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Audacity Plugins

This site features download links to various plugins for Audacity, which can be used to extend Audacity's functionality to better match your workflow.

On this site you can find download links for equalizer plugins, filter plugins, delay and reverb plugins, and more. If you'd like to download many plugins at once, you also can try entire plugin suites.

All plugins in the realtime effects section can be used instantly and non-destructively from the effects panel in Audacity.

This means that you can change the effect settings while playing, and come back to your settings after doing many other things - without the need to undo anything, or to wait for the effect to be applied to the track.

Further information about using realtime effects can be found here:

Installing plugins

When you install plugins, they're automatically enabled the next time you restart Audacity. Alternatively, you can also locate and enable them via Effect -> Plugin Manager -> Rescan Plugins. Further instructions can be found here:

Popular plugins

, a plugin to check if an audio book complies with the ACX guidelines
, an effects library from Audacity's sister project.
, a library to import and export various media formats such as M4A, WMA and MP4.

Contributing to this site

If you want to add your own plugin, or one that you found on the web, read the .

Plugin formats supported by Audacity

Nyquist plugins

Audacity has built-in support for Nyquist effects on all operating systems. You can download additional Nyquist plugins, or create your own using the . Nyquist code can be conveniently tested using "" under the Effect menu.

LV2 plugins

Audacity has built-in support for plugins, which are an extensible successor of LADSPA effects. LV2 plugins are mostly built for Linux, but Audacity supports LV2 on all operating systems. To install LV2 plugins, place them in the then use the Plugin Manager to enable the new plugins as in the plugin installation instructions.

VST plugins

Audacity can load VST effects (but not VST instruments) on all operating systems.

VST3 plugins

Starting with Audacity 3.2, VST3 effects are also supported on all operating systems.

Audio Unit plugins

On Mac OS X only, you can add plugins to the .

LADSPA plugins

LADSPA is superseded by LV2. These plugins are mostly built for Linux.

Other sources

This site is only meant as a starting point for effects. There are many more plugins, particularly VST3 ones, available on various vendor sites, directories and online stores. Some of them are listed below:

realtime effects

Plugin Suites

Families of plugins by the same authors

Muse FX

A collection of essential effects. Contains a Chorus, Compressor, De-esser, Delay, Master, Noise Gate, Pitch Fix, Reverb, Rotary, and two EQ effects.

MuseFX is a free and expanding collection of high quality plugins for Audacity, MuseScore 4 and any VST3 compatible host. MuseFX brings the tools you need to elevate your mix to a higher level.

Available for all platforms

GVST

A set of simple VST effects, most known for the GSnap (auto-tune) effect.

Available for Windows and macOS

MFreeFXBundle

A collection of 37 effects in categories like Analysis, Saturation & distortion, Dynamics, Equalization, Mastering, Mixing, Pitch and others.

ToneBoosters

Powerful plugins for any type of audio mixing: eq, dynamics, reverb, and more.

Details

"Demo" mode is fully functional except for preset saving. You can use Audacity's preset system instead.

What is Included:

Barricade 4 : Limiter
BitJuggler : Sampler / Processor

Kilohearts Essentials

Kilohearts Essentials is free collection of 30+ plugins.

Available for Windows only

Reaplugs

A selection of some effects found in Reaper.

Intel OpenVINO

Music separation, remix and generation, noise suppression, and transscription

Available for macOS

MacOS already ships with a range of Audio Unit effects which you can try out

Available for Linux

swh Plugins

A set of LADSPA effects. They can be found in most Linux repositories.

Calf Studio Gear

An LV2 plugin suite featuring a wide array of plugins for audio mixing and mastering. As of version 0.90.0 there are 47 plugins in the suite.

Note: The virtual instruments included are not supported in Audacity.

The Calf plugins can be found in the repositories of most Linux distributions, additionally, compilation instructions can be found here:

Pitch and Tempo

MuseFX PitchFix

An easy to use pitch correction/snapping/autotune effect. Part of the pack.

Details

See the pack for installation instructions.

Dynamics Processing

Muse FX Compress

An easy to use compressor effect. Part of the pack.

Details

See the pack for installation instructions.

AI plugins

Intel OpenVINO

A set of AI-enabled effects, generators, and analyzers. These AI features run 100% locally on your PC 💻 -- no internet connection necessary! OpenVINO™ is used to run AI models on supported accelerators found on the user's system such as CPU, GPU, and NPU.

Music Separation🎵 -- Separate a mono or stereo track into individual stems -- Drums, Bass, Vocals, & Other Instruments.
🧹 -- Removes background noise from an audio sample.
🎶 -- Generates music from a text prompt or continues an existing song
🎤 -- Uses to generate a label track containing the transcription or translation for a given selection of spoken audio or vocals.

NVIDIA Broadcast

A virtual device that sits between your microphone and Audacity and other programs which allows you to use an AI denoiser.

Caution: NVIDIA Broadcast only works on Windows machines with a NVIDIA RTX GPU.

Further, it only works on spoken word content; musical content is treated as noise and filtered out.

Details

Requires NVIDIA GeForce RTX 2060, Quadro RTX 3000, TITAN RTX or higher

Full setup guide:

Technically, NVIDIA Broadcast isn't a plugin but a virtual device. You will find it in Audacity's audio settings as an input. It does not show up in the Plugin Manager.

Nyquist Plugins

Generator Plugins

Various generator-type Nyquist plugins. Licensed under GPL v2 unless otherwise noted.

Noise Generators

Harmonic Noise

Generates approximately harmonic tones by mixing narrow bands of noise

Details

Author: Steven Jones.

Generates approximately harmonic tones by mixing narrow bands of noise. For each note in the MIDI note list an n-partial tone is produced. Each partial of each tone is actually a narrow band of noise centered at the ideal harmonic frequency. The center frequency of each band is harmonically related to the fundamental and the amplitude decreases inversely with the harmonic number.

Parameters:

MIDI Note List: [c2 c3 ef4 g4 bf4 c5] - MIDI notes may be specified either as integers or using the Nyquist constants 'g3' for third octave g, 'bf4' for fourth octave b flat, 'gs2' for second octave g sharp and so on.
Number of Harmonics: [1 - 32, default 8] - that is, the number of partials for each note generated
Duration: [1 - 30 seconds, default 10]

Narrowband Noise

Narrowband Noise generates noise within a specified frequency range by ring-modulating a sine wave with low-pass filtered noise. The effect is like band-pass filtered noise.

Details

Author: Steve Daulton (after "Noise Band" by Steven Jones).

Narrowband Noise generates noise within a specified frequency range by ring-modulating a sine wave with low-pass filtered noise. The effect is like band-pass filtered noise.

Parameters:

Center Frequency (Hz): [10 to 10000 Hz, default 440]

Sequence Generators

Rndtone

Generates random sine waves.

Details

Author: Steven Jones.

Generator Utilities

Nyquist Generate Prompt

Nyquist Generate Prompt makes it easy for Nyquist developers to test their code for "Generate" plugins in one step.

Details

Authors: Steve Daulton, Edgar Franke, Steven Jones and

Nyquist Generate Prompt makes it easy for Nyquist developers to test their code for "Generate" plugins in one step, unlike the Nyquist Prompt built into Audacity under the Effect menu, which requires several steps including pre-loading audio to test. Simply type your Nyquist generate code into the edit fields, then Left Click, TAB or SHIFT + TAB from one edit line to the next or previous line.

For example:

The first line generates 10 seconds of white noise and assigns it to the variable "mysound". The second line applies a low-pass filter sweeping down from 5 kHz to 100 Hz over a period of 10 seconds to "mysound".

Tuning fork

A software tuning fork, with a table giving C notes and their equivalent MIDI note numbers near the top of the screen.

Details

Author:

A software tuning fork, with a table giving C notes and their equivalent MIDI note numbers near the top of the screen.

Parameters:

Tone duration: [0 - 300 seconds, default = 120]

Effect Plugins

Distortion Effects

Harmonic Enhancer

Adds high frequency harmonics to brighten very dull recordings that don't respond to .

Details

Author: Jvo Studer

Sequencer Effects

Audio Selection Sequencer 2

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.

Details

Author:

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:

Tempo: [beats per minute (default 210), beats per sequence (default 8), starting offset (beats) (default 0)]
Pan stereo selection: [0 = no 1=yes (default)]
Timing randomization: [0 - 100 plus or minus percent, default 0]

Analyzers

Loudness compliance checks

Mastering loudness for various streaming services

Various streaming platforms and other institutions expect content to have a certain range for peaks, RMS, LUFS, noise floor or a combination of the aforementioned. These plugins provide the relevant checks for the platforms in question.

Service

ACX-Check.ny

ACX, Audible

amazon.ny

Amazon Music, Alexa

apple.ny

Apple Music

Details

Authors: Will McCown, Steve Daulton, Philip Collier

Original ACX plugin by Will McCown. Modifications for services other than ACX by Philip Collier.

Displayed Results

Peak level Maximum peak level in the selection

Contributing

Adding plugins to this site

If you'd like to contribute first make a gitbook account and then go to to get access to the Plugin space. From there, editing .

Alternatively, you can also make a pull request to the on Github.

Requirements for Plugins

Developing your own plugins and scripts

Plugins

Audacity supports the following formats:

Nyquist (Lisp or SAL) Audacity's built-in scripting format. Allows for easy generation of UI elements and already is tuned to Audacity's needs. Note that this plugin format is not supported as realtime effects. Documentation:

Creating your own Nyquist Plugins

Basics

Volume Basics

This page explains how to use Nyquist to change the volume of Audacity tracks in different ways.

Note: All [comments] and [explanations] are written in square brackets, so they cannot be confused with (Lisp code).

LISP vs SAL

SAL is a new alternative syntax to LISP. Although Nyquist is based on the LISP programming language, you can write almost any Nyquist program in SAL. Most people prefer SAL syntax to LISP syntax because SAL is a bit more like popular programming languages that include Java, C, and Python.

You can specify which one you're using with the respective :

You can only use one or the other in your script.

Changing the Volume of an Audacity Track

To change the volume of an Audacity track with Nyquist, the easiest way is to use the Nyquist function:

scale(number, sound)

The "scale" function multiplies the amplitude [volume] of the "sound" by the given "number". A number of 0.5 will make the sound become only half as loud as before, while a number of 2 will make the sound become double as loud as before.

Example:

See for an explanation how the Nyquist prompt works.

To run this example in the Nyquist Prompt, ensure that "Use legacy (version 3) syntax" is not selected.
For more information about the *track* keyword, refer to the

1. Create an audio track with some audio [eg a short recording]

2. Now click Tools > Nyquist Prompt. A window with a text field will appear where you can type in:

Important: Do not forget to type the parentheses. The parentheses are part of the Lisp language Nyquist is based on. Without the parentheses the Nyquist Lisp interpreter will not be able to understand your code.

Important: Do not forget to type the parens and comma. These are part of the SAL language.

After clicking "OK" in the "Nyquist Prompt" window the "scale" function will take the Audacity sound and return a "scaled-down" sound with half the volume to Audacity. The result of the last computation of the Nyquist code always gets automatically returned to Audacity.

If you try "scale" with big numbers you will notice that you can return sounds with volumes taller than the Audacity track which will sound very distorted if you play them afterwards in Audacity. So an important lesson to learn is that Nyquist gives you the freedom to do whatever you want but it's now on you to take care that the result will still sound good afterwards.

An alternative command for amplifying a sound is the command.

MULT may be used with numbers, sounds or multi-channel sounds. When using MULT with a sound and a number, each sample value in the selected sound is multiplied by the number, which is essentially the same as using the SCALE command.

Independent Stereo Volume Basics

This page explains how to use Nyquist to change the volume of left and right stereo channels independently.

As discussed previously, the audio data <sound>from an Audacity track is passed to Nyquist in a variable '*track*'. If the track is mono (just one channel) then the <sound> is simply the value given to the variable '*track*'. However, for a stereo track there are two sounds, and these are passed to Nyquist as two elements of an array. The name of the array is '*track*'.

To access a specific element of an array we use the command.

As a stereo track has 2 channels, the array '*track*' has two elements which are numbered 0 and 1. the 0th element is the audio data from the left channel and the other element contains the audio data from the right channel.

Delay Basics

This page explains how to use Nyquist to add a feedback delay effect in Audacity.

Note: All [comments] and [explanations] are written in square brackets, so they cannot be confused with (Lisp code).

To add a feedback delay effect to an Audacity track with Nyquist, the easiest way is to use the Nyquist "feedback-delay" function:

(feedback-delay sound delay feedback)

The "feedback-delay" function applies feedback delay to sound. The delay must be a number (in seconds). The sample rate is the maximum from sound and feedback (if feedback is also a sound). The amount of feedback should be less than 1 to avoid an exponential increase in amplitude. Also since output is truncated at the stop time of

Tutorials

The SCRATCH Symbol Tutorial

This page provides a brief introduction to using the *SCRATCH* symbol in Nyquist programming.

*SCRATCH* is a global symbol, specific to Nyquist in Audacity, which is not deleted in-between plugin runs. It provides a way for information to survive from one invocation of a plugin to the next. However, you should not rely on the "value" of *SCRATCH* beyond a single invocation of a plugin as it could be overwritten by another plugin. It is better to use property lists of *SCRATCH*. That way, you get a whole name space rather than a single variable name, and with careful naming of the property keys, name collisions can be avoided.

To pass data from plugin "effectX-partA" to "effectX-partB":

1. Assign a property name based on the effect name, e.g.: 'EFFECTX [or in SAL, which does not support the single-quote notation of LISP, write QUOTE(EFFECTX). ]

2. "effectX-partA" should delete any old property value:

exec remprop(quote(*SCRATCH*), quote(effectx)) ;; in SAL
(remprop '*SCRATCH* 'effectx) ;; in LISP

3. "effectX-partA" should compute a new property value v and save it:

 exec putprop(quote(*SCRATCH*), v, quote(effectx)) ;; in SAL
 (putprop '*SCRATCH* v 'effectx) ;; in LISP

4. "effectX-partB" should access the property using:

 set v = get(quote(*SCRATCH*), quote(effectx)) ;; in SAL
 (get '*SCRATCH* 'effectx) ;; in LISP

5. When "effectX-partB" finishes, it should remove the property:

But there may be cases where you do some analysis and want to use the analysis data multiple times. You might even have multiple analysis plugins operating on different inputs to collect data to feed into a plugin with multiple inputs. In this case, which might be quite common, you should not call REMPROP(), but this has the problem of leaving data on the *SCRATCH* property list indefinitely.

In cases where *SCRATCH* data is not deleted immediately after use, and where there is the potential to leave large amounts of memory there, there should be another effect, e.g. "effectX-partCleanup", that simply calls:

allowing the user to explicitly free up any data stored on the 'EFFECTX property. It would be reasonable to omit the "effectX-partCleanup" effect if the data stored on the property list has a maximum size of, say, 10KB. The problem we want to avoid is properties with unbounded size getting left in the heap until Audacity is restarted.

Stereo Tracks Tutorial

This tutorial provides a brief introduction to using stereo tracks in Nyquist programming.

If a sound from an Audacity stereo track was given to Nyquist, the *TRACK* variable contains an array of sounds. Because all Nyquist "snd-..." low-level functions only can process mono signals, to use such a function, the *TRACK* array first must be split into single mono signals and afterwards be re-combined into an array before it is given back to Audacity.

In Sal, one could write:

Or in LISP, one could write:

(arrayp *track*) - tests if '*track*' is an array

Additional resources

Audacity custom themes

Audacity supports custom themes in the form of ImageCache.png files.

Installation instructions can be found here:

Audacity themes forum

Themes created by the Audacity community

Creating custom themes

Instructions on creating your own custom theme can be found here:

EQ curves

EQ XML to TXT converter

Convert older EQ presets to use in current Audacity

Download and installation

First, download the plugin below:

Then install the plugin in Audacity via Tools -> Nyquist Plugin Installer and restart Audacity. Detailed steps are available at

Weighted curves

A-weighted curve

A-weighting is a commonly used curve defined in the International standard IEC 61672:2003. It was originally developed for measuring low level noise in audio equipment.

1KB

A-weighting.xml

Open

External scripts

External scripts don't run within Audacity itself. Instead, they communicate with it using mod-script-pipe.

Further information can be found here:

vinyl2digital

For batch processing of vinyls

(if (arrayp *track*)
    (vector
        (snd-function (aref *track* 0))   ; left stereo channel
        (snd-function (aref *track* 1)))  ; right stereo channel
    (snd-function *track*))               ; mono signal

Details

Author: David R.Sky

LFO Surf generator. A signal whose frequency is generally below the human ear's ability to hear as a tone, usually 20 cycles per second [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 new stereo track in Audacity

Parameters:

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 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 more 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 60)

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].

7. Lower filter frequency: [Hz] 8. Upper filter frequency: [Hz] Both the above determine how low and how high the low-pass filter sweeps the surf noise.

9. Bass frequency to boost: [Hz]You can boost the volume of frequencies of the surf sound below this setting, to get a deeper-sounding surf. Somewhat equivalent to the bass knob on your stereo.

10. Bass boost : [dB]Sets how much to boost the above bass frequency. 0 dB means no boost, 6 dB means double the amplitude of the bass frequency, and so on.\

Details

Author: David R.Sky

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 plugin emulates that surf cycle, in either mono or stereo.

Start a new session of Audacity. To generate stereo surf, first open a blank stereo track

Parameters:

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.

6. Sweep starting filter frequency: [Hz]

7. Sweep ending filter frequency: [Hz]The above two parameters set the starting and ending frequencies for the low-pass filter to sweep the sweep portion of Oxygene surf. A low-pass 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 low-pass filter frequency of the crash.

11. Crash bass frequency boost: [dB]How much to increase the volume of the above filter frequency and below. 0 dB means no boost, 6 dB means double the amplitude of this bass frequency, and so on.

12. Post-crash silence duration: [seconds]How much silence before the Oxygene surf cycle repeats.

Prompt Basics

This page explains how to use the Audacity Nyquist Prompt to test-run Nyquist code snippets.

Setting up

The Audacity Nyquist prompt appears in Audacity's "Tools" menu.

For testing generate-type commands in Nyquist, you can use the Generate Prompt plugin instead.

Load a Sound File

Sound files are imported into Audacity via: File > Import > Audio or the shortcut CTRL + SHIFT + I.

If you have no pre-existing sound files to work with, you can create your own mono or stereo tracks via the Audacity "Generate" menu.

The Nyquist Prompt

Select the track(s) and click Tools > Nyquist Prompt.

The Nyquist Prompt appears like this:

Hello, world!

Simply enter the code below and press Apply to receive a "hello world" prompt:

The programming language used above is Lisp. The Nyquist prompt automatically assumes you're using Lisp if the first character is an (. If it isn't, the prompt assumes you are writing SAL, which is more similar to C-like syntax. If you prefer SAL, a "hello world" program looks like this:

The Nyquist Debugger

The prompt also comes with a debug button. It shows you various warnings and non-fatal errors after the program finished. For example, if you forget the quotation marks around "hello world", just hitting apply will just do nothing, whereas hitting debug will tell you where you went wrong:

Basic Nyquist Commands

The current Nyquist manual is here: .

A useful index of Nyquist commands is here:

Audacity uses the *TRACK* variable to reference the current audio file/selection. Thus, you can use basic commands such as mult or sum with *track* and the Nyquist prompt will replace the file/selection with the result (or as Audacity calls it, "returned audio").

Note: Prior to version 4 syntax, the vaiable S was used instead of *TRACK*. You may encounter this in some old plugins or old documentation. The old syntax is obsolete and should not be used in new code

Simple Examples

Note: These examples are focused upon using Audacity to manipulate digital signals (clearly Audacity is better suited to audio, but features such as Nyquist can open many other uses). For those interested, the signal used is an from a remote control.

Applying a DC offset to a signal

Type the following into the Nyquist Prompt (using LISP syntax): (sum *track* 1) ``Or type the following equivalent SAL command: return *track* + 1

The whole signal has now moved up to above zero.

Modulating with a carrier frequency

To multiply a signal with a generated carrier signal, you can use the following commands:

The (hzosc 19000) produces 19kHz sine wave carrier.

The (osc-pulse 19000 0) produces 19kHz square wave carrier (note the 0 is the bias or 50/50 duty cycle, -1 to 1 = 0%-100% pulse-width ). Applying the 19kHz square wave carrier obtains this result.

The top and bottoms of the signal can then be clipped using the Hard Limiter option from the effects menu (0dB limit and Wet level 1) if required.

The above examples show how you can use the many Nyquist commands to perform basic signal processing without using scripts.

Note: Unfortunately, this isn't the end of the road for this sample; it is near, but the curved "head/tail" of the signal causes a problem for the digital signal being produced [and it was also upside-down, too...]. This will hopefully form the basis of some more complex examples, since I shall need to use Nyquist to:

Find the zero crossing points
Then only apply the carrier frequency to those regions above zero. Or find another suitable command...

Property List Tutorial

Variables and Property Lists

In general terms, a variable is a symbol which contains a value. The symbol can be any valid name, and its value may be changed (hence "variable"). In Nyquist, the value may be of any data type (for example, a number, a character, or even a sound) and may be changed from one data type to another. Unlike some programming languages, variables do not need to be declared before use - they can just be set, and then they exist.

In addition to the value of the variable, one or more "properties" may also be attached to the symbol. Each "property" has a name and a value. The properties are known collectively as the symbol's "property list".

Setting the value of a symbol "binds" the value to the symbol. A symbol that has no value (not even "nil") is said to be "unbound".

In addition to the value of a symbol, we can also attach properties. This is a way of associating a list of items, each with their own value, to a single variable. Each item is called a key or indicator, and we can give each key a value. This list of items is called a "property list" (or plist for short).

To get the value of a property, we use the GET command.

When getting the value of a property, we do NOT want to evaluate either the variable (symbol) or the key symbol, so we must "quote" both symbols to prevent evaluation.

Examples

The following examples may be run in the .

For a full list of global properties, see the .

TRACK NAME Property

When the type of a plugin is process or analyze, Audacity sets the value of *TRACK* to the currently selected audio, and sets a lists of properties related to that track. The plugin processes one track at a time in sequence, and the *TRACK* variable is set each time for the track that is being processed.

The value of *TRACK* provides direct access to the selected audio, and its property list provides access to other properties of the track. The NAME property provides the name of the Audacity track that is currently being processed. To access the value of the NAME property, we use the command.

The GET command returns the value of the property (the name of the track), which may be assigned to another variable and used elsewhere in the code. For example, to print a pretty message:

Although we can change the value of the NAME property (using ), doing so will NOT change the name of the Audacity track. The value of the NAME property is only a copy of the track name, created by Audacity when the plugin runs. If required, the name of the track could be changed using the scripting command . More generally, modifying a *TRACK* property does not modify the track.

TRACK CLIPS Property

This property contains a list of start and end times of each audio clip in the track. This property is more likely to find uses in than in standard Nyquist plugins. Note that this property refers to the entire selected track, and not only the selected portion of the track.

TRACK CLIPS data for mono tracks

For mono tracks, the CLIPS property is a list, containing a two element list for each clip in the selected track. A mono track with two audio clips will look like ((s1 e1)(s2 e2)) where s1 and s2 to are the start times of the two clips, and e1 and e2 are the end times.

This code snippet will print the start and end times of the first audio clip in a mono track:

It is important to remember that Nyquist sees the start of the current selection as "time=zero". Thus if we wish to actual track times in Nyquist, we must offset the times by the start time of the current selection. We can create a point label at time=0 with (list (list 0 0 "")), but this is relative to the start of the current selection. If we want to create a label at time=zero as shown in the regardless of where the selection starts, then we must offset the label times by the start time of the selection.

Fortunately it is easy to find the absolute start time of the selection by using "START" property of the *SELECTION* variable: (get '*selection* 'start). We can create a label at an absolute time (relative to Audacity's Timeline) like this:

We can now put this all together and create a label for each audio clip in the selected track:

TRACK CLIPS data for stereo tracks

Just as stereo sounds are represented as an array of sounds, so the track "clip" data for stereo tracks is an array of lists (one list per channel). A stereo track with two audio clips in each channel will look like: #(((s1 e1)(s2 e2))((s3 e3)(s4 e3))) where s1 to s4 to are the start times, and e1 to e4 are the end times.

This code snippet will print the start and end times of the first audio clip in the left channel of a stereo track:

In a similar manner to the mono example, we can create a label for each clip in a stereo track:

;Lisp input: (print hello world)
error: unbound variable - HELLO
if continued: try evaluating symbol again
1> 

;Sal input: retun hello world
>>> parse error: Syntax error.
>>> in NIL, line 1, col 14.

return hello world 
             ^

;type analyze
(setf track-clips (get '*track* 'clips))
(setf start (get '*selection* 'start))

(let (labels) ;initialise "labels" variable with NIL value.
  (dolist (c track-clips labels)
    (setf label
      (list (- (first c) start)
            (- (second c) start)
            ""))
    (push label labels)))

;type analyze
(setf track-clips (get '*track* 'clips))
(setf start (get '*selection* 'start))
(setf labels ())  ;an empty list

(defun add-labels (data text)
  (dolist (c data)
    (setf label
      (list (- (first c) start)
            (- (second c) start)
            text))
    (push label labels)))

(multichan-expand #'add-labels track-clips #("Left" "Right"))
labels ;Reurn labels

MIDI key: [0 - 127, default 60]

Analysis plugins

Plugin Suites

Vamp Plugin Pack

A plugin collection for various Vamp Analyzers. The plugins included are all open source software developed and published by various different authors and institutions:

BBC
Carl Bussey
Chris Cannam
Jamie Bullock

Download page:

Individual Analyzers

ACX Check

This analyzer was developed as an aid for audiobook producers. It displays a number of useful statistics about the selected audio, and compares them to the specifications published by (an Amazon.com subsidiary).

Details

Author: Steve Daulton

Based on a plugin by Will McCown.

Displayed Results

Peak level Maximum peak level in the selection

MAAT GON

A Goniometer, also known as XY-Oscilloscope or Phase Scope.

Note: Requires registration. This Analyzer can be found in the Effects category.

Details

Information from the vendor:

This “phase scope” plug–in quickly conveys global trends and troubles, with a visual out–of–phase warning built in. Plus, an optional autogain feature insures that you’ll see an understandable display with a very wide range of input amplitudes.

Usable metering is essential for any engineer, and GŌN’s options are equally practical, with a notable absence of controls. Unlike an oscilloscope which takes up room on your producer’s desk or meter bridge, GŌN’s lack of any knobs or switches means its straightforward user interface contains only a single button, the preference’s gear icon at upper right. Control over gain, focus, “phosphor” color, drawing style and persistence are all there in the preferences, and everything can be saved as a personal preset.

All of GŌN’s functionality is wrapped in an information-rich yet visually unobtrusive user interface that occupies only a small slice of screen real estate. Likewise, the plugin is very "light weight" demanding an absolute minimum of CPU resources so it won´t slow down your host. As a bonus to everyone in the audio community, we’re giving you this free and rather useful plug-in. Thanks for visiting, and we hope to see more of you in the future!

Peak Finder rft

Either places a single label at the first instance of a peak volume, or multiple labels at all the instances of that peak.

Details

Author: Edgar-rft.

Parameters

Place labels at: [Choice: first peak only, all equally loud peaks (default)] The default setting will create labels for all peaks at the maximum absolute amplitude. "Absolute" amplitude disregards whether the value is positive or negative, so peaks may be up or down.

Pitch Detect

This plugin attempts to detect and display the musical pitch and frequency of the selected note. In most cases the default settings will work best. The other options are provided to handle special cases such as analyzing synthetic signals that are outside of the usual musical range.

Details

Author: Steve Daulton

By default, the plugin detects the pitch by analyzing the first 0.2 seconds of the selection. In most cases this should work well. If required the analyzed section can be set to the first part of the selection ranging from the first 10th of a second (0.1 seconds) up to one second.

Parameters

Frequency range: [Choice: 20-1000 Hz, 100-2000 Hz, 1 kHz-10 kHz. Default 100-2000 Hz] In most cases the default should be used as most musical pitches are in the range 100 to 2000 Hz.

Other sources

Additional Vamp Plugins can be found on

Information From the vendor:

Multi-tap delay, no practical limit on tap count
Up to 10 second delay per tap

MConvolutionEZ

An easy-to-use highly optimized convolution reverb, part of the pack

Details

MConvolutionEZ is an easy-to-use highly optimized convolution reverb. Comes with lots of impulse responses for rooms, halls, plates, guitar cabinets, effects.

Advanced GUI
Compare multiple settings: A to H Switching and A to D Morphing.

MCharmVerb

A Lush algorithmic reverb effect, , part of the pack

Details

MCharmVerb is a great sounding lush algorithmic reverb based on the MTurboReverb engine.

Advanced GUI
Compare multiple settings: A to H Switching and A to D Morphing.

OrilRiver

A free algorithmic stereo reverb

Details

12 early reflections variations
5 reverb tail variations

TAL-Reverb-4

A plate reverb with a vintage 80's character

Details

Information from the developer website:

Modulated vintage reverb sound.
Very diffuse sound.

U-he Protoverb

A natural sounding reverb based on the concept of a room simulator

Details

Information from the developer website:

Protoverb is an experimental room simulator reverb. Most algorithmic reverbs try to avoid resonances or model the reflections of sound from a room’s walls. Protoverb does the opposite. It builds up as many room resonances as possible, modelling the body of air in the room. No need to modulate or colour the signal. The result is a very natural sounding reverberation with interesting characteristics:

Notes held for a longer time tend to build up resonance, as if the air takes a while to get excited Multiple instruments remain distinct, without disappearing in a wash When you play a short melody, the room seems to repeat a ghost echo of that melody These properties are found in churches and large halls, but are rarely in conventional algorithmic reverbs.

To achieve this, Protoverb works with many parallel, serial and networked delays. With such a structure, no mathematical formula can make it sound right, it is down to trial and error (and luck) using random values. Protoverb generates random delay line lengths, networks and feedback strategies.

Voxengo OldSkoolVerb

OldSkoolVerb offers you a comprehensive set of parameters permitting you to achieve various reverbs ranging from plate reverb to room reverb to hall reverb sound.

Details

Information from the developer website:

Plate, room, and hall reverbs
5 reverb modes

Valhalla Supermassive

ValhallaSupermassive lets you create massive reverbs, harmonic echoes and space sounds

Details

Information from the developer website:

ValhallaSupermassive is based around feedback delay networks. The individual delays can have up to 2 second of delay time, with user control over the delay lengths, the feedback, how the delays mix with each other, and the modulation rate and depth of the delays. The sonic results range from choruses and flangers, to echoes that fade in and out over time, to massive lush reverbs, and onwards to weird spatial effects that have to be heard to be believed.

Delay and Reverb

A delay effect is similar to an echo, in that the sound is repeated (after a brief time delay) one or more times after the original sound.

Bouncing Ball Delay

A delay plugin where, like a bouncing ball, the bounces get faster and faster.

Details

Author:

A delay plugin 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:

Decay amount [dB]: how much quieter each bounce is.
Delay time [seconds]:
Number of bounces [times]:

Note: The latest available delay plugin in Audacity includes bouncing ball delay and pitch shifting. but not panning.

Bouncing Ball Delay with Panning

Combines the with a panning effect.

Details

Author:

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.

This plugin has bugs and may produce unpredictable results with some settings, but is basically functional.

Parameters:

Decay amount

Bouncing Ball Delay with Tone Shift

Combines the with Tone Shift plugins.

Details

Author:

Combines the Bouncing Ball Delay and Delay with Tone Shift plugins. 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:

Decay amount [dB]:

Reverse Bouncing Ball Delay

The fastest bounces come first, gradually slowing down - reverse of the effect.

Details

Author:

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

Parameters:

Decay amount: [0.00 - 5.00 dB, default 0.50]

Reverse Bouncing Ball Delay with Tone Shift

The fastest bounces come first - reverse of the , and each bounce is tone shifted.

Details

Author:

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

Parameters:

Decay amount: [0.00 - 5.00 dB, default 0.05]

Chimes delay

Adds random delay to your audio, and randomly changes the pitch of each delay if you specify a note list.

Details

Author: Steve Daulton

Based on chimesdelay.ny by

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

Delay BPM with Panning

Applies a delay effect based on a specified tempo (beats per minute), with optional panning and filtering of the delayed sound.

Details

Author: Felipe Zanabria (based on code by Steve Daulton)

Applies a delay effect based on a specified tempo (beats per minute), with optional panning and filtering of the delayed sound. Requires a stereo track.

Parameters:

Decay: [0 - 24 dB, default -6dB] - how much quieter each subsequent delay is.

Delay with High-Pass Filter

Applies a high-pass filter to a delay so that with each subsequent delay, the filter's cut-off frequency is increased.

Details

Author:

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 plugin 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:

Decay amount: [0 - 24 dB, default zero] - how much quieter each subsequent delay is

Delay with Low-Pass Filter

Applies a low-pass filter to a delay so that with each subsequent delay, the filter's cut-off frequency is reduced.

Details

Author:

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 plugin 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:

Delay with Pitch Change

A delay plugin except, each delay is pitch shifted.

Details

Author:

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

Parameters:

Decay amount: [0 - 24 dB, default 0]

Delay with Stereo Flip

This is a stereo delay effect: with each delay, the stereo channels are flipped from left to right and vice versa.

Details

Author:

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:

Decay amount: [0 - 24 dB]

Delay with Tone Shift

Similar to pitchshift.ny except you can define in semitones how much each delay is to be pitch shifted.

Details

Author:

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).

Both plugins 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.

Dynamics Processing

Broadcast Limiter II

Gives you the possibility to overdrive an audio track without introducing ugly digital distortion noise.

Details

Author: Edgar-rft

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

Parameters:

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

Broadcast Limiter III

Similar to Broadcast Limiter II with an added Exciter

Details

Author: Edgar-rft

Is in principle a similar "soft clipping" effect as , 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:

Exciter: controls the high-range distortion in linear numbers from 1 to 10

Hyperexp

The Hyperexp effect is a type of compression.

Details

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".

Level Speech

This effect is designed to mitigate problems in speech recordings where there are very large variations in the loudness / amplitude of the recorded voice or voices.

Details

Author: Steve Daulton

This effect is designed to mitigate problems in speech recordings where there are very large variations in the loudness / amplitude of the recorded voice or voices. A typical situation might be a conference recording where one person's voice is much louder than the other people present.

Technically, it is a special kind of "dynamic range compression" effect, to "level" out variations in amplitude (reduce dynamic range), particularly for speech recordings. Usage of the effect is greatly simplified compared to most traditional compressor effect, by combining control of all of the effect parameters into a single "Leveling Amount" control.

Parameters:

Limiter (Audacity <3.6)

Use the Limiter effect to pass signals below a specified input level unaffected or gently reduced, while preventing the peaks of stronger signals from exceeding this threshold. Mastering engineers often use limiting combined with make-up gain to increase the perceived loudness of an audio recording during the audio mastering process.

This limiter effect provides two basic types of effect; "limiting" and "clipping". The "limiting" effect is a special kind of that responds very rapidly to peaks in the waveform. The "clipping" effect is a kind of that changes the shape of the waves by "clipping" off the high and low peaks.

Details

Author: Steve Daulton

Usage:

Type

Soft Limit (default) progressively reduces the as the amplitude of the waveform approaches the threshold and prevents the waveform from exceeding that level.

Limiter (legacy)

A "lookahead" dynamic range limiter to compress peaks that extend beyond the set threshold value.

Details

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:

Limiter (2)

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

Details

Author: Steve Daulton.

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

Parameters:

Limit to (dB): (-10 dB to 0 dB)

Noise Gate

Noise Gates may be used to cut the level of noise between sections of a recording.

Details

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:

Select Function: [Apply the Noise Gate effect | Test the noise level | View one of the Help screens].

This effect requires the entire audio selection to be loaded into RAM. If there is insufficient available memory, the plugin and Audacity will crash. The maximum length that can be processed is dependent on the sample rate, length of audio selection, operating system, and available RAM. Please test carefully before using this effect on long (> 45 minutes) tracks to verify the limits on your computer.

Pop Mute

The effect is like an "upside-down" . Whereas a attenuates sounds that are below a specified threshold level, Pop Mute attenuates sounds that are above a specified threshold level.

Details

Author: Steve Daulton

The effect is like an "upside-down" . Whereas a 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.

Text Envelope

Provides an alternative to the "Envelope Tool" that is accessible for visually impaired and other users that do not use pointing devices

Details

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:

Select function:

Plugin Reference

This page offers a detailed look at the structure and syntax of Nyquist Plugins. It is intended for people who wish to write their own plugins.

If you are looking for extra Nyquist plugins to use, see Nyquist Plugins.
If you are especially interested in Nyquist in Audacity, we highly recommend subscribing to our Audacity forum, which has a section specifically for matters relating to Nyquist.

Overview

Nyquist is a superset of the XLISP programming language, (a dialect of LISP and supports both a LISP syntax and an alternative syntax called SAL. A general introduction to Nyquist is given on the main Nyquist page.

The Audacity Nyquist Interface is implemented in the following files in the Audacity source code:

audacity/src/effects/nyquist/Nyquist.cpp
audacity/lib-src/libnyquist/nyx.c

Nyquist is in many ways a separate program from Audacity (and is also available as a ). Nyquist is embedded in Audacity as a software that may be called from the main Audacity application to provide additional functionality, most notably as "plugins".

When a Nyquist plugin is selected from one of Audacity's menus, Audacity locates the plugin script (the ".NY" file), starts a Nyquist session and passes the Nyquist code to be interpreted by the Nyquist library. Nyquist then attempts to run the program. The result of running the Nyquist script is returned to Audacity.

Nyquist Plugin Header

As in most other computer languages, Nyquist supports code comments. Any line beginning with a semi-colon (;) is entirely ignored by Nyquist. Comments may also be placed at the end of a line of code by beginning the comment with a semi-colon. Nyquist ignores everything from the semi-colon to the end of the line.

In Audacity, special comments are used in Nyquist plugins to pass information to Audacity. As with other code comments, these are ignored entirely by Nyquist, but provide instructions to tell Audacity how to create the plugin.

Plugin Header Example

In this example (derived from the effect), the first four lines make up the essential headers that are required in all Nyquist plugins.

;nyquist plug-in This header tells Audacity that this file is a Nyquist plugin.
;version 4 This header specifies that this is a "4th generation" plugin.
;type process This header defines the plugin as a "process" (Effect) type.

The other headers enable various options and properties, including the plugin's controls.

Full descriptions of all plugin headers are provide on the page.

Nyquist Plugin Widgets

Nyquist plugins support a range of "widgets", which are control elements for a graphical user interface (GUI). These elements are unique to Nyquist in Audacity.

The GUI is created by Audacity when a plugin contains one or more widget header commands.

As with other plugin headers, they begin with a semicolon (;) and are ignored by Nyquist, except that the variable name of the widget is initialized with a value and made available to Nyquist. This means that Nyquist can access the value that has been set in the widget as the value of the variable specified in the widget header.

Note: The "Text widget" is an exception to the above at it is for display purposes only and does not set a value.

Each widget begins with ";control", and is followed by a number of parameters, one of which defines the type of widget. There are currently nine widget types, though not all of them are available for old versions of Audacity.

When a plugin is launched, Audacity searches the plugin .NY file for valid headers. Every ";control" line that is found gets parsed by Audacity into several tokens, where each token is separated by spaces. If the tokens match one of the defined patterns, then Audacity adds the widget to the GUI, otherwise they are ignored.

Syntax for widgets

;control

var-name

text-left

widget-type

text-right

initial-value

minimum

maximum

Note: "real" (deprecated) is an alternative name for "float" and is provided as legacy support for old plugins. It should not be used in new code.

Italic words in the table denote data types. Because tokens are separated by whitepace, strings containing whitespace must be written within quotation marks.

Note: Older versions of Audacity may not support all of these controls, which may lead to an "unbound symbol" error. Plugin users are encouraged to use the current version of Audacity, to ensure that they can benefit from all of the latest features.

The following code may be run in the of Audacity 2.3.1 or later, and produces the GUI shown above:

The detailed syntax for each widget type is described on the Nyquist Plugins Widgets page.

Tracks and Selections

Audacity passes information about the current Audacity session to the Nyquist interpreter, along with the Nyquist code. Typically, when the code runs, it acts on data that has been passed from Audacity, which often includes the selected audio data, and returns data back to Audacity. Here we shall look at the ways that audio data is passed from Audacity to Nyquist, and from Nyquist back to Audacity.

Passing Selected Audio to Nyquist

For process and analyze type plugins, Nyquist runs the plugin code on each selected track in turn. The audio data is passed from Audacity to Nyquist as a variable called *TRACK* (the asterisks are part of the name).

For mono tracks, the value of *TRACK* is a "sound" (which is a Nyquist ).
For stereo tracks, the value of *TRACK* is an "array". An array is a special kind of ordered list. The array has two elements, both of which are "sounds". The first element holds audio data from the left channel, and the second element hold data from the right channel.

The selected audio is only available to "process" and "analyze" type plugins (including the compound types "tool process" and "tool analyze").

Time and Durations

The way that time and durations are handled depends on the type of plugin.

For process and analyze type plugins, the start time of the selection is seen as "time = 0", and the length of the selection is seen as one unit of time. The absolute length of the sound in seconds can be computed one of the following ways:

NOTE: get-duration answers the question: "If a behavior has a nominal duration of 1, how long will it be after warping it according to the Nyquist environment?" Since many built-in behaviors like OSC and LFO have nominal durations of 1, In process effects, Audacity sets up the environment (including warp) to stretch them by the selection's duration. Otherwise, if you wrote (OSC C4), the result would have a duration of one second instead of the duration of the selection.

In 'generate' effects, this does not happen, so the length specified by the effect is the length that is produced. For example, if you write (OSC C4 3.5), a generate type effect will produce a tone of duration 3.5 seconds.

For generate type plugins, the length of the selection (if there is a selection) is usually ignored by Nyquist. However, if for some reason the length of the selection needs to be known, then *SELECTION* START and END properties may be used (version 4 plugins or later).

When generating sounds with a 'generate' type plugin, durations in Nyquist represent seconds. For example, to generate a 2.5 second sine tone, you could write:

but if the same code is used in a 'process' type plugin, the generated tone will be 2.5 x the duration of the track selection:

If a duration in seconds is required in a 'process' type plugin, this may be done using the command:

The above examples may be run in the .

Global Variables and Reserved Variable Names

In addition to the standard global variables defined in the , Nyquist in Audacity also has global variables that relate specifically to Audacity. Listed here are a few common standard Nyquist globals, and global variables that are relevant to Nyquist in Audacity.

: [float] The sample frequency of the control signals (such as those created by . By default 1/20 of the sample rate of the track.
*DECIMAL-SEPARATOR* : [char] A comma character (#\') or dot character (#\.) depending on the language selected in .
*FILE-SEPARATOR* : [char] The character that separates directories in a path, e.g. "/" (#\/) for Unix, ":" (#\:) for Mac, and "\" (#\\) for Win32.

Other global variables provided by Nyquist can be found in the .

Global Property Lists

Property lists are defined for the global variables *AUDACITY*, *PROJECT*, *SELECTION*, *SYSTEM-DIR*, *SYSTEM-TIME*, and *TRACK*.

For examples using property lists, see the .

AUDACITY

This property list was added in Audacity version 2.1.0

Value: Unbound (not defined).

LANGUAGE : [string] The country code for the language set in . Example - when Audacity's locale setting is for English, (print (get '*audacity* 'language))will print en.
VERSION : [integer list] A list in the form (Audacity_version Audacity_release Audacity_revision) Example: Print the full Audacity version for Audacity 2.1.3 (let ((version-list (get '*audacity*' version))) (format nil "Audacity version ~a.~a.~a" (first version-list)(second version-list)(third version-list))) Prints: Audacity version 2.1.3

PROJECT

Value: Unbound (not defined).

LABELTRACKS : [integer] The number of label tracks in the project.
MIDITRACKS : [integer] The number of note tracks in the project.
NAME : [string] The name of the current project. A project that has not been named (not yet saved) returns an empty string.

SELECTION

Value: Unbound (not defined).

BANDWIDTH : [float] The bandwidth of the frequency selection in octaves (Spectrogram or Spectrogram (log f) track view).
CENTER-HZ : [float] The centre frequency selection in Hz (Spectrogram or Spectrogram (log f) track view).
CHANNELS : [integer] The number of selected audio channels.

SYSTEM-DIR

Value: Unbound (not defined).

BASE : [string] The Audacity installation directory.
DATA : [string] The Audacity data directory. This is where the audacity.cfg, pluginregistry.cfg and EQCurves.xml files are located.
DOCUMENTS [string] The system default documents directory.

SYSTEM-TIME

This property list was added in Audacity version 2.1.1.

Value: A list in the form '(year, day, hours, minutes, seconds), where each list item is an integer.

Example: 5 minutes past 2pm January 1st 2022 would be the list: 2022, 1, 14, 5, 0.

Note that "day" is a number in the range 1 to 366, counted from the start of the year.

This is the time that the Nyquist code is parsed, NOT the time it is executed. It cannot therefore be used for timing events that occur while the code is running. Possible uses for *SYSTEM-TIME* could include automatic naming of files written to disk, or for "" random processes.

DATE : [string] The date formatted according to the current locale. Example: "dd/mm/yy".
DAY : [integer] Day of the month.
DAY-NAME : [string] The name of the day (Example: "Monday").

TRACK

Value: The sound from a selected mono audio track, or an array of two sounds from a selected stereo track (see the Nyquist Stereo Track Tutorial).

Properties of *TRACK* all relate to the track that is being processed. Currently Nyquist only processes audio tracks. When multiple tracks are selected, Nyquist processes each audio track in turn (top to bottom).

CHANNELS : [integer] The number of channels in the track (mono = 1, stereo = 2).
CLIPS : [list or array] For mono tracks, a list of start and end time of each audio clip. For stereo tracks, an array containing a list of clips for each channel. Due to a limitation in Nyquist, the "clips" property can hold a maximum of 1000 start / end times. If an audio channel contains more than 1000 clips, the first 1000 will be listed, and the 1001th item will be NIL. See for an alternative way to get clip times that will work with 1000's of clips.
END-TIME : [float] The track end time.

The VIEW property may change in future versions of Audacity, so is not recommended for public release plugins. During Preview, audio tracks are copied to temporary tracks which are not visible, so the returned "VIEW" value is NIL.

Return Values

Nyquist supports many "data types", including "numbers" (integer or floating-point), "characters" (such as the letter "A", the number "4", or any other ), "strings" (text), "" (a list of data), "" (special kind of indexed list), and "sounds" (a sound / digital signal).

The result of the last computation within the plugin code will be given back from Nyquist to Audacity. According to the data type of the returned value one of the following actions will be invoked in Audacity:

Mono Sound

If a "sound" is returned, the sound will be re-inserted into the selected part of the Audacity track, (or a new track for "generate" type plugins). If the returned sound is shorter or longer than the original sound, the selection will be reduced or augmented. If a mono sound is returned to a stereo track, the same mono sound will be inserted into both channels of the stereo track.

Multi-Channel / Stereo Sound

Nyquist handles multi-channel sounds as an array of sounds. The first element of the array is the left channel, and the second element is the right channel. Audacity currently supports a maximum of two channels in a track (stereo).

Returning an array of sounds to a mono track is an error.To return a stereo sound without error, a stereo track must be selected before running the Nyquist code.

For more information about stereo tracks, see the .

String / Text

When the return value is a character or string, a dialog window will appear with the data displayed as text.

Number

A dialog window will appear with the number displayed as text.

Labels

If an appropriately formatted list is returned to Audacity, a label track will be created below the audio track(s).

For point labels the format is:

((number ``"string") (number ``"string") ... )

The list to create a label track must contain one or more lists, each of which must have:

number - (an integer or float) is the time in seconds from the beginning of the Audacity selection, where the label will appear.
"string" - a string to be displayed in the label's text field.

For region labels, each label list must contain two int-or-float elements, one for the start and one for the end of the label region.

((number````number ``"string") (number````number ``"string") ... )

Audacity will always place returned audio or labels relative to the start of the and not the start of the .

Empty String

An empty string may be used as a "null return", which means that the plugin returns nothing, and no error. An example use would be if you wish to process only the first selected track.

Playing sounds with Nyquist

When using the or (most) Nyquist plugins that have a , if the plugin returns a sound, it may be previewed using the Preview button. In addition to this, it is also possible to play sounds directly from Nyquist, using the function. The PLAY function will be executed when the plugin code runs.

The PLAY function is described in the . The information here describes aspects that are specific to Nyquist in Audacity.

Note that the Nyquist PLAY command does not use Audacity's . Nyquist uses the system default device for playback, and has a default buffer of 100 ms. The buffer setting may be changed with the command. The actual latency values are dependent on the computer sound system and may differ from those requested by Nyquist.

To see the list of available devices when Nyquist plays, set *snd-list-device* to "true" before the play command, then use Debug button to see the output.

(setf *snd-list-devices* t) (play *track*)

To select a specific output device, set *snd-device* to the index number of the device you wish to use (not all devices are valid). For example, if ALSA hw(0,0) is device 0, select it with:

(setf *snd-device* 0) (setf *snd-list-devices* t) (play *track*)

It is not possible to interrupt Nyquist playback. The sound will continue playing until it reaches the end, and other Audacity functions are disabled until Nyquist playback has finished.

To limit the amount of audio that will play, the length of the sound must be defined before starting playback. For example, to play the first 1 second of a selection, you could write (LISP syntax):

Note also that if a plugin uses Nyquist PLAY command, using Previewwill cause an error on some machines because Audacity may not be able to access the audio device while it is being used by Nyquist. For "own use" plugins, if you have more than one sound card, a possible workaround to enable Preview and Nyquist PLAY simultaneously, is to set the Nyquist playback device to a different device to that which Audacity uses. For public release plugins the Preview button should normally be disabled if Nyquist PLAY is used.

Plugin Translations

Nyquist plugins provide two mechanisms for translation into the language specified in Audacity's preferences.

One method is exclusively for plugins that are shipped by default with Audacity and requires compiling Audacity from source, plus several other steps involving . Unfortunately this method is not documented, so only a brief description is provided here.

The other method may be used by plugin developers for their "third party" plugins, but can be used only for returned strings and not for the main plugin interface.

Translation for shipped effects

Header comments in translated plugins begin with a dollar "$" character rather than the usual semicolon ";" character. The strings to be translated are then wrapped in a function that has a one character name "_" (underscore).

Example: In the effect, the header is changed from:

;name "Adjustable Fade"

to:

$name (_ "Adjustable Fade")

;control lines become $control. As with other translated headers, the $ symbol replaces the normal semicolon. The line is ignored by Nyquist as a comment, but is visible to Audacity for translation and for creating a .

The options in Multiple-choice widgets may be translated. For example:

;control var (_ "Translated left text") choice ((_ "Yes") (_ "No")) 0

The above example is not safe to use. See the "Limitations" section below.

Macros and portability: To allow portability of settings and macros, choices may include a non-translated form of each choice. The non-translated form is used by Macros but is not normally visible to users.

;control var (_ "Translated left text") choice (("Label1" (_ "Translated Label 1")) ("Label2" (_ "Translated Label 2"))) 0

Other user visible strings: Other strings that are visible to users are marked for translation using the "underscore" function. An example from the Adjustable Fade plugin:

(format nil (_ "~aPercentage values cannot be negative.") err)

Limitations

Control characters like (new line), (tab) are not supported. The workaround is to use Lisp format directives instead:

Bad (does not work correctly):

(print (_ "Line one.\nLine two."))

Good:

(print (format nil (_ "Line one.~%Line two.")))

Note that translations may contain Unicode characters that are not supported by Nyquist. It is therefore important that any translated strings that need to be machine readable (such as Multiple-Choice widget options) should also have a non-translated form.

;control var (_ "Translated left text") choice (("Label1" (_ "Translated Label 1")) ("Label2" (_ "Translated Label 2"))) 0

Translation for third party effects

It is not currently possible to provide translations for a third party plugin's main interface, unless the name is the same as a translated Nyquist plugin that is already shipped with Audacity. It is highly recommended to NOT reuse the same name as a shipped plugin for a third party plugin.

Translations may be provided for return value messages and debug messages. To do this, the string must be marked for translation by wrapping it in the "underscore" function. The translation must also be provided in a specially formatted list variable *locale*.

The format for the *locale* list is: (LIST (language-list) [(language-list) ...])

where "language-list" is a list in the form: (LIST country-code (translations))

and the "translations" are list in the form: (LIST (List "string" "translation") [(List "string" "translation")...])

The *locale* list may contain any number of "language-list" elements, where each language-list is a list of two strings.
The first element of each "language-list" is the country code (for example "en" or "fr"),
The second item of each "language-list" is a list of translation pairs.
A "translation pair" is a list containing the default string (usually in English) and the translated string (in the language specified by the country code).

A working example may be found in the .

Filters and EQ

Band-Stop Filter

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

Details

Author: Steve Daulton

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

Parameters:

Center Frequency: 100.0 to 10000.0 [Hz]. Set the 'Center Frequency' slider, or type in a value for the center of the frequency band to block.

Stop-Band Width: 0.250 to 4.000 [Octaves]. 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.

Chebyshev Type I Filter

Type I Chebyshev filters can provide a steeper than but at the expense of more in the .

Details

Author: Kai Fisher

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

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

Parameters:

Classic EQ

An Equalizer (EQ) that can modify more than one band at a time.

Details

Authors: Josu Etxeberria and David R.Sky.

An Equalizer (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.

Comb Filter

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.

Details

Author:

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:

Customizable EQ

Customizable single-band equalizer.

Details

Author:

Parameters:

Center frequency: [Hz, 20 - 20000, default 440]
Band width in octaves

Desk EQ

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

Details

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 "". The two "mid" filters are bell shaped peak/dip filters which affect frequencies around a center 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:

High Pass Filter with Q

A high pass filter with Q, or resonance.

Details

Author:

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 and low pass filter (LFO) for ability to modulate a fixed resonance cut-off frequency.

Parameters:

Cut-off frequency: [20 - 10000 Hz, default 1000]

High Pass Filter (LFO)

A high pass filter with a low frequency oscillator (LFO).

Details

Author:

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 plugin modulates the cut-off frequency up and down, like on an electronic synthesizer.

Parameters:

LFO frequency: [0 - 20 Hz, default 0.2] - defines the speed of the oscillation, higher is faster

High Pass Filter (LFO) - Alternative version

Details

Author:

Parameters:

Center cut-off frequency: [20 to 20000 Hz, default 640]

Hum Remover

A filter for removing the sound of from recordings.

Details

Author: Steve Daulton

A filter for removing the sound of from recordings.

The frequency of mains electricity is 60 Hz in the US, 50 Hz in Europe. This can create electrical interference in recordings with many . To remove the hum, this effect applies a series of notch filters based on the frequencies of mains electricity and the harmonics, which have frequencies that are at exact multiples of that frequency.

To minimize loss of audio data, the number of harmonics may be adjusted so that only as many notches as required to eliminate the audible hum are applied. There are often more odd harmonics than even harmonics, so this effect allows the number of odd and even harmonic filters to be set independently.

Unless the amount of hum is very bad, high level audio will often mask the hum, making removal unnecessary, but during quiet parts of the recording the hum may be unpleasantly obtrusive. This effect therefore has a threshold level control so that only quiet sounds (where the hum will be most noticeable) are filtered.

The effect can often provide a useful guide as to which frequencies need to be removed.

First, select 50 or 60 Hz with the first control as appropriate, then set the other controls to maximum. Preview the effect frequently while reducing one control at a time to find the minimum settings required to remove the hum.

Low Pass Filter (LFO)

A low pass filter with a low frequency oscillator (LFO).

Details

Author:

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 plugin modulates the cut-off frequency up and down, like on an electronic synthesizer.

Parameters:

LFO frequency: [0 - 20 Hz, default 0.2] - defines the speed of the oscillation, higher is faster

Low Pass Filter (LFO) - Alternative version

Details

Author:

Parameters:

Center cut-off frequency: [20 20000 Hz, default 640]
LFO depth (radius):

Low Pass Filter with Q

A low pass filter with Q, or resonance.

Details

Author:

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 and for ability to modulate a fixed resonance cut-off frequency.

Cut-off frequency: [20 - 10000 Hz, default 1000]

Multiband EQ

A multiband equalizer.

Details

Author:

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.

Mutron

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

Details

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:

Center/Cut-off: [0 - 10000 Hz, default 100] - sets the static filter frequency

Notch Filter

A notch filter cuts out a "notch" in the spectrum of your audio.

Details

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 .

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:

Parametric EQ

A parametric equalizer is a variable equalizer effect which provides control of three parameters: amplitude, center frequency and bandwidth.

Details

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 plugin 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:

Frequency (Hz): [10 to 10000 Hz, default 1000 Hz] - sets the center frequency of the filter

Random Low Pass Filter

Like someone is playing around with the cut-off frequency knob of your low pass filter.

Details

Author:

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:

Max filter sweep speed: [0.01 - 10.0 Hz, default 0.2] - maximum speed of the random filter cut-off changes

Resonant Filter

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

Details

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:

Shelf Filter

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

Details

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:

Ten Band EQ

An Equalizer (EQ) that can modify one band at a time.

Details

Author:

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

Vocal reduction and isolation

Warning: This effect can cause data loss on macOS. See for details.

This is the vocal reduction and isolation effect that was included in Audacity prior version 3.5.

Details

Vocal Reduction and Isolation attempts to remove or isolate center-panned audio from a stereo track. Vocals are sometimes (but not always) recorded in this way. The simplest and quickest removal method subtracts one channel from the other, but the result will be (dual) mono. This method is available at "Remove Center Classic: (Mono)" under Action. All other "Remove" options in this effect preserve the stereo image.

The Isolate options all return (dual) mono output. The narrowness of the center slice can be adjusted with the "Strength" slider. Center isolation is not possible using the classic subtraction method.

The Analyze option displays the amount of correlation between the stereo channels and the degree of likelihood that center removal or isolation will work effectively.

WARNING: Note carefully that when you apply an effect to a time-stretched clip the changed speed of the clip will be automatically

Scripting reference

This page lists scripting commands.

Yes these are the same commands as in the menus, same as in "Commands and Shortcuts" too, just presented differently.

Presented in a format useful to people using:
- Macros or
- Python Scripting or
- AUD-DO.
Most of this page is automatically generated.

A few commands are intentionally omitted from the Macro Manager (such as "Close:") because they are unsuitable for use in Macros.

This table shows the items in the menus that are scriptable.

The Scripting Ids, parameters and defaults are all likely to change between versions.
Boolean values must be given as 1 (true) or 0 (false).

The File Menu provides commands for creating, opening and saving Audacity projects and importing and exporting audio files

Scripting Id

Action

Parameters

Description

File: Save Project

Various ways to save a project.

Scripting Id

Action

Parameters

Description

File: Export

For exporting audio files

Scripting Id

Action

Parameters

Description

File: Import

For importing audio files or label files into your project

Scripting Id

Action

Parameters

Description

The Edit Menu provides standard edit commands (Undo, Redo, Cut, Copy, Paste and Delete) plus many other commands specific to editing audio or labels

Scripting Id

Action

Parameters

Description

Edit: Remove Special

For more "advanced" removal of audio

Scripting Id

Action

Parameters

Description

Edit: Clip Boundaries

Create or remove separate clips in the audio track. A clip inside an audio track is a separate section of that track which has been split so that it can be manipulated somewhat independently of the other clips in the track.

Scripting Id

Action

Parameters

Description

Edit: Labels

These commands are to add and edit labels.

Scripting Id

Action

Parameters

Description

Edit: Labeled Audio

Labeled Audio commands apply standard Edit Menu commands to the audio of one or more regions that are labeled. The labels themselves are not affected.

Scripting Id

Action

Parameters

Description

Select Menu has commands that enable you make selections of tracks or parts of the tracks in your project.

Scripting Id

Action

Parameters

Description

Select: Tracks

Tracks

Scripting Id

Action

Parameters

Description

Select: Region

For modifying, saving and restoring a selection.

Scripting Id

Action

Parameters

Description

Select: Spectral

For making a selection of a frequency range.

Scripting Id

Action

Parameters

Description

Select: Clip Boundaries

For modifying a selection, taking account of clips.

Scripting Id

Action

Parameters

Description

View Menu has commands that determine the amount of detail you see in all the tracks in the project window. It also lets you show or hide Toolbars and some additional windows such as Undo History.

These commands control playback in Audacity. You can Start, Stop or Pause playback of the audio in your project.

Scripting Id

Action

Parameters

Description

Transport: Recording

These commands control recording in Audacity. You can Start, Stop or Pause recording in your project. You can either start a recording on your existing track or on a new track.

Scripting Id

Action

Parameters

Description

Transport: Scrubbing

Scrubbing is the action of moving the mouse pointer right or left so as to adjust the position, speed or direction of playback, listening to the audio at the same time - a convenient way to quickly navigate the waveform to find a particular event of interest. Speed changes are made by rotating the mouse wheel while scrubbing.

Scripting Id

Action

Parameters

Description

Transport: Cursor to

These commands let you move the cursor to the start or end of the selection, track or any adjacent Clip that you may have

Scripting Id

Action

Parameters

Description

Transport: Transport Options

This submenu lets you manage and set various options for transport (playing and recording) in Audacity

Scripting Id

Action

Parameters

Description

Sorts all tracks in the project from top to bottom in the project window, by Start Time or by Name.

Scripting Id

Action

Parameters

Description

Generate Menu lets you create audio containing tones, noise or silence.

Scripting Id

Action

Parameters

Description

Generate: Built-in

Shows the list of available Audacity built-in effects but only if the user has effects "Grouped by Type" in Effects Preferences.

Scripting Id

Action

Parameters

Description

Generate: Nyquist

Shows the list of available Nyquist effects but only if the user has effects "Grouped by Type" in Effects Preferences.

Scripting Id

Action

Parameters

Description

Audacity includes many built-in effects and also lets you use a wide range of plugin effects.

Scripting Id

Action

Parameters

Description

Help: Diagnostics

A set of diagnostic tools

Scripting Id

Action

Parameters

Description

These are additional commands that do not appear in any menu

Scripting Id

Action

Parameters

Description

Playback equalization for 78 rpm shellacs and early 33⅓ LPs

The audio on almost every phonograph record is not the same as that originally performed. For technical reasons the signal's frequencies need to be modified when cutting the disc. Playback equalization (EQ or de-emphasis) is necessary to restore the signal's original frequencies. Only thus can music lovers enjoy the original sound of the music performed long ago from their rare discs.

Usage

The most relevant EQ curves are presented as Table 1:

You can download some of them from the section of this page, and import them into Audacity using or
You can generate any curve yourself with the plugin
You can set the sliders of any digital or analog graphic equalizer manually
You can determine appropriate settings for any adjustable analog pre-amplifier.

Which EQ curve will be needed for a specific record label is answered:

for 78 rpm shellacs in
for early LPs in

Using EQ curves in Audacity

Once , you can import the curve in the Filter Curve EQ or Graphic EQ effects, by clicking Presets & settings -> Import....

Equalization (EQ) Curves explained

When phonograph records are made, the sound being recorded is deliberately distorted by reducing the volume of the low frequencies and increasing the volume of the high frequencies. This process, known as 'pre-emphasis', allows the low frequencies to be accommodated in the limitations of the record groove and reduces the effect of high frequency surface noise. If pre-emphasis was not carried out, the bass notes in the music would create a groove in the record that oscillated so wildly that the stylus could jump out of it on playback, and the treble notes would be drowned out by the surface noise of the stylus in the groove.

On playback, the pre-emphasis must be reversed in order to restore the original sound. This is known as 'de-emphasis' or equalization (EQ).

Modern vinyl records use a method of pre-emphasis and de-emphasis adopted by the Recording Industry Association of America (RIAA) in the 1950's, and the EQ curve used is known as the RIAA curve. However, before the RIAA curve was adopted, each record label used its own EQ curve for recording and, for these records (78rpm and early vinyl), the correct EQ curve must be used for playback.

Each EQ curve is a combination of two filter characteristics; a bass boost curve, defined by a 'Bass Turnover' (or 3dB) frequency, and a treble cut curve, defined by a '10 kHz Gain Roll-off' parameter, i.e. a defined level of treble cut at 10 kHz.

As an example, Figure 1 below shows the characteristic of the bass boost curve defined by a 500Hz Bass Turnover, and Figure 2 shows the characteristic of the treble cut curve defined by a 10 kHz Gain Roll-off of -13.7dB. These two curves, when combined, give the characteristic shown in Figure 3.\

The EQ curve may also include a Low Frequency Shelving filter, (although it is absent from the definition of most 78 rpm EQ curves). This addition reduces the effect of the bass boost at very low frequencies (typically 50 or 100 Hz) in order to attenuate low-frequency noise such as turntable rumble. Figure 4 shows the effect of a 50 Hz LF shelving filter being added to the curve in Figure 3. This is in fact the RIAA standard EQ curve.

Acoustically recorded (pre-electric) 78 rpm records have a completely different characteristic because they were cut with a different type of cutter (For more details see: Acoustic recordings). In some early EQ curves of electrically recorded shellacs, while there is a bass boost curve, there is no treble cut necessary – i.e. the 10 kHz gain roll-off is zero.

Note that in the combined EQ curve, the gains at the Bass Turnover frequency and at 10 kHz may be different from those specified by the parameters. This is not an error, but is due to fact that the gains of the bass boost and treble cut curves are simply added together.

Because 78 rpm EQ curves were non-standard and, in many cases, accurate records were not kept to show what EQ curves were used when recording 78s, there is a degree of uncertainty about what is the correct playback EQ for many record labels. The tables below have been obtained from websites which, in their turn, have compiled data from a number of sources and should prove reasonably accurate. However, the ear of the listener is the final arbiter - if it doesn't sound right, it isn't right!

According to NAB standards, the nominal speed of a 78 rpm record is precisely 78.26 rpm +/- 0.3% (for North America and other countries with an utility frequency of 60 Hz). According to British Standards Institution it is 77.92 rpm +/- 0.5% (for Britain, Europe and other countries with a mains frequency of 50 Hz).

EQ Curves

Pre-equalization of most records – especially of shellacs – was always determined by the cutter head used and often by internal regulations of the record company. Both left quite some room for the recording engineer to make changes to improve the sound. Also if Graumann uses 250 Hz in playback of an EMI disc and Copeland votes for 300 Hz this is not a contradiction. Both mean the same EQ curve but have different opinions on what sounds best. This should encourage you to try both versions and take the one which sounds right to your ears.

Please do not worry about fractions of a dB! Still in the 1960s an accuracy of a curve of +/- 2 dB was considered to be standard. In the 1950s +/- 3 dB were a very fine result and nobody will ever know if recording engineers in the 1940s or 1930s applied their curves correctly (or if they applied them at all!) So the spread in pre-equalization during recording will outnumber any bias in playback equalization by far.

Table of EQ Curves

The most relevant EQ curves are presented in Table 1. All curves are described from the point of view of a playback or de-emphasis curve, where bass / low frequencies must be amplified / boosted and where treble / high frequencies must be attenuated / cut in order to achieve the original sound that had been recorded. The corresponding pre-emphasis curve used for cutting the master disk is inverse.

When comparing with lists provided by the manufacturers of equalizer pre-amps it should be considered that those may be misleading, as they might not quote the correct parameters of the curve but rather the next-best possible settings of these devices. For example the Conductart OWL 1 Sound Restoration Module – a renowned pre-amp of the 1980s – offered settings of flat/ 250/ 375/ 500/ 750/ 1000/ RIAA for turnover and flat/ 5/ 8.5/ 12/ 14/ 16/ RIAA for roll-off. Thus for the widely used standard “AES 400N-12” settings of 375 Hz and -12 dB were listed; for the “Bartok 629C-16” curve it was 750 Hz and -16 dB and for “LONDON LP 500C-10.5” it was 500 Hz and -8.5 dB. These recommendations were quoted or copied by other authors and three “new characteristics” had come to existence.

Table 1 gives the three parameters to characterize any EQ curve: the turnover frequency f1 for bass shelf, f2 for bass boost and f3 for treble cut (or alternatively the three corresponding time constants τ1, τ2 and τ3). These are the necessary conditions to compute and plot any EQ curve, determine the correct settings of a digital equalizer or to solder an electronic filter circuit.

The gain at two typical frequencies will give you a rough impression of what the curve does to the audio from the record. The frequencies are:

50 Hz, where usually the bass shelf becomes effective
10 kHz, because the treble curve is often indicated by roll-off at 10,000 Hz

The very descriptive “code” is a good tool to avoid misunderstanding when EQ curves come under various alias names (what they do too often).

The first 3-digit number indicates the turnover frequency of bass boost (f2)
The letter in the middle is N … (“None”) if no bass shelving is applied, or R … 20 dB (named R after RCA or RIAA) B … 18 dB A … 16 dB C … 14 dB (named C after Columbia LP curve) X … 12 dB
The last number shows the reduction or roll-off of treble at 10 kHz and is always preceded by a minus sign. Please mind that this is only a code and that the exact value – after normalization to 0 dB at 1000 Hz – might be different.

This code can be used to find the correct settings of most equalizer pre-amps.

(Example: “RIAA 500R-13.7” means for the RIAA curve that bass must be boosted below 500 Hz, but no more than + 20 dB and that treble must be cut at 10,000 Hz to – 13.7 dB)

The geographic region and the time period are added to allow a qualified guess for the required EQ should a record label not be listed. In general, American recording curves were more deliberate in cutting bass and boosting treble. The British and Europeans tended to apply only the necessary minimum of bass attenuation and often no treble boost at all, that means they had a “flat” treble curve. Early pre-emphasis curves were simply built and rather soft. In the late years (after 1945) curves became highly sophisticated, with the third turnover frequency f1 added to manage the bass shelving and also with gain ranging from -20 dB to + 20dB.

Table 1: Playback Equalization Curves

Region

Timeperiod

Curve Name, alias names

Time constants

Turnover frequencies

Bass shelf

Bass boost

Treble cut

Citations

Table 1 is mostly based on Tab. 2a of: Heinz O. Graumann, Schallplatten-Schneidkennlinien und ihre Entzerrung, in: FUNKSCHAU 1958 / Heft 15, pp 359 ff* computed frequencies to get 6 or 8 dB @ 10 kHz

[1] CCIR used by Deutsche Grammophon modified with 50 Hz bass shelving => IEC N78 [Brice]

[2] Used by British Decca and for London releases in US and UK, mostly M33. Cf.: Wolfgang Leister, The London Curve, in ARSC Journal, vol.48:2, Fall 2017, p.163

[3] Gary A. Galo, The Columbia LP Equalization Curve, ARSC conference March 2008; Gary A. Galo, Disc Recording Equalization Demystified, in ARSC Journal Fall 1996

[4] Old RCA is one of the original RCA curves for shellacs. It continued to be used for 33⅓ LPs by RCA-Victor, Brunswick, Concert Hall, Coral, Decca (Amer.) and Westminster. The turnover f3 and the time constants are computed values for an equalizer set at 800N-8.

78 rpm EQ Curve Generator

EQ Curves for Audacity can be generated from these Frequency and Roll-off values using the experimental Nyquist plugin "78 RPM EQ Curve Generator". This plugin is obtainable from the top of this and requires Audacity 1.3.13 or later. Please give feedback on this plug-in, or ask for help if you need it, by replying to that Forum topic.

Extract 78EQCurveGen.ny from the zip file downloaded from the above Forum topic.
Place 78EQCurveGen.ny in the "Plug-Ins" folder inside the Audacity installation folder, then launch or restart Audacity. For more help installing the .ny file to the correct location, click here.
Click . You can find help inside the plugin by choosing one of the Help options in "Select Function or Help".

EQ Curves Library

Here you can find some useful EQ curves for download to Audacity for use in Effect -> EQ and Filters -> Graphic EQ and Filter Curve EQ. (Curves are in TXT file format, as required in current Audacity and compatible down to version 2.4.0).

78 rpm shellacs

EQ Curve file

Description

pre-RIAA 33⅓ LPs

EQ curve files

Description

Individual fine tuning

In some cases it will be not enough to apply the correct EQ to get the desired result. According to the condition of the record and to personal listening preferences you might consider one of the following methods:

The 250-or-300-Hz dilemma: To use “European250” or “Blumlein300” EQ seems to be not a question of right or wrong but of personal listening preferences. In general, Continental European authors prefer 250 Hz (derived from the original Western Electric recommendation) for Decca (Brit. and Europ.), Brunswick, Cetra, Columbia (Brit.), EMI-HMV and Parlophone. Englishman Copeland favors 300 Hz (derived from the recording characteristic of the Blumlein cutter) for British EMI, His Master’s Voice and Columbia and also for Odeon. Copeland puts it like this: “[…] but when I’m not sure I use 300 Hz.” [Copeland, Manual, p 129]. Being derived from the previous 250-or-300-Hz curve the same applies to “DECCA78” which was used from 1944 for shellacs with the ffrr system. So you should also feel free to decide the 250-or-300-Hz question according to your own listening preference: 300 Hz will give an extra amplification of bass in play back (ca 1.5 dB at 50 Hz).
To remove low frequency noise Robinson (MidiMagic) recommends a low cut filter at 100 Hz with just 6 dB/octave. (Especially for many acoustic recordings which have only noise below 150 Hz or for the “long-playing” shellacs of RCA Victor of 1931/32). This filter will do exactly the same as the “C”-type bass shelf of Columbia LP curve.

Remarks for Analog Purists

Hiss and high frequency scratch due to old worn records: When digitizing such recordings Audacity’s will do a good job to improve the sound once and for all. Those who prefer entirely analog replay with an adjustable pre-amplifier will have the opportunity to improve the sound every time they replay. They can cut / attenuate the frequency range most affected by the noise. A higher value for roll-off at 10 kHz than the “correct” EQ will usually give a better result than a simple treble filter – but: at the expense of the high frequencies of the audio itself.
Dull, lifeless sound: If you improve poor bass on discs of any speed by choosing a higher bass turnover frequency than the “correct” EQ, there will be the welcome side effect of moderately amplifying midrange frequencies. This will bring life to the core octaves of a piece of music by improving instrument and vocal characteristics.

Acoustic recordings and Broadcast Transcription Discs

Acoustic recordings (before 1926) are beyond the scope of this page. In these pioneer years speeds varied from 70 to 90 rpm, groove modulation could be lateral, vertical or diagonal and some records were even cut outward with the audio starting at the center. A special turntable and a range of styli / needles are needed to replay.

All acoustics were recorded without any pre-equalization, simply because a modification of the audio was impossible before electric microphones and amplifiers came into use. Nonetheless there are conflicting opinions as to the result:

According to various authors the acoustical recording process had an "inherent mechanical equalization", which results – within the limited frequency range of approx. 150 to 4000 Hz – in a constant velocity characteristic one would only expect from a magnetic cutter. With a magnetic cartridge this would command to be replayed “flat”. Please note that the acoustical recording characteristic is not equalized at all.

According to Robinson an acoustical recording must have a constant-amplitude characteristic which will be correctly reproduced by playing back with a gramophone needle or a piezoelectric crystal pickup. A magnetic cartridge will – by its constant velocity characteristic – double the amplitude whenever the frequency doubles. To compensate for the magnetic pickup MidiMagic recommends an “800N-16” EQ curve, which comes close to the theoretical characteristic of a constant velocity device. Some more information is here on and on . Please note that the acoustical recording characteristic is not equalized at all.

Kolkowski’s results of a of an acoustic recording session show that bass needs heavy equalization if the losses in the recording horn (here below 400 Hz) shall be compensated for. Parametric equalization may be used at resonant frequencies. Treble should be amplified (!) to compensate for the high-frequency roll-off of the recorder. Due to the individual properties (defects) of recording horns and recorders there will be no “general characteristic” of acoustics and therefore no general EQ.

gives useful information on vintage labels as Edison, Berliner, Pathé or Zonophone and playback EQ recommendations.

Scientific help regarding valuable historic recordings can be found at (International Association of Sound and Audiovisual Archives) and at (Association for Recorded Sound Collections).

Broadcast Transcription Discs are not in the focus of this page either. Those were recordable lacquer discs, mostly 16 inch in diameter, played at 33⅓ or 78 rpm. They were professionally used by radio broadcasters. Some more information is in .

In America many of these discs were recorded under the standard of the National Association of Broadcasters (NAB) of 1942. The very same recording curve had been used by NBC under the name “Orthacoustic” since mid 1930s. This “NAB Transcription (1942)” playback EQ setting can be downloaded here.

The British Broadcasting Corporation (BBC) used a rather exotic curve as a house standard. The version in use after 1949 has been reconstructed from Longford-Smith’s publication of 1952 as an Audacity EQ setting “BBC Transcription (1949)” and can be downloaded here.

78 rpm shellac labels and their EQ

This page is about electrical recordings since 1925 on 78 rpm discs made of shellac! The invention of the Electrical Recording System by Bell Laboratories / Western Electric which was licensed to industry leaders Columbia Records and Victor set some de-facto standards: speed is always 78 rpm, cut is always lateral (same as later mono LPs) and the groove type is always Normal Groove (also named coarse groove). Therefore shellacs are sometimes referred to as N78 (which stands for normal groove discs, played at 78 rpm).

A turntable capable of 78 rpm will be useful. You will need a MONO stylus with 2.5 mil (64 μm), for early electricals possibly one with 3 mil (76 μm) and this Audacity Wiki!

It is assumed that you replay your discs “flat” (without any analog de-equalization) and apply the necessary EQ after digitizing with Audacity . If it is necessary to play the record through a system that applies modern RIAA equalization, select the "RIAA" curve in Audacity's Filter Curve EQ effect and use the Invert button to invert and thus remove the incorrect RIAA equalization before applying the appropriate equalization to the recording.

In case sources did not agree on one EQ curve, their different opinions are listed and you will have to trust your ears.

Table 2: 78 rpm Shellac Labels and Their EQ

Remarks

[Bc] ... Due to changes in the setup of the Blumlein cutter the characteristic of the recordings could vary between 180 Hz-FLAT and 500 Hz-FLAT, resulting in +/- 4 dB at 50 Hz. Copeland suggests 300 Hz as an average value.
[CAP]... used its own “Capitol curve” 400N-12.7. Play back with “AES” 400N-12.3!

Early 33⅓ LP labels and their EQ

After the launch of the “long-playing record 33⅓ rpm” by Columbia in 1948 (which used vinyl discs and a narrower groove width – microgroove records or M33) record producers experimented a lot to fully exploit the potential of the new medium. Bass shelving came into use to limit the necessary bass boost in playback and – as a consequence of the extended frequency range – necessary gain reached values as high as +/- 20 dB. So recording characteristics varied considerably!

The “poor sound quality” of some early LPs is nowadays considered to be mostly a result of the wrong EQ in playback.

Standardization was reached with the “New Orthophonic” curve of RCA which was to become the world standard by the name of RIAA. In America most labels switched to RIAA around 1955 – Europe followed by 1962.

How to identify RIAA recordings:

“ORTHOphonic” or “New Orthophonic” (after August 1952), “NARTB” or “New NARTB” (after June 1953) or “New AES” (after 1954) indicate RIAA equalization.
The following labels should have used only RIAA all the time: Argo, Bethlehem, Classic Editions, Chess, Clef, Composer Recordings, McIntosh, Montilla, New Jazz, Norgram, Prestige, Romany, Roulette, Savoy, Vocalion and Walden [High Fidelity Magazine, MidiMagic].

In case that sources did not agree on one EQ curve, their different opinions are listed and you will have to trust your ears.

Table 3: Early 33⅓ LP Labels And Their EQ

Remarks

[1]... This EQ can be traced back to Langford-Smith (1952), who vaguely describes a “London LP curve (Jan. 1951)”. Powell reads this as 300Hz (wrong!) and -14 dB (correct!). Copeland judges evidence “to be very defective.” No evidence of usage.
[CAP]... used its own “Capitol curve” 400N-12.7. Play back with “AES” 400N-12.3!

Sources, links and reading references

Sources of tables 2 and 3:

AT ... AudacityTeam own research
ES ... and the updated EQ list from the of their re-equalizer pre-amp in which they also have very useful tips how to identify LPs by their matrix number (British London / Decca, American Decca, American Columbia, RCA Victor). Please mind that they quote settings of their device which have to be translated back into parameters of EQ curves.

Recommended analog reference

James R. Powell, Jr. and Randall G. Stehle, "Playback Equalizer Settings for 78 RPM Recordings", Third Edition, Gramophone Adventures, Portage, MI, 1993, 2002, 2007. A compilation of Powell’s subjective recommendations of Owl 1 settings for approx. 1800 discs / 400 labels (mostly American and Jazz). Reprint of 3rd edition available at .
Fritz Langford-Smith, Radiotron Designer's Handbook, Wireless Press, Sydney, Fourth edition, 1952.
James Moir, High Quality Sound Reproduction, Chapman & Hall Ltd., London, 1958

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