provides an alternative to the "Envelope Tool" that is accessible for visually impaired and other users that do not use pointing devices. This effect provides a means to shape the volume level of a track or selection by fading from one control point level to the next. Control points are defined by a pair of numbers, the first of which sets the time position of the control point and the second defines the amplification level. Initial and final amplification settings may also be defined. Help screens are available in the 'Select Function' control of this effect.

Parameters:

Note: Decimal values must use a dot as the decimal separator.
 * 1) Select function: [choices: Apply Effect, View Quick Help, View Examples, View Tips. Default = "Apply Effect"]
 * 2) Time Units: [choices: milliseconds, seconds, minutes, percent. Default = seconds]
 * 3) Amplification Units: [choices: dB or Percent. Default = dB]
 * 4) Initial Amplification [Numeric input. Default = none]
 * 5) Final Amplification [Numeric input. Default = none]
 * 6) Intermediate Control Points as pairs of time and amplification [Pairs of numbers. Default = none]

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Filters
Band Stop Filter (bandstop.ny) [[Media:Bandstop.ny|View]] | [[Media:Bandstop.zip|Download]] Type 3 plug-in (Requires Audacity 1.3.4 or later).

Author: Steve Daulton.

A band-rejection filter that passes most frequencies unaltered, but stops those in a specific range.

Set the 'Centre Frequency' slider, or type in a value for the centre of the frequency band to block.

Set the 'Stop-Band Width' to determine how wide the cut frequency band will be. Smaller numbers will produce a narrower 'notch' and larger numbers will cut a broader band of frequencies.

This filter uses steep high pass and low pass filters to achieve the band stop effect. The filters iterate to improve the band stop efficiency for narrow band width and can thereby perform close to total blocking down to almost 1/4 octave.

For even narrower notches a notch filter should be used.

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Chebyshev Type I Filter (chebyI.ny) [[Media:ChebyI.ny|View]] | [[Media:ChebyI.zip|Download]]

Author: Kai Fisher

A Chebyshev filter with options for high-pass or low-pass operation.

Type I Chebyshev filters can provide a steeper roll-off than Butterworth filters but at the expense of more ripple in the passband. The plug-in provides unity gain (except for ripple) in the passband. This plug-in is capable of providing an exceptionally steep cut-off transition by selecting a "high order".

Parameters:


 * 1) Filter Type: [choice: Lowpass / Highpass] (default Lowpass)
 * 2) "Order: [choice 2 to 30 in steps of 2] (default 6) The higher the "order" number, the steeper the cut-off transition from the passband to stopband.
 * 3) Cut-off Frequency: [1 to 48000 Hz] (default 1000 Hz). The actual filter frequency is limited to half of the track sample rate (the Nyquist frequency). For example, if the track sample rate is 44100 Hz, then setting the Cut-off frequency to any value greater than 22050 will produce the same result as setting the frequency to 22050 Hz.
 * 4) Ripple: [0.0 to 3.0 dB] (default 0.05) Lower values will produce less ripple in the passband at the expense of a less steep cut-off. Higher values will produce a steeper cut-off but with more ripple in the passband. The difference in ripple and cut-off slope is likely to be most noticeable with low order filters and may be noticed as a slight boost or ringing in the passband just before the cut-off frequency.

When Ripple is set to zero, the passband response is essentially flat and the filter has the characteristics of a Butterworth filter.

The high-pass and low-pass filters may be used one after the other to produce a "flat topped" band-pass effect, in which the lower cut-off is provided by the high-pass filter and the upper cut-off provided by the low pass filter. The passband is the frequency band that passes between these two cut-off frequencies.

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Classic EQ (15bandEQ.ny) View | Download | MP3 example clip 1 | clip 2

Authors: Josu Etxeberria and David R.Sky.

An Equaliser (EQ) that can modify more than one band at a time. You have 15 bands to choose from and can manipulate all of of them independently by moving their sliders.

Example clips: clip 1 is a phrase spoken twice, first with no equalisation and then with the five lowest frequency bands raised 10 dB; in clip 2, the five highest frequency bands are raised 10 dB.

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Comb Filter (comb.ny) View | Download |  MP3 example clip

Author: David R.Sky

The name 'comb' filter comes from how it acts on the audio spectrum 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 Hz and succeeding frequencies. 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.

Parameters:


 * 1) Comb frequency: [Hz, 20 - 5000, default 440]
 * 2) Comb decay: [0 - 0.1, default 0.025]
 * 3) Normalization level: [0.0 - 1.0, default 0.95]

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Customizable EQ (eqcustom.ny) View | Download

Author: David R.Sky

Parameters:


 * 1) Center frequency: [Hz, 20 - 20000, default 440]
 * 2) Band width in octaves [octaves, 0.1 - 5.0, default 1.0]
 * 3) Gain: [dB, -48.0 - +48.0, default 0.0]
 * 4) Apply normalization? [Default = "no"]
 * 5) Normalization level: [0.0 - 1.0, default 0.95]

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Desk EQ (desk-eq.ny) [[Media:Desk-eq.ny|View]] | [[Media:Desk-eq.zip|Download]]

Author: Steve Daulton

This EQ is modelled on the EQ section of the Allen & Heath(TM) GL series mixing desk.

It is a 4-band EQ (equaliser) with two semi-parametric mids and provides independent control of four frequency bands plus a low frequency roll-off switch (HPF). Allen & Heath (along with Soundcraft and Neve) are well known for their distinctive "British EQ". The two "mid" filters are bell shaped peak/dip filters which affect frequencies around a centre point which can be swept from 500 Hz to 15 kHz, and 35 Hz to 1 kHz respectively. The width of the band is selected to provide effective control for both creative and corrective equalisation.

Parameters:


 * 1) 100 Hz HPF: (+/- 15 dB) attenuates frequencies below 100 Hz by 12 dB per octave. It may be used to reduce low frequency noise such as microphone popping, stage noise and tape transport rumble.
 * 2) HF Gain: sets the gain of the high frequency shelf filter which boosts or cuts high frequencies. Positive values will tend to make the sound "brighter". Negative values will tend to make the sound less bright.
 * 3) High-Mid Frequency: (500 Hz to 15 kHz) sets the centre frequency of the high-mid band filter.
 * 4) High-Mid Gain: (+/- 15 dB) sets the gain of the high-mid band filter.
 * 5) Low-Mid Frequency: (35 Hz to 1 kHz) sets the centre frequency of the low-mid band filter.
 * 6) Low-Mid Gain: (+/- 15 dB) sets the gain of the low-mid band filter.
 * 7) LF Gain: (+/- 15 dB) sets the gain of the low frequency shelf filter. Positive values will tend to give the sound more bass and negative values will reduce the bass.

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High Pass Filter with q (highpass2.ny) View | Download

Author: David R.Sky

A high pass filter with q, or resonance. A high pass filter attenuates frequencies below a given cut-off point. The higher q is, the more the cut-off frequency will resonate (produce a tone). Applied to white noise, both this filter and the low pass filter with Q can be used to produce wind-like sounds at a constant frequency. See the high pass filter (LFO) and low pass filter (LFO) for ability to modulate a fixed resonance cut-off frequency.

Parameters:


 * 1) Cutoff frequency: [20 - 10000 Hz, default 1000]
 * 2) Filter q (resonance): [0 - 5, default 1]

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High Pass Filter (LFO) (hplfo.ny) View | Download | MP3 example clip 1 | clip 2

Author: David R.Sky

A high pass filter with a low frequency oscillator (LFO). A high pass filter attenuates frequencies below a given cut-off point. The LFO in this plug-in modulates the cut-off frequency up and down, like on an electronic synthesizer.

Parameters:


 * 1) LFO frequency: [0 - 20 Hz, default 0.2] - defines the speed of the oscillation, higher is faster
 * 2) Lower cutoff frequency: [20 - 20000 Hz, default 160]
 * 3) Upper cutoff frequency: [20 - 20000 Hz, default 2560]
 * 4) LFO starting phase: [-180 to + 180 degrees, default 0]

Example clip 1: LFO frequency of 1.0 Hz, lower frequency 113 Hz, upper frequency 3620 Hz, applied to 110Hz square wave. Example clip 2: LFO frequency of 5.0 Hz, lower frequency 113 Hz, upper frequency 3620 Hz, applied three times to a voice.

Alternative version

(lfohp.ny) View | Download

Author: David R.Sky

Parameters:


 * 1) Center cutoff frequency: [20 to 20000 Hz, default 640]
 * 2) LFO depth (radius): [0.0 to 10.0, default 1] - how far (in octaves) from center f the filter sweeps.
 * 3) LFO frequency: [0.0 to 20.0, default 0.2]
 * 4) LFO starting phase: [-180 to + 180 degrees, default 0]

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Low Pass Filter (LFO) (lplfo.ny) View | Download |  MP3 example clip 1 |  clip 2 |  clip 3 |  clip 4

Author: David R.Sky

A low pass filter with a low frequency oscillator (LFO). A low pass filter attenuates frequencies above a given cut-off point. The LFO in this plug-in modulates the cut-off frequency up and down, like on an electronic synthesizer.

Parameters:


 * 1) LFO frequency: [0 - 20 Hz, default 0.2] - defines the speed of the oscillation, higher is faster
 * 2) Lower cutoff frequency: [20 - 20000 Hz, default 160]
 * 3) Upper cutoff frequency: [20 - 20000 Hz, default 2560]
 * 4) LFO starting phase: [-180 to + 180 degrees, default 0]

Example clips 1 - 3: LFO frequency of 0.2 Hz, lower frequency 320 Hz, upper frequency 1280 Hz, applied to white noise once, twice and three times respectively. Example clip 4: LFO frequency of 1.0 Hz, lower frequency 320 Hz, upper frequency 1280 Hz, applied to 640 Hz square wave.

Alternative version

(lfolp.ny) View | Download

Author: David R.Sky

Parameters:


 * 1) Center cutoff frequency: [20 20000 Hz, default 640]
 * 2) LFO depth (radius): [0.0 to 10.0, default 1] - how far (in octaves) from center f the filter sweeps.
 * 3) LFO frequency: [0.0 to 20.0, default 0.2]
 * 4) LFO starting phase: [-180 to + 180 degrees, default 0]

back to table of contents Low Pass Filter with Q (lowpass2.ny) View | Download

Author: David R.Sky

A low pass filter with q, or resonance. A low pass filter attenuates frequencies above a given cut-off point. The higher q is, the more the cut-off frequency will resonate (produce a tone). Applied to white noise, both this filter and the high pass filter with Q can be used to produce wind-like sounds at a constant frequency. See the low pass filter (LFO) and high pass filter (LFO) for ability to modulate a fixed resonance cut-off frequency.


 * 1) Cutoff frequency: [20 - 10000 Hz, default 1000]
 * 2) Filter q (resonance): [0 - 5, default 1]

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Multiband EQ (multibandeq.ny) View | Download

Author: David R.Sky

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.

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Mutron (mutron.ny) View | Download

Author: Steven Jones.

Loosely based on the Mutron stomp box from the late 70's. Basically it is a filter controlled by an envelope follower.

Parameters:


 * 1) Center/Cutoff: [0 - 10000 Hz, default 100] - sets the static filter frequency
 * 2) Depth: [-10000 - +10000 Hz, default 5000] - sets the negative or positive filter modulation depth
 * 3) Band Width: [50 - 400 Hz, default 100] - controls the resonance, lower values being more resonant
 * 4) Mode: [0="Low" 1="High" 2="Notch" 3="Band" (default)] - sets the filter mode: 0 = "Low pass", 1 = High pass, 2 = Band Reject (cut a notch at the filter frequency), 3 = Band Pass

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Notch Filter (notch.ny) View | Download

Authors: Steve Daulton and Bill Wharrie.

Like its name suggests, a notch filter cuts out a "notch" in the spectrum of your audio. The default frequency (60 Hz) can remove much of the hum that recordings can acquire from 60 Hz mains supply (as used in North and Central America and much of South America). You can set Frequency to 50 Hz to counteract mains hum in other countries. See chart of mains frequencies by country.

Filter frequencies above 10000 Hz may be entered by typing the value but are only valid up to half of the sample rate of the audio being processed. Q values outside of the slider range can be entered by typing the values but must be greater than 0.01.

Parameters:


 * 1) Frequency: [0 - 10000 Hz, default 60 Hz]
 * 2) Q: [0.1 - 20.00, default 1.00] -  determines the width of the notch. Below 1 creates a wider notch, above 1 creates a narrower notch.

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Parametric EQ (Parametric.ny) [[Media:Parametric.ny|View]] | [[Media:Parametric.zip|Download]]

Author: Steve Daulton and Bill Wharrie

A parametric equalizer is a variable equalizer effect which provides control of three parameters: amplitude, center frequency and bandwidth. This plug-in provides control of one frequency band that can be "tuned" to a user defined center frequency. The width of the affected frequency band may be adjusted with the "Width" control and the defined frequency band may be boosted or attenuated according to the "Gain" control.

Parameters:


 * 1) Frequency (Hz): [10 to 10000 Hz, default 1000 Hz] - sets the center frequency of the filter
 * 2) Width: [0 to 10, default 5] - determines the width of the affected frequency band. Greater width settings affect a broader range of frequencies. Smaller width affects a narrower band of frequencies. Numerically the width setting is approximately the half gain width in half octaves, thus the default setting of 5 has a half gain width of approximately 2.5 octaves.
 * 3) Gain (dB): [-15 to +15 dB, default 0 dB (no effect)] - how much the filter center frequency is boosted or attenuated.

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Random Low Pass Filter (randomlp.ny) View | Download

Author: David R.Sky

Like someone is playing around with the cut-off frequency knob of your low pass filter. Because of the way the random signal is generated, the lower the maximum speed is, the higher the depth factor must be to produce a similar depth of filtering changes. If you generate white noise then apply this effect, you can to some extent simulate constant pitch wind noise.

Parameters:


 * 1) Max filter sweep speed: [0.01 - 10.0 Hz, default 0.2] - maximum speed of the random filter cut-off changes
 * 2) Filter depth factor: [1 - 300, default 20] - how extreme the random filter cut-off changes are
 * 3) Maximum cutoff frequency: [20 - 5000 H, default 2000] - the filter's maximum cut-off frequency

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Resonant Filter (resonant.ny) [[media:Resonant.ny|View]] | [[media:Resonant.zip|Download]]

Author: Steve Daulton

A filter with low pass, high pass and band pass options with a "resonance" control.

Audio filters are commonly designed to have a smooth frequency response that is essentially flat in the pass band then rolls off to a lower level in the stop band, but in some cases it is desirable to use a filter that has a peak and accentuates frequencies close to the defined filter frequency. Such filters are commonly used in sound synthesis to cause "ringing" at specified frequencies. This tends to be most effective with sounds that have complex frequency content, such as noise.

Parameters:


 * 1) Filter frequency: [1 to 20000 Hz] (default: 1000 Hz) - The corner frequency of the filter. The frequency must be below the Nyquist Frequency (half the sample rate) or an error message will be displayed.
 * 2) Resonance (Q): [0.1 to 100] (default: 10) - The amount of resonance. Higher values will produce a more pronounced and narrower peak at the corner frequency. Lower values will produce a less prominant peak with values below 0.7 showing no peak at all.
 * 3) Filter type: [choice: Low Pass, High Pass, Band Pass] (default: Low Pass) - Low pass allows frequencies below the corner frequency to pass through the filter and reduces frequencies above the corner. High Pass allows frequencies above the corner to pass and reduces frequencies below the corner. Band Pass reduces frequencies that are below the corner and reduces frequencies that are above the corner, allowing only a band of frequencies around the corner frequency to pass.
 * 4) Output Gain: [-60 to 0 dB] (default -12 dB) - Because the resonance accentuates frequencies around the corner frequency it is often necessary to reduce the output level of this effect. Lower (more negative) values reduce the level more.

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Shelf Filter (shelf.ny) [[media:shelf.ny|View]] | [[media:shelf.zip|Download]]

Author: Steve Daulton

A shelf filter with options for high shelf, low shelf or mid-band.

Low-shelf filter passes all frequencies, but increases or reduces frequencies below the shelf frequency by specified amount. High-shelf filter passes all frequencies, but increases or reduces frequencies above the shelf frequency by specified amount. Mid-band shelf filter passes all frequencies, but increases or reduces frequencies between the low and high cut-off frequencies by specified amount.

Parameters:


 * 1) Filter type: [low-shelf / high-shelf / mid-band] - specifies which type of filter
 * 2) Low frequency cut-off: [1 to 10000 Hz] - The corner frequency for the low shelf filter, or the lower corner frequency for the mid-band filter.
 * 3) High frequency cut-off: [0.1 to 20 kHz] - The corner frequency for the high shelf filter, or the upper corner frequency for the mid-band filter. The high frequency cut-off must be less than half the track sample rate.
 * 4) Filter gain: [+/- 30 dB] - how much to boost or cut the filtered audio. Positive values boot and negative values reduce the level.

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Ten Band EQ (10bandeq.ny) View | Download

Author: David R.Sky

An Equaliser (EQ) that can modify one band at a time. Select the band number (1 to 10) and gain (-24 to +24 dB).

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Modulation Effects
Dual Tape Decks (dualtapedecks.ny) View | Download

Author: David R.Sky

Simulates two tape decks playing identical tapes, but out of synchronisation with each other. Written to produce 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.

The plug-in can produce some interesting stereo effects, though note that due to the greater "cross-talk" of speakers, listening in speakers and headphones will sound different. 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.

Parameters:


 * 1) LFO frequency: [Hz, 0.001 to 25.000]
 * 2) Starting phase: [degrees, -180 to +180, default 0]
 * 3) Phase difference: [degrees, 0 to 360, default 180]
 * 4) Depth: [0.001 to 2.000] - The larger depth is, the more pronounced the pitch and tempo shift become until there is a noticeable warble.

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Flanger (linear) (flangerl.ny) [[Media:flangerl.ny|View]] | [[Media:flangerl.zip|Download]]

Author: David R.Sky

Unlike a regular flanger (which cycles up and down repeatedly), this plug-in creates a single high-frequency flange, and you can set the position of that high-frequency point, anywhere between the start and end of the selection and beyond. The plug-in works by mixing the original selected audio with a slightly shrunk (shorter) version of itself.

Works on mono and stereo audio.

Parameters:


 * 1) High frequency flange position: [percent, - 100 to + 200, default 0] - the position in the signal where the high frequency portion of the flange is heard. If set to 0%, the high frequency will be at the start of the selection; at 50% in the exact middle; at 100%, at the end. If you set this value below 0% or above 100%, you won't hear the highest flange frequency peak but will hear a falling or rising flange effect, as if the peak lay outside the start or end of the selection.
 * 2) Time decrease: [milliseconds, - 0.1 to 200, default 5.0] - how much the length of the original selection is decreased.
 * 3) Wet level: [percent, 1 to 99, default 50] - the 'wet' signal is the shortened signal, and the wet level is how much (in percent) to mix with the dry (unaltered) audio.
 * 4) Wet: inverted or positive: [0=inverted, 1=positive, default positive]

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Isochronic modulator (isomod.ny) [[Media:isomod.ny|View]] | [[Media:isomod.zip|Download]]

Author: Steve Daulton

A variable tremolo plug-in customised with controls for length and fade in/out speed of each pulse. The modulation frequency (speed) and depth transform gradually from the initial settings to the final settings. A modified square wave is used as the modulation waveform.

The effect is typically applied to a single tone to create isochronic tones. The author supplies the plug-in only as a demonstration of audio processing without endorsing or claiming any relevance to the theory or practice of brainwave entrainment.

Parameters:


 * 1) Pulse Width: [percent, 0 - 100, default 40] - How long each pulse will be "on". Higher values will make the sound "on" for longer. 50% gives a "square pulse" where the sound will be on for half the time.
 * 2) Fade Time: [percent, 0 - 100, default 15] - adjusts the fade in and fade out speed of the pulse. Higher values produce a more gradual fade in and out of the pulses. At 100% the fade in/out times will be half of the pulse width. At 0% there will be no fade.
 * 3) Initial Modulation Frequency: [Hz, 1 - 20, default 7.0]
 * 4) Final Modulation Frequency: [Hz, 1 - 20, default 2.0]
 * 5) Initial Modulation Depth: [percent, 0 - 100, default 100]
 * 6) Final Modulation Depth: [percent, 0 - 100, default 100]

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Random Amplitude Modulation (randomamp.ny) View | Download

Author: David R.Sky

Similar to Random Panning, this time playing around with the volume knob. Because of the way the random signal is generated, the lower the maximum speed is, the higher the depth factor must be to produce a similar depth of amplitude changes.

Parameters:


 * 1) Max amp sweep speed: [0.01 - 20.0 Hz, default 0.5] - maximum speed of the random amplitude changes
 * 2) Amp sweep depth factor: [1 - 300, default 80] - how extreme the random amplitude changes are

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Random Pitch Modulation (randompitch.ny) View | Download

Author: David R.Sky

Randomly modulates the pitch of your audio. As with other randomly-controlled effects, the the lower the maximum speed is, the higher the depth factor must be to produce a similar depth of random changes. This effect works on mono and stereo audio. In stereo, each channel has different random pitch modulation applied. "Max pitch mod depth" can be explained thus: at higher warping depth settings, pitch mod depth should be made higher, otherwise there will be momentary periods without pitch changes. With lower warping depth settings this does not happen, and the effect can be re-applied repeatedly to give further random pitch changes.

Parameters:


 * 1) Warping depth: [0.001 - 2.000, default 0.100] - controls the number of pitch changes that occur
 * 2) Max sweep speed: [0.01 - 20.0 Hz, default 0.50] - maximum speed of the random pitch changes, higher values increase "warbling" effect
 * 3) Sweep depth factor: [1 - 300, default 80] - how far
 * 4) Max pitch mod depth: [0.01 - 3.00, default 0.50]

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Ring modulator (ringmod.ny) [[Media:ringmod.ny|View]] | [[Media:ringmod.zip|Download]]

Author: David R.Sky

A ring modulator is a tremolo effect, but instead of using an LFO to amplitude modulate audio, an audio signal is used. The result is a combination of the sum of and the difference between the two input signal frequencies e.g., two sine waves of 440 Hz and 660 Hz produce a result of 220 Hz (difference) and 1100 Hz (sum).

This plug-in also allows use of triangle, sawtooth and pulse waveforms, so the results are the sums and differences between the harmonics of the modulating signal and harmonics of the signal being modulated.

Parameters:


 * 1) Modulation frequency: [Hz, 20 to 5000, default 500]
 * 2) Amount: [percent, 0 to 100, default 100]
 * 3) Waveform: [0=sine, 1=triangle, 2=sawtooth, 3=pulse, default sine]
 * 4) Pulse bias: [percent, -100 to +100, default 0] - if the pulse waveform is selected, bias is the pulse width. The default 0 gives a "square" wave. Lower values give a narrower positive signal, higher values a wider positive signal.

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Variable Tremolo (vari-tremolo.ny) [[Media:Vari-tremolo.ny|View]] | [[Media:Vari-tremolo.zip|Download]]

Author: Steve Daulton

This plug-in produces a "Tremolo" effect in which the frequency and depth of the tremolo varies from an initial setting to a final setting. The effect makes the loudness "wobble" and you can set the speed of the wobble and the amount that it wobbles for both the start of the selection and the end of the selection.

Parameters:


 * 1) Tremolo Shape: [sine,triangle,sawtooth,inverse sawtooth,square. default = "sine"]
 * 2) Sine: The volume rises and falls smoothly up and down.
 * 3) Triangle: The volume alternates between rising at a constant rate and falling at a constant rate.
 * 4) Sawtooth: The volume rises abruptly but falls gradually.
 * 5) Inverse Sawtooth: The volume rises gradually but falls abruptly.
 * 6) Square: The volume jumps abruptly between higher and lower levels.
 * 7) Starting Phase: [0 to 360 degrees, default = 90] - This sets the starting point for the tremolo cycle. At 90 degrees the tremolo starts at the higher level. At 0 degrees the tremolo starts at the lower level.
 * 8) Initial Tremolo Frequency: [1 to 20 Hz - default = 4] - The speed of the tremolo effect at the start of the selection.
 * 9) Final Tremolo Frequency: [1 to 20 Hz - default = 12] - The speed of the tremolo effect at the end of the selection.
 * 10) Initial Tremolo Amount: [0 to 100 % - default = 40] - How much the initial volume varies as a percentage of the original level.
 * 11) Final Tremolo Amount: [0 to 100 % - default = 40] - How much the final volume varies as a percentage of the original level.

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Sequencer Effects
Audio Selection Sequencer 2 (seq2.ny) View | Download | MP3 example clip 1 | clip 2 |  clip 3

Author: David R.Sky

Developed from the previous sequencers 1a and 1b (these can't be recommended due to the interface being too tall, and distortion problems). You can turn any short piece of imported, recorded or generated audio into a (repeated) sequence of notes based on chosen tempo, beats per sequence and semitone values. Any sound can be used (a guitar pluck or bell sound, cat meow, someone saying "hello", in fact - any sound). Comes with an 8-note default sequence already programmed in. Can also pan stereo audio, and transpose successive sequences. To generate a rest, use r, n,, or a blank svp input line.

Parameters:
 * 1) Tempo: [beats per minute (default 210), beats per sequence (default 8), starting offset (beats) (default 0)]
 * 2) Pan stereo selection: [0 = no 1=yes (default)]
 * 3) Timing randomization: [0 - 100 plus or minus percent, default 0]
 * 4) Number of repeated randomized sequences: [0 - 8, default 0]
 * 5) Overall transpose value: [(default 0), then for successive measures (default 0 0 5 5 0 0 -5 -5)]
 * 6) Sequences to generate: [1 - 96, default 4]
 * 7) 1st Semitone, volume, pan value[s]: [default (0 1 0) (4 .5 .2) (7 .5 .8) (2 1 .5)]
 * 8) 2nd SVP value[s]: [default (12 .5 .8) (7 .5 .2) (4 1 1) r]
 * 9) 3rd, 4th. 5th and 6th SVP value[s]: [filled in by user]

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Time, Pitch and Tempo
Extract Audio (extract.ny) View | Download

Author: David R.Sky

Extracts audio from a selected area without using a mouse or cursor keys. The current Audacity 2.x has a Selection Toolbar providing a screen-reader friendly display of selection start time and duration which you could use for similar purpose, but if you're using 1.2.x with a screen reader, this plug-in provides a solution. It also has an easy option to extract a percentage of the selected audio. For example, selecting 50% "start percent" and 100% "end percent" will leave you with only the last half of your selection.

Parameters:


 * 1) Time or percent?
 * 2) Start time: [seconds, maximum 600]
 * 3) End time: [seconds, maximum 600]
 * 4) Start percent:
 * 5) End percent:

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Regular interval audio splitter (equasplit.ny) [[Media:equasplit.ny| View]] | [[Media:Equasplit.zip| Download]]

Author: David R.Sky

Previously called "Audio chunker". Splits audio into desired number of segments by inserting silences of specified duration (up to 10.0 seconds). You can also specify fade-in and fade-out lengths for each segment. By setting number of segments to ' 1 ' you can use audio splitter to apply a specified fade-in and fade-out to a single length of audio in one action.

Parameters:


 * 1) Number of audio segments: [1 to 120, default 10]
 * 2) Silence duration between segments [seconds]: [0.01 to 10, default 1]
 * 3) Fade-in/fade-out length [milliseconds; 0 = no fade]: [0 to 500, default 20]

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Tempo Change (tempo.ny) View | Download |  MP3 example clip

Author: David R.Sky

For those who are confused by Audacity's "Change Speed" effect where to make the tempo twice as slow you apply a -50% change and to make it twice as fast apply a 100% change, try this plug-in. Its default settings multiply the tempo by 0.5, making the tempo twice as slow (dividing by 2.0 has the same effect). To make the tempo twice as fast, simply multiply by 2.0 (or divide by 0.5). The default setting (the opposite of Audacity's "Change Speed" default) might be handy for example to return tapes dubbed at 2x speed to normal speed.

Parameters:


 * 1) Tempo change factor: [0.1 to 8.0, default 0.5]
 * 2) Multiply or divide: [0=multiply (default), 1=divide] - multiplies or divides by the tempo change number.

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Time Shifter (timeshif.ny) View | Download

Author: David R.Sky

A plug-in for performing the same task as the Time Shift Tool in Audacity, without using a mouse. The effect works thus: if the shift value is positive, silence is inserted before the selection. If the shift value is negative, audio is removed from the start of the selection. If your selected track is mono, set the value in the left/mono edit field, if your track is stereo, set the values in both the left/mono and right channel shift value fields. Setting slightly different values in both fields when you have a stereo track can be used for special effect.

Note that positive shifts can lead to the audio being truncated at the right edge when shifting a stereo track. To avoid this, split the stereo track using the Track Drop-down Menu, select each track in turn and apply the shift you want using the Mono/left channel shift field. If the audio starts after time zero and is preceded by empty space, convert the empty space to silence with Project > Quick Mix in legacy Audacity up to and including version 1.3.13. In current Audacity, select the track, press, then press , then. This prevents truncation of the audio providing there is sufficient silence at the start of the track.

Parameters:


 * 1) Mono/left channel shift: [-1000 - +1000, default 0]
 * 2) Right channel shift: [-1000 - +1000, default 0]
 * 3) [0=milliseconds 1=seconds] (default is milliseconds)

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Turntable Warping MS (turntablewarp-ms.ny) View | Download

Authors: David R.Sky, Roger B. Dannenberg

Simulates unplugging and plugging in a turntable while it's powered, and related effects such as speeding it up before unplugging it. Improved over the previous turntablewarp.ny - this version lets you warp both mono and stereo audio. The default settings simulate unplugging a turntable - the audio starts to slow down halfway through the selection area, and drops to 40% of original volume by the end.

Parameters:


 * 1) Initial step: [-36 - +36 semitones, default 0] - the resulting pitch change at the start of the selection, defined as number of semitones above or below the original pitch (1 step = 1 semitone, so 12 steps = 1 octave.)
 * 2) Initial amplitude: [0 - 100 percent, default 100] - the resulting volume at the start of the selection, related to the original amplitude (so 100% = no change).
 * 3) Change time: [0 - 100 percent, default 50] - the point in time in the selection (set internally to the original pitch and volume) at which the warping values change to reach and move away from this point. What this means is that depending on your start and end step and amplitude values, you can design warps that slow down then speed up; speed up then slow down; speed up to a particular pitch then remain there; slowly speed up then quickly speed up.
 * 4) End step: [-36 - +36 semitones, default -12] - the resulting pitch change at the end of the audio. compared to the original pitch.
 * 5) End amplitude: [0 - 100 percent, default 40] - the resulting volume at the end of the selection, compared to the original volume.

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