MuseFX can be downloaded from the "Effects" tab in the Muse Hub
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 * Compressor 4 : Dynamics processor * Dual VCF : non-linear filters * Enhancer : preset filter eq * Equalizer 4: Digital remasterd Anolog dynamic eq * Flowtones : synthesizer * GonioMeter : Stereo Image Analyzer * MBC : Multiband Compression and saturation * Morphit : Headphones correction and personalization * ReelBus 4 : Magnetic tape recording, echo and flanger simulator * Reverb 4 : Lush reverberation and shimmer * Sibalance 4 : Spectral de-esser and harshness remover * Spectogram : Insightful time and frequency visualization * VoicePitcher 4 : Vocal doubling, pitch shifting and freezing effectDetails
See the pack for installation instructions. Has modes optimized for male/female voices and more, and can snap to major, minor and even more rare scales (eg Lydian).Details
MAutoPitch is a simple but great sounding automatic tuning and pitch correction plugin designed for vocals and other monophonic instruments. Besides making the audio more in-tune, MAutoPitch also provides creative features such as formant shift and stereo-expansion. * Advanced GUI * Compare multiple settings: A to H Switching and A to D Morphing. * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * Automatic gain compensation (AGC) * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.Details
See the pack for installation instructions. Details
See the pack for installation instructions. Details
Copyright (C) 2006-2016, Cockos Incorporated VST PlugIn Technology by Steinberg Media Technologies GmbH Information From the vendor: * Multi-tap delay, no practical limit on tap count * Up to 10 second delay per tap * Tap lengths can be in time (s/ms) or quarter notes * Feedback, LPF/HPF, resolution reduction per tap * Stereo width per tap * Volume/pan per tapDetails
**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. * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.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. * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * Automatic gain compensation (AGC) * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.Details
* 12 early reflections variations * 5 reverb tail variations * 3-band equalizer for the wet signal * Sample Rate: from 44100 to 192000 Hz. * Lots of presetsDetails
Information from the developer website: * Modulated vintage reverb sound. * Very diffuse sound. * Fast build up time, also with long reverb sounds. * Works on almost every audio material. * Easy to use. * Only stereo channels supported.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.Details
Information from the developer website: * Plate, room, and hall reverbs * 5 reverb modes * Reverb mode editor * Stereo processing * 64-bit floating point processing * Preset manager * Undo/redo history * A/B comparisons * Contextual hint messages * All sample rates support * Zero processing latency * User interface color schemes * Resizable user interface * Retina and HighDPI supportDetails
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.Details
**MBitFun** is a serious tool for extreme distortion lovers. It converts the audio into limited fixed-point precision form, from a 1 single bit up to 16 bits per sample, and lets you access each bit, applying several operations. * Advanced GUI * Compare multiple settings: A to H Switching and A to D Morphing. * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * Automatic gain compensation (AGC) * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.Details
**MFreqShifter** is an extremely versatile frequency shifter. Unlike pitch-shifters it doesn't keep harmonic relationships and can provide everything from mild stereo expansion to complete sonic destruction. * Advanced GUI * Compare multiple settings: A to H Switching and A to D Morphing. * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * Automatic gain compensation (AGC) * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.Details
MSaturator performs smooth saturation in a definitive tube-like way improving the clarity of the resulting sound. Its excellent sound quality makes it flexible enough to double as a distortion module for guitars and other instruments. * Advanced GUI * Compare multiple settings: A to H Switching and A to D Morphing. * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * Automatic gain compensation (AGC) * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.Details
MWaveFolder is an analog-inspired distortion module with a unique character ranging from mild harmonic enhancement to complete sound destruction. * Advanced GUI * Compare multiple settings: A to H Switching and A to D Morphing. * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * Automatic gain compensation (AGC) * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.Details
MWaveShaper goes beyond a traditional wave-shaping plugin. Unlike the conventional approach of providing a few predefined patterns, MWaveShaper lets you construct your own shape creating a much greater range and control over your sound. * Advanced GUI * Compare multiple settings: A to H Switching and A to D Morphing. * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * Automatic gain compensation (AGC) * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.Details
See the pack for installation instructions. Details
See the pack for installation instructions. Details
Copyright (C) 2006-2016, Cockos Incorporated VST PlugIn Technology by Steinberg Media Technologies GmbH Information from the vendor: * FFT based dynamics processor * Supports FFT sizes of 128-32768 * Useful in/out frequency response display * Supports defining response curves both using any number of points, or freehand mouse * EQ - can be used as a linear phase mastering EQ, or as a super-effective surgical EQ * Compressor - can compress at a fixed ratio with a per-band threshold * Gate - can gate with per-band threshold * Subtract - can build noise profiles and subtract noise from the signalDetails
MCompressor provides refined compression. It features an adjustable compression shape, giving you the power to create dynamic sound effects, and a range of up-sampling options for a crystal clear sound. * Advanced GUI * Compare multiple settings: A to H Switching and A to D Morphing. * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * M/S, single channel, up to 8 channelds surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.Details
Copyright (C) 2006-2016, Cockos Incorporated VST PlugIn Technology by Steinberg Media Technologies GmbH Information From the vendor: * Unlimited band compressor * Great metering per band * Fantastic sounding filters * Solo current band mode * Tons of controls per band (ratio, threshold, knee, attack, release, makeup, program dependent release, feedback detector, RMS size) * Adjusting bands in graph is easy (modifiers to change ratio, gain, etc)Details
© 2011-2022 AudioThing Ltd. Homepage:Details
Description from the vendor * Different modulation types: LP 12dB, BP 12dB, HP 12dB, Pan, Volume * Super clean SVF filter with diode clipper in the feedback path. Capable of self oscillation. * Easy to use spline editor. * Stereo offset for modulation when in filter mode. * Host sync with different sync options (normal, dotted...) * Trigger button sets the modulation position to the start without lose host sync. Can be automated. * Legacy Mode: Just for backward compatibility. Please don't use if you create something new. * File based presets. * MIDI NoteOn trigger option.Details
Description from the vendor * Generate Filter Grooves in real-time * Syncs to Host tempo * Wet/Dry mix control * MIDI Controllable * Infinity Mode * Quick Load presets * Instant/Tap Bypass for live performance * Quantization settings * Motion selector * Customizable Sequence RangeDetails
* Advanced GUI * Synchronization to host tempo * Adjustable oscillator shape technology * Automatic gain compensation (AGC) * Safety limiter * Modulators * Multiparameters * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for download page.Details
* Advanced GUI * Sinc interpolation * Adjustable up-sampling 1x - 1024x * Adjustable oscillator shape technology * Automatic gain compensation (AGC) * Safety limiter * Modulators * Multiparameters * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for download page.Details
See the pack for installation instructions. Details
See the pack for installation instructions.Details
Copyright (C) 2006-2016, Cockos Incorporated VST PlugIn Technology by Steinberg Media Technologies GmbH Information From the vendor: * Unlimited band IIR based equalizer * Support for any number of many types of filters (shelfs, bands, LPF, HPF, notch, bandpass, allpass) * Frequency response and phase response display * Display of approximate note+octave for frequencies * Per-band bypass control * Full-view of graph optional for precise editing * Mouse modifiers/mousewheel usable for editing bandwidth of points in graphDetails
Information from the vendor * 16-band graphic equalizing * Freehand drawing mode * Linear-phase equalizing * +/- 12 dB gain range per band * Stereo and multi-channel processing * Internal channel routing * Channel grouping * Mid/side processing * 64-bit floating point processing * Preset manager * Undo/redo history * A/B comparisonsDetails
Information from the vendor * 32-band graphic equalizer * Selectable -12 to +12 dB or -6 to +6 dB gain * Integrated low-cut filter * Adjustable input gainDetails
Copyright © 2012–2022 Red Rock Sound Information from the vendor * 10 bands of proprietary equalization. * Familiar graphics operation on one octave centers. * 12 dB of boost/cut per band. * Proportional Q narrows filter Q at extremes. * IN/OUT switch — allows the user to bypass the Graphic Eq for before/after comparisons.Details
**MConvolutionEZ** is an extremely easy-to-use and powerful 6-band equalizer. It also provides an advanced visualization including a spectrum analyzer and sonogram. * Advanced GUI * Compare multiple settings: A to H Switching and A to D Morphing * 9 filter types for each band with quick adjustment * Powerful spectrum analyzer and sonogram * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * M/S, single channel, up to 8 channelds surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.Details
### Vendor description #### 1950 Meets 2019 The W2395c is a classic Baxandall EQ combined with a powerful and beautiful sounding mid band. As a solitary or complementary equalizer with that special something, its particular tone and slightly interactive bands let your tracks stand out big time. In close cooperation with its creator, analog mastermind [Roger Schult](https://www.rogerschult.com/), we are delighted to offer this amazing plugin completely for free. #### Control and Shape The W2395c excels in controlling the midrange within the context of broad tonal shaping of bass and treble. The high shelf is great for carving out percussive sounds as well as opening or darkening instruments and voices. The partnering low band provides a precise and solid grip on the sub frequencies, while the mids manage the balancing act between natural and beefy with stunning ease. #### Straight yet Flexible Operating the W2395c is as straight forward as it gets, and still the few controls provide great flexibility. The low band can be switched from 80 to 110 Hz, the highs from 2 to 5 kHz, and the semi-parametric mids offer 3 different Q factors to choose from. The drive control is a plugin-exclusive extra for adding some nice grit to its supernaturally natural tone character. #### Freely yours With its super intuitive controls and musically interacting curves, the W2395c is a special and unique gem in the world of equalizers. It enhances anything from very dynamic sources like vocals or drums to harsh cymbals or dull sounding guitars and synths – and it does an excellent job on the mix buss as well. A truly exciting EQ experience is waiting for you… right now, all free.Details
See the pack for installation instructions. Details
See the pack for installation instructions. Details
© 2022 VALHALLA DSP, LLC. COMPATIBILITY: * WINDOWS: WINDOWS 7/8/10 * PLUGIN FORMATS: 64-BIT VST2.4/VST3/AAX * MAC: OSX 10.8/10.9/10.10/10.11, MACOS 10.12/10.13/10.14/10.15, MACOS 11 BIG SUR * PLUGIN FORMATS: 64-BIT VST2.4/VST3/AAX/AU \\ Information from the vendor:Details
MRingModulator performs classic ring-modulation effects using one or two oscillators. With a clean interface that gives easy access to more advanced controls like our adjustable phase difference and shape features, including editable custom waveforms and harmonics. * Advanced GUI * Compare multiple settings: A to H Switching and A to D Morphing. * Unique visualisation engine with classic meters and time graphs * MIDI controllers with MIDI learn * Automatic gain compensation (AGC) * M/S, single channel, up to 8 channels surround and up to 64 channels ambisonics processing * Extremely fast, optimized for newest AVX2 and AVX512 capable processors * Supports VST, VST3, AU and AAX interfaces on Windows and macOS See the [pack ](/realtime-effects/plugin-suites#mfreefxbundle)for installation instructions.Details
See the pack for installation instructions. A noise gate keeps noise in quiet parts out while not affecting the audio you want at all. Details
See the pack for installation instructions. Has presets for female and male voices, harshness, sibilance, steel guitar and strings rosin.Details
Copyright (C) 2006-2016, Cockos Incorporated VST PlugIn Technology by Steinberg Media Technologies GmbH Information from the vendor: * FFT based dynamics processor * Supports FFT sizes of 128-32768 * Useful in/out frequency response display * Supports defining response curves both using any number of points, or freehand mouse * EQ - can be used as a linear phase mastering EQ, or as a super-effective surgical EQ * Compressor - can compress at a fixed ratio with a per-band threshold * Gate - can gate with per-band threshold * Subtract - can build noise profiles and subtract noise from the signalDetails
Copyright (C) 2006-2016, Cockos Incorporated VST PlugIn Technology by Steinberg Media Technologies GmbH Information From the vendor: * Ultra-configurable gate * Sidechain filters, sidechain input * Lookahead for pre-open * Hold control * Hysteresis control * Variable RMS size * Can send MIDI events on gate open/close * Wet/dry mix, noise mix (can add noise when gate is open)Details
Copyright © 2022 NVIDIA Corporation Requires NVIDIA GeForce RTX 2060, Quadro RTX 3000, TITAN RTX or higher Full setup guide:Details
Copyright © 2022 NVIDIA Corporation Requires NVIDIA GeForce RTX 2060, Quadro RTX 3000, TITAN RTX or higher Full setup guide:Details
Author: David R.Sky Generates a nasal-sounding tone composed of the base frequency plus n-1 harmonics. If for example you choose a frequency of 100 Hz with n=4 harmonics, this plugin will generate a tone comprised of 100, 200, 300 and 400 Hz, of equal amplitude. (The more harmonics, the more nasal and high-pitched the tone sounds.) Parameters: 1. **Frequency or MIDI number:** \[0=frequency, 1=MIDI (default)] 2. **Frequency:** \[20 - 5000 Hz, default 110] 3. **MIDI note number:** \[16 - 127, default 45] 4. **Number of harmonics:** \[1 - 60, default 12] 5. **Buzz tone duration:** \[0.1 - 120 seconds, default 5.0] 6. **Volume:** \[1 - 100 percent, default 95]Details
Authors: David R.Sky, Dominic Mazzoni, Roger Dannenberg, W. Borgert Touch Tones (also known as DTMF or Dual Tone Multi Frequency Tones) are the tones made by key pads on telephones. Each tone is comprised of two separate tones at different pitch, hence "dual tone". Parameters: 1. **Number of DTMF Tones:** \[1 - 120, default 20] 2. **Include military tones A-D:** \[0 = no (default), 1 = yes] 3. **Option to include silent intervals:** \[0 = no (default), 1 = yes] 4. **Volume:** \[0.001 - 1.000, default 0.3] 5. **Tone length** \[0.001 - 1.000 seconds, default 0.1] 6. **High to low tone ratio:** (or twist) \[0 - 4 dB, default 0] - "Twist" is the volume ratio between the higher-pitched and lower-pitched tones in any given tone. So a twist value of 0 dB means the higher-pitched tone is no louder than the lower-pitched tone. A twist value of 4 dB means the higher-pitched tone is 4 dB louder than the lower-pitched tone. 7. **Post silence duration:** \[0 - 1 seconds, default 0.1]Details
Authors: David R.Sky, Dominic Mazzoni, Roger Dannenberg, W. Borgert Touch Tones (also known as DTMF or Dual Tone Multi Frequency Tones) are the tones made by key pads on telephones. Each tone is comprised of two separate tones at different pitch, hence "dual tone". Type in your telephone number, or an 'alphabetized' number such as "1800audacity". Includes the US Military's A, B, C and D tones to the right of the regular number keypad. Parameters: 1. **Tone string:** \[1800audacity (default)] 2. **Tone duration:** \[1 - 1000 milliseconds, default 100] 3. **Silence duration after tone:** \[0 - 1000 milliseconds, default 100] 4. **Twist:** \[0 - 4 dB, default 0] - "Twist" is the volume ratio between the higher-pitched and lower-pitched tones in any given tone. So a twist value of 0 dB means the higher-pitched tone is no louder than the lower-pitched tone. A twist value of 4 dB means the higher-pitched tone is 4 dB louder than the lower-pitched tone. 5. **Volume:** \[1 - 100 percent, default 80]Details
Author: Steve Daulton HQ-Tone is a high quality (no alias) tone generator. Like the *Square (no alias)* option in Audacity's built-in [Tone](https://manual.audacityteam.org/man/tone.html) generator, the waveforms generated by this plugin produce bandwidth limited waveforms to avoid [aliasing](https://en.wikipedia.org/wiki/Aliasing) distortion. This plugin is typically faster than Audacity's built-in "Square - no alias" generator, particularly for low frequencies, and provides more waveform choices: Parameters: 1. **Waveform:** \[Choice: Sine, Square, Triangle, Sawtooth, Inverse sawtooth (default: Square] 2. **Frequency (Hz):** \[20 - 10000 Hz, default 440 Hz] - the frequency of the generated waveform. 3. **Amplitude (0 to 1):** \[0 to 1, default 0.8] - The amplitude of the generated waveform. Note that due to technical limitations, the actual waveform may be very slightly different from the requested amplitude. However, the amplitude can be expected to be very close to the specified level (within about 0.001 dB). 4. **Duration:** \[must be greater than zero] - Time units may be selected by clicking the downward pointing arrow on the right side of the control. 5. **Phase (-180 to 180):** \[-180 to +180 degrees] - The starting phase of the waveform. Rising across zero is taken to mean a phase of 0 degrees for all waveforms. 6. **Smoothing:** \[Choice: Yes / No (default: Yes)] - Waveforms that have near instantaneous rise or fall times have noticeable ripple. When this option is enabled, additional damping is applied to smooth out the ripples, producing a *cleaner* looking waveform, at the expense of slightly reduced high frequency harmonics.Details
Author: Steven Jones. Generates a modulated pulse tone. Parameters: 1. **MIDI key:** \[0 - 127, default 60] 2. **Cents:** \[0 - 99 cents, default 0] - Detune amount 3. **Duration:** \[1 - 30000 milliseconds, default 10000] 4. **Mod Rate:** \[1 - 100, default 1] - number of modulation cycles 5. **Mod Depth:** \[-100 - +100 percent, default 90] 6. **Mod Wave:** \[0 = tri (default), 1 = up sawtooth, 2 = down sawtooth] - waveform of tone 7. **Width:** \[0 - 100 percent, default 0] - fixed pulse width 8. **Amplitude:** \[0 - 100 percent, default 100] If the sum of the fixed width and the instantaneous modulation amount is outside the interval \[0 - 99], the output will go to full off or full on.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: 1. **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. 2. **Number of Harmonics:** \[1 - 32, default 8] - that is, the number of partials for each note generated 3. **Duration:** \[1 - 30 seconds, default 10] 4. **Band Width:** \[1 - 1000 Hz, default 2] - depending on the width of this noise band, the result can sound very noisy or distinctly tonal with a heavy chorusing effect. 5. **Odd Harmonics Only:** \[0 = all harmonics (default), 1 = odd-numbered harmonics only]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: 1. **Center Frequency (Hz):** \[10 to 10000 Hz, default 440] 2. **Bandwidth (Hz):** \[1 to 10000 Hz, default 50] 3. **Amplitude (0 - 1):** \[0 to 1, default 0.8] 4. **Duration (minutes):** \[0 to 20 minutes, default 0] 5. **Duration (seconds):** \[0 to 60 seconds, default 30] 6. **Stereo Output:** \[No / Yes, default No] To generate true stereo noise (left and right channels different), a stereo track must be selected and the Stereo Output control must be set to "Yes". As with all Nyquist generator plugins, if a track is not selected, Audacity will create a mono track and attempt to place the generated sound into that track. If Stereo Output is set to yes and a stereo track is not selected the plugin will return the error message: ***"Nyquist returned too many audio channels."***Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) A sine tone of one constant frequency is generated in the left channel of a stereo track, and a series of changing tones of slightly different frequencies are generated in the right. The differences between the left- and right-channel frequencies are termed "beat frequencies". In addition, a stereo "surf" noise is generated based on [pink noise](http://en.wikipedia.org/wiki/Pink_noise). This is a lower-frequency "rushing" sound compared with "hissing" white noise. Parameters: 1. **Left channel tone frequency:** \[50 - 1000 Hz, default 100]. 2. **Beat frequency \[Hz], duration \[minutes], time to change to next beat frequency \[minutes]:** There are six of these edit fields in which you may enter up to three indicated values, separated by a space. The first of these edit fields has default values of 17.5 0.25 0.25 and must contain some non-zero value for duration. If you enter only a single value into any of the subsequent fields, the duration of that beat frequency will be zero. If you leave any of these edit fields blank they will be ignored. In the sixth field you may enter a final beat frequency and duration of that frequency. 3. **Adjust total time:** \[1 - 60 minutes, default = 0 (no adjustment)]. 4. **Fade-in and fade-out times:** \[0 - 120 seconds, default 10] 5. **Stereo surf frequency:** \[0 - 2 Hz, default 0.1] - If this setting is above zero, the surf sound will be panned back and forth somewhere between the left and right audio channels at the specified frequency, how far depending on the sixth variable: 6. **Stereo surf spread:** \[0 - 100 percent, default 80] - The larger this number, the further the surf sound will move away from the center pan position (0% results in the surf sound remaining in the center). 7. **Tone to surf volume ratio:** \[0 - 100 percent, default 70] - adjusts the relative volume of the tones and surf sound. 0 = no tone (only surf) and 100 = no surf (only tone).Details
Author: [David R.Sky](https://audionyq.com/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](https://audionyq.com/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.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Emulates a Korg MS-10 synthesizer patch. Parameters: 1. **Audio type:** \[0=fire 1=explosion, default=1] 2. **Sound duration:** \[0.1 to 60 seconds, default=3] 3. **Attack time:** \[0 to 500 milliseconds, default=50] 4. **Explosion decay time:** \[0 to 500 milliseconds, default=500] 5. **Decay down to this level:** \[1 to 100 %, default=30] 6. **Cutoff frequency:** \[100 to 10000 Hz, default=3800] 7. **Filter quality:** \[Q: 1 to 20, default=10] 8. **Bass boost frequency:** \[10 to 1000 Hz, default=300] 9. **Bass boost:** \[0 to 60 dB, default=30] 10. **Clipping amount:** \[0 to 99 %, default=55]Details
Author: Steven Jones. KLSTRBAS (for "cluster bass") generates dense sounds by combining several waveforms with a fixed frequency ratio between them. Early Roland drum machines created cymbal sound in part by combining multiple square waves with non-integral frequency ratios. The combined signal was then high-pass filtered to produce a very dense cluster of high frequency harmonics. The genesis of KLSTRBAS was a failed attempt to create cymbal sounds using this technique. Parameters: 1. **MIDI key:** \[0 - 127, default 45] 2. **Decay:** \[0 - 30 whole seconds, default 2] 3. **Fractional Decay:** \[0 - 99 hundredths of a second, default 0] - synth kick drum sounds can be produced by setting Decay time to zero and fractional decay to a low value. 4. **Density:** \[1 - 6, default 4] - Sets the number of component waveforms, defined as four times the density value. Higher densities produce a deeper flange effect but can also cause the sound to go out of tune. 5. **Detune:** \[0 - 99, default 0] {these two parameters affect the relative 6. **Flange:** \[0 - 4, default 2] frequencies of the component waveforms} 7. **Wave table:** \[0=sine 1=tri 2=sqr 3=saw (default}] - type of component waveform. These are not band limited so [aliasing](http://en.wikipedia.org/wiki/Aliasing) may result if either MIDI key or generated frequencies are too high. *Technical note:* The frequency of each component is determined by the MIDI key number and the detune and flange parameters. Specifically the nth component has a frequency of:\ p \* (1 + d/100 + g)^n where:\ p is the fundamental frequency determined by the key number,\ d is the detune amount 0 <= d <= 99, and\ g is derived by the flange parameter (g = 1/(10^(4-f)) for flange value f)Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) and Steve Daulton. Pluck generates a synthesized pluck tone with abrupt or gradual fade-out. The sound is the same as the [Pluck effect](https://manual.audacityteam.org/o/man/pluck.html) that is shipped with Audacity, except that the pitch of the note is determined by entering a frequency in Hz (cycles per second) rather than a MIDI note. Parameters: 1. **Pluck frequency (Hz):** \[10 to 10000, default 261.626] The default frequency is equivalent to MIDI note 60. 2. **Fade-out type** \[abrupt or gradual, default "abrupt"] Determines how rapidly the pluck sound decays. 3. **Duration \[seconds]:** Specifies the length of the pluck sound. The default is 1.0 second. A table showing the relationship between note names, MIDI note numbers, and frequency (A440 standard tuning) is available [HERE](http://subsynth.sourceforge.net/midinote2freq.html).Details
Author: Steven Jones. Simulates a realistic bell tone based on the pioneering work of Jean Claude Risset. This plugin is an adaptation of a demonstration lisp file by Pedro Jose Morales contained in the standard Nyquist distribution. Parameters: 1. **MIDI key:** \[0 - 127, default 72] 2. **Decay:** \[0 - 30 seconds, default 10] 3. **Fractional Decay:** \[0 - 99 hundredths of a second, default 0]Details
Author: Steven Jones. Generates random sine waves. The generated tones have random frequencies, attack and decay times. Parameters: 1. **Duration:** \[1 - 30 seconds, default 20] 2. **Density:** \[1 - 100 generated tones, default 60] 3. **Floor:** \[20 - 1000 Hz, default 300] - Lowest frequency of tones 4. **Ceiling:** \[20 - 1000 Hz, default 600] - Highest frequency of tonesDetails
Author: Steven Jones. Algorithmic generator sequencer number 1. Note also the [Audio Selection Sequencer 2](https://wiki.audacityteam.org/wiki/Nyquist_Effect_Plugins#Audio_Selection_Sequencer_2) on the [Effect Plugins page](https://wiki.audacityteam.org/wiki/Nyquist_Effect_Plugins) which sequences *pre-existing* audio samples. The sq1 sequencer generates complex sequences of tones by using the sum of three square-wave low frequency oscillators to frequency-modulate two oscillators. The oscillators output one of four waves (sine, tri, square and saw) and may be frequency adjusted relative to each other. The wave tables are not band-limited so [aliasing](http://en.wikipedia.org/wiki/Aliasing) will result for sufficiently high frequencies. There is also an overall three-stage amplitude envelope. Parameters: 1. **Center:** \[0 - 10000 Hz, default 440] - the unmodulated carrier frequency 2. **Detune:** \[0.25 - 4.00, default 1.01] - the frequency of oscillator 2 relative to oscillator 1 3. **Wave:** \[0=sine, 1=tri (default), 2=square, 3=saw]. Wave shape for both oscillators 4. **Attack:** \[0 - 10 seconds, default 1] 5. **Sustain:** \[0 - 10 seconds, default 1] 6. **Decay:** \[0 - 10 seconds, default 1] 7. **f1:** Frequency of LFO 1 8. **a1:** Amplitude of LFO 1 - LFO amplitudes are calibrated in Hz indicating the corresponding frequency shift in the audio oscillators. 9. **f2:** Frequency of LFO 2 10. **a2:** Amplitude of LFO 2 11. **f3:** Frequency of LFO 3 12. **a3:** Amplitude of LFO 3 Note that the three LFOs are interchangeable.Details
Authors: Steve Daulton, Edgar Franke, Steven Jones and [David R.Sky](https://audionyq.com/david_r_sky) 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: ``` (setq mysound (noise 10)) (lp mysound (pwl 0 5000 10 100 10)) ``` 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"*.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) A software tuning fork, with a table giving C notes and their equivalent MIDI note numbers near the top of the screen. **Parameters:** 1. **Tone duration:** \[0 - 300 seconds, default = 120] 2. **Constant or fade out:** \[0=constant volume or 1=fade out, default = constant] 3. **MIDI or frequency:** \[0=MIDI 1=frequency, default = MIDI] - choose whether to generate tone with reference to a MIDI note number or a frequency. **Note:** Middle C = MIDI note 60, A above Middle C (440 Hz) = MIDI note 69. 4. **MIDI note:** \[16 - 127, default = 69] - You can use non-integer values here (such as 60.75) 5. **Frequency:** \[20 - 20000 Hz, default = 440]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) If you have a stereo track and want to amplify or attenuate one channel only without using the mouse, this plugin will do it. Parameters: 1. **Channel** 0=left channel, 1=right channel (default 0). 2. **Volume** \[dB]: amplify or attenuate the channel (default 0 dB, no change in volume).Details
Author: Jvo Studer. Frequency-selective filter to crossfeed (mix) bass frequencies to center (mono). Requires stereo tracks. Simulates the "Elliptic EQ" filter found on vinyl mastering consoles by means of a first order high-pass filter in the side-channel (L-R difference). This is useful to bring the bass drum (or beat in electronic music) to the center. If bass frequencies are partially out of phase there will be some bass loss which can be compensated for with the Bass Boost shelf filter (typically +0.5 dB to +2 dB will be sufficient). Parameters: 1. **Crossover Frequency:** \[10 - 500 Hz, default = 150] 2. **Bass Feed Proportion:** \[20 - 100 %, default 95] 3. **Bass Boost:** \[0 - 6 dB, default 0.5]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Removes center-panned content in stereo tracks by inverting and making mono. Can be used to mitigate vocals in music tracks if the vocals are panned to center. Optionally you can choose a band of frequencies to invert, rather than the whole channel. This may be less destructive of the content panned away from center. The resulting audio retains two channels, but sounds mono because both channels are panned to center. Parameters: 1. **Invert band or channel:** \[0="band", 1="channel", default = channel] 2. **Remove frequencies above...:** \[20 - 20000 Hz, default 500] 3. **Remove frequencies below\...:** \[20 - 20000 Hz, default 2000]Details
Author: Steve Daulton. A multi-purpose tool that can perform almost any type of channel mixing task on an unsplit stereo track. Typical uses include: * Copying one channel of a stereo track to the other channel * Converting a stereo track into 2 channel mono * Stereo "widening" (or narrowing) * Swapping left and right channels of a stereo track * Vocal Removal * [Mid-Side](http://en.wikipedia.org/wiki/Microphone_practice#M.2FS_technique:_Mid.2FSide_stereophony) decoding. The plugin has 15 presets for the above and similar tasks. If the presets are not used, custom values for the mix of original left and right channel in the left and right outputs can be entered in the parameters as listed below. All values are percentages with a possible range of -100% to +100% and all default to zero value. <-- LEFT CHANNEL OUTPUT --> 1. **from original Left channel (%):** 2. **from original Right channel (%):**\ <-- RIGHT CHANNEL OUTPUT --> 3. **from original Left channel (%):** 4. **from original Right channel (%):** This [Zip archive](https://wiki.audacityteam.org/w/images/3/3d/Channel-mixer.zip) download also includes a comprehensive 'FAQ' help file. **List of Presets:** 1. Mono (Average): Creates identical left and right with 50% from the Left channel and 50% from the Right channel. 2. Mono (Both Left): Makes both channels the same as the original Left channel. 3. Mono (Both Right): Makes both channels the same as the original Right channel. 4. Extra Narrow: Almost mono but retains a little stereo separation. 5. Narrow Stereo: Reduces the stereo width (requires that the original track is true stereo). 6. Wide Stereo: Expands the stereo width (requires that the original track is true stereo). 7. Extra Wide: Same as 'Wide Stereo' but more extreme. 8. Centre to Left: Moves sounds from the middle of the stereo stage to the Left. 9. Centre to Right: Moves sounds from the middle of the stereo stage to the Right. 10. Swap Left/Right: What it says on the tin. 11. Vocal Remover (L/R invert): Removes sound from the center of the stereo mix and inverts the right channel. 12. Vocal Remover (mono): Removes sound from the center of the stereo mix. 13. Invert Left: Inverts the Left channel. 14. Invert Right: Inverts the Right channel. 15. Mid-Side Decode: A simple Mid-Side decoder.Details
Author: \[multiple contributions] Despite the name this effect does not perform an automatic "crossfade" between two tracks or adjoining audio. Instead Cross Fade In applies a "curved" fade that is different from the linear Fade in. The curve used is one that will result in equal RMS (average) volume throughout the faded section when used in conjunction with the "Cross Fade Out" effect, to perform a crossfade.Details
**Cross Fade Out** applies a "curved" fade that is different from the linear Fade Out. The curve used is one that will result in equal RMS (average) volume throughout the faded section when used in conjunction with the **Cross Fade In** effect, to perform a crossfade.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Define the length of fade-in and fade-out selections without using a mouse or cursor keys. Note Audacity has a Selection Toolbar providing a screen-reader friendly display of selection start time and duration which you could use for similar purpose. Parameters: 1. **Fade in time:** \[seconds, maximum 30] 2. **Fade out time:** \[seconds, maximum 30]Details
Authors: [David R.Sky](https://audionyq.com/david_r_sky), Dominic Mazzoni Audacity lets you pan with the keyboard instead of the mouse, but if you prefer the pan to modify the waveform immediately, this plugin will statically pan your stereo audio anywhere between left and right channels. There is only one parameter: 1. **Pan position:** \[0=left, 1=right, default is 0.5 (center-panned)]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Panning is controlled by a low frequency oscillator. Only works on unsplit stereo tracks. Pan the audio to center before use for best results. Parameters: 1. **LFO frequency:** \[Hz, 0.02 - 20, default 0.1] 2. **LFO waveform:** \[sine, triangle, saw, inverted saw, pulse] 3. **Pulse waveform duty cycle:** \[percent, default 50] 4. **LFO starting phase:** \[degrees, -180 - +180, default 0] 5. **Leftmost pan position:** \[percent, default 5] - 0%=left channel, 50%=center, 100%=right channel 6. **Rightmost pan position:** \[percent, default 95] - 0%=left channel, 50%=center, 100%=right channelDetails
Author: [David R.Sky](https://audionyq.com/david_r_sky) Randomly pans stereo audio from one side of the stereo field to the other - just like someone is playing around with the panning knob. Requires an unsplit stereo track. Parameters: 1. **Maximum random panning speed:** \[Hz, 0.01 - 10.00, default 0.2] - how fast the random panning changes occur 2. **Maximum stereo width:** \[percent, 0 - 100, default 100] - how far away from center the signal is panned. 0% gives no panning, 100% results in the signal being randomly panned between hard left and hard right pan positions.Details
Author: Steve Daulton A stereo spatializer effect. Creates an artificial stereo effect that can be useful for giving some "depth" to mono recordings. Mono tracks must be converted to a 2 channel track before using the effect. To do so, click above the Mute and Solo buttons in the Track Control Panel, choose Edit > Duplicate then click on the name of the upper track and select "Make Stereo Track" from the dropdown menu. Parameters: 1. **Select source channel:** \[Left (upper) or Right (lower)] 2. **Delay factor (%):** \[0 to 100, default 30] Higher values will produce a wider stereo effect but may sound echoey. 3. **Effect mix (%):** \[0 to 100, default 80] 0% = Dry (original signal with no effect), 100% = Wet (effect and no original signal). Higher settings will produce a more pronounced stereo effect but may leave a "hole" in the center of the stereo field. Lower values produce a more subtle effect with the original signal centered mid-stage.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Evenly pan your stereo audio, starting at one point in the stereo field and ending at another. -10 corresponds to 100% left, 0 to center and +10 to 100% right. Parameters: 1. **Start position:** from \[where -10 - +10, default -10] 2. **End position:** from \[where -10 - +10, default +10]Details
Author: Steve Daulton For repairing damage to one channel of a stereo track by overwriting the damaged region with audio from the other channel. Select the damaged audio and allow additional space for cross-fading, then apply the effect. The plugin includes an option for *Stereo Simulation* which will often make repairs to stereo recordings less noticeable. For tracks that have little or no stereo channel separation and for synthesized tones, best results will probably be achieved with Stereo Simulation disabled. Parameters: 1. **Which Channel to Repair:** \[Left Channel or Right Channel] The upper channel of a stereo pair is the Left channel. 2. **Stereo Simulation:** \[Enabled or Disabled] 3. **Cross-fade Time:** \[0% to 50%] 0% will cut directly from one channel to the other without fading. At 50% the fades will occupy the entire selection. For the default 20% fade, the selection should be at least 40% longer than the actual damage to be repaired.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) The original Stereo Butterfly plugin, the name coming from a butterfly's wings, which can be spread wide (1, full stereo), closed (0, sounding mono), or somewhere in-between. Stereo Butterfly can even mirror the left and right channels (-1... the butterfly's flipped!). And also anywhere between the extremes from -1 to 1. Parameters: 1. **Stereo width** \[width, between -1.0 and +1.0]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Second in the Stereo Butterfly series. It takes stereo audio and makes it sound like the left and right channels are switching back and forth with each other. You can define the LFO (low frequency oscillator) rate. As in Stereo Butterfly (static), -1 is stereo channels fully flipped with each other, 0 sounds like mono, and 1 is full regular stereo. The difference here is that you can define two widths, so defining how you want the stereo to be manipulated over time. For instance, from -1 to +1 means Stereo Butterfly flips the left and right channels with each other at the frequency you set. If you set the two numbers at 0 and +1, the stereo audio will change between mono-sounding and regular stereo. Set at -1 and 0, the effect will be of fully flipped stereo changing to mono-sounding. Any other numbers you choose between -1 and +1 will give intermediate effects. Parameters: 1. **LFO frequency:** \[between 0.01 and 20 Hz] 2. **Width1:** Stereo width from \[range from -1.00 to +1.00] 3. **Width2:** to Stereo width \[range from -1.00 to +1.00]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Third in the series of Stereo Butterfly plugins. As with the previous two, 0 setting sounds like mono, +1 is regular stereo, -1 is left and right channels flipped with each other. Select which value to start at and which value to finish at. The default is from 0 to 1, which creates the effect of your stereo audio starting out sounding mono, then gradually widening to full stereo as the selection progresses. Start and finish values may lie anywhere between -1 and +1. Parameters: 1. **Spread stereo from...** \[range from -1.00 to +1.00] 2. **to:** \[range from -1.00 to +1.00]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Gives the illusion of widening stereo audio. The effect produces different results depending on whether you are listening to the audio through speakers or headphones, and the distance stereo speakers are apart. The widener works by inverting both left and right channels of stereo audio, then panning those inverted signals somewhere between the center pan position and the opposite channel. Parameters: 1. **Inverted signal volume:** \[-48 dB - -6 dB, default -18 dB] 2. **Pan position:** \[0 (center) to -100 (opposite channel), default 0] 3. **Time offset:** \[0 - 20 ms, default 0] - applying an offset can enhance the illusion.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) 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: 1. **Decay amount** \[dB]: how much quieter each bounce is. 2. **Delay time** \[seconds]: 3. **Number of bounces** \[times]: **Note:** The latest available delay plugin in Audacity includes bouncing ball delay and pitch shifting. but not panning.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Combines the Bouncing Ball Delay with a panning effect. A delay effect in which the echo get faster, like a bouncing ball. Each echo is panned further from center by the designated amount. This plugin has bugs and may produce unpredictable results with some settings, but is basically functional. Parameters: 1. **Decay amount** \[dB]: 2. **Delay time** \[seconds]: 3. **Number of bounces** \[times]: 4. **Pan spread movement** \[move]: - defines the extent to which each bounce will be increasingly far from center **Note:** The latest available delay plugin in Audacity includes bouncing ball delay and pitch shifting, but not panning.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) 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: 1. **Decay amount** \[dB]: 2. **Delay time** \[seconds]: 3. **Number of bounces** \[times]: 4. **Tone shift (whole)** \[semitones]: 5. **Tone shift (cents)** \[cents]: **Notes:** 1. The value for the decay amount (in dB) for an increasing pitch can be left at the default 0. However, with decreasing pitch, the lengths of the delays increase over time, overlapping with each other. In this case, clipping can occur if the decay value is left at 0. 2. **Note:** The latest available delay plugin in Audacity includes bouncing ball delay and pitch shifting.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) The fastest bounces come first, gradually slowing down - reverse of the bouncing ball delay effect. Includes normalisation. **Note:** The latest available delay plugin 3 in [Audacity](https://web.audacityteam.org/download/) includes reverse bouncing ball delay without normalisation, and also includes pitch shifting. Parameters: 1. **Decay amount:** \[0.00 - 5.00 dB, default 0.50] 2. **Delay time:** \[0.01 - 1.00 seconds, default 0.05] 3. **Number of bounces:** \[1 - 100, default 15]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) The fastest bounces come first - reverse of the bouncing ball delay, and each bounce is tone shifted.\ **Note:** The latest available delay plugin in Audacity includes reverse bouncing ball delay and pitch shifting. Parameters: 1. **Decay amount:** \[0.00 - 5.00 dB, default 0.05] 2. **Delay time:** \[0.01 - 1.00 seconds, default 0.02] 3. **Number of bounces:** \[1 - 100, default 15] 4. **Tone shift (whole):** \[-24 - +24 semitones, default -1] 5. **Tone shift (cents):** \[-100 - +100, default 0]Details
Author: Steve Daulton Based on chimesdelay.ny by [David R.Sky](https://audionyq.com/david_r_sky) Adds random delay to your audio, and randomly changes the pitch of each delay if you specify a note list (which is where the name 'Chimes Delay' comes from). Each number in the note list indicates how many semitones your audio should be pitch shifted (along with matching tempo shift). For example, 0 indicates no pitch shift, 12 indicates a rise of 12 semitones (one octave), -5 indicates drop of 5 semitones (like going from C down to G below that C note). **Example:** If the audio you have loaded into Audacity is C3, the above note list would randomly produce the following major-sounding notes: C1 C2 G2 C3 E3 G3 C4 D4 G4 If you delete this note list, a list of notes will be generated between a lower and upper number. The default values of these two numbers are -12 semitones (decrease of 1 octave) and +24 semitones (increase of 2 octaves) respectively. If your audio is stereo, each random delay with random volume and pitch change will also be randomly panned anywhere between left and right. (It is best that your audio is first panned to center before applying Chimes Delay.) **Tips:** * Adding a bit of regular delay and/or other effects before applying Chimes Delay results in a richer sound. * If you want a particular note (from the note list) to be repeated more often, you can enter it more than once in the list. * If you simply want your audio randomly delayed with no multiple pitch changes, either enter just one number into the note list, or enter the same number into the minimum and maximum notes fields. * It is possible that total length of your resulting audio will be maximum delay time \*plus\* the duration of your original audio. This may be still longer if the final delay(s) is/are decreased in pitch (resulting in a reduced tempo). Warning: If your original audio is non-musical, Chimes Delay will not make it musical! \ Parameters: 1. **Chimes note list:** \[default list: -24 -12 -5 0 4 7 12 14 19] 2. **Minimum note:** \[semitones from -12 to +48] 3. **Maximum note:** \[semitones from -12 to +48] 4. **Maximum delay time:** \[seconds from 0.5 to 120] - the maximum delay of the random delays 5. **Minimum volume:** \[percentage] - the minimum random volume that each random delay can have. If you want no random amplitude changes, set this field to 100 percent. 6. **Number of chime delays:** \[from 1 to 100] - how many delays within the maximum delay time. Acknowledgement due to Steven Jones whose "Harmonic Noise" generator plugin is the source for Nyquist code to handle a string-input note list (David Sky's original version only).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: 1. **Decay:** \[0 - 24 dB, default -6dB] - how much quieter each subsequent delay is. 2. **Tempo:** \[1 - 500 bpm, default 120 bpm] - Specifies the tempo for the delay. 3. **Note values:** \[Choice: "Half note" to "Sixty-fourth note", default "Eighth note"] - Acts as a tempo multiplier. The default "Eighth note" produces echoes at "double tempo", whereas "Sixteenth Note" would produce echoes at 4x the tempo setting. 4. **Number of echoes":** \[1 - 100, default 5] 5. **Pan spread":** \[-1 to +1, default 0] - the amount of side to side panning of the echoes. When positive, the echo is initially panned to the right. When negative, the echo is initially panned to the left. Subsequent echoes pan side to side. When zero, the echo is mono at the center of the stereo mix. 6. **Optional filters:** \[Choice: None, High pass, Low pass, Band pass. Default: None] 7. **Filter frequency:** \[10 - 10000 Hz, default 1000 Hz] - The filter frequency (if enabled). 8. **Filter width (bandpass only):** \[0 - 10000 Hz, default 1500 Hz] - If the optional "Band Pass" filter is selected, the "Filter frequency" is the center frequency of the filter, and this control sets the width (in Hz). 9. **Wet level:** \[-24 - 0 dB, default 0 dB] - Reduces the level of each echo. Note that this effect does not extend the selected audio. If you require space for echoes after the end of the track, the selection must be made long enough to accommodate the final echoes.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Applies a high-pass filter to a delay so that with each subsequent delay, the filter's cut-off frequency is increased. A high-pass filter attenuates sound below a given cut off frequency, therefore when this 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: 1. **Decay amount:** \[0 - 24 dB, default zero] - how much quieter each subsequent delay is 2. **Delay time:** \[0 - 5 seconds, default 0.5] 3. **Number of echoes:** \[1 - 30, default 10] 4. **Start cut-off frequency:** \[100 - 5000, default 1000] - the high-pass cut-off frequency at the start of the delay. 5. **Cut-off increase:** \[octaves, 0.1 - 5.0, default 0.5] - how much to increase the filter cut-off point with each delay.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Applies a low-pass filter to a delay so that with each subsequent delay, the filter's cut-off frequency is reduced. A low-pass filter attenuates sound above a given cut off frequency, therefore when this 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: 1. **Decay amount:** \[0 - 24 dB, default zero] - how much quieter each subsequent delay is 2. **Delay time:** \[0 - 5 seconds, default 0.5] 3. **Number of echoes:** \[1 - 30, default 10] 4. **Start cut-off frequency:** \[100 - 20000, default 1000] - the low-pass cut-off frequency at the start of the delay. 5. **Cut-off reduction:** \[octaves, 0.1 - 5.0, default 0.5] - how much to reduce the filter cut-off point with each delay. 6. **Normalization level:** \[0 - 1.0, default 0.95]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) 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: 1. **Decay amount:** \[0 - 24 dB, default 0] 2. **Delay time:** \[0 - 5 seconds, default 0.5] 3. **Number of echoes:** \[1 - 30, default 10] 4. **Pitch change factor:** \[1.001 - 3.0, default 1.1] 5. **Pitch: increase or decrease:** \[default = increase] - whether each delay is increased or decreased in pitch 6. **Normalization level:** \[0.0 - 1.0, default 0.95]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) This is a stereo delay effect: with each delay, the stereo channels are flipped from left to right and vice versa. Inspired by a sound effect heard in the opening track of Mike Oldfield's "Songs From Distant Earth." Parameters: 1. **Decay amount:** \[0 - 24 dB] 2. **Delay time:** \[0 - 5 seconds] 3. **number of delays:** \[1 - 100] 4. **Normalization level:** \[0 - 1, default 0.95]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Similar to pitchshift.ny except you can define in semitones how much each delay is to be pitch shifted. A shift of 1 semitone means each delay is increased in pitch by 1 semitone, a shift of -1 means a decrease of 1 semitone. Includes whole semitones plus semitone cents (hundredths of a semitone). 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.Details
Author: Jvo Studer Adds high frequency harmonics to brighten very dull recordings that don't respond to [Equalization](https://manual.audacityteam.org/o/man/equalization.html). On already good recordings, you can add a little extra "sparkle" or "air". The harmonics are generated by soft-clipping the high frequency band as in a diode limiter, then recombining this signal with the original. Parameters: 1. **Enhancer Crossover Frequency:** \[2000 to 4500 Hz - default=3200] - lower this for very dull sources, or increase it to add only very slight or subtle high frequency harmonics. 2. **Enhancer Drive:** \[-10 to 10 dB - default = 0] - increase this to generate more harmonics and vice versa. 3. **Harmonic Generator Mode:** \[Even order,Odd order - default = Even] - generates even harmonics or odd harmonics. Even harmonics tend to be less harsh. 4. **Enhancer Noise Gate Threshold:** \[-40 -16 dB - default = -28] - increase this to prevent adding un-necessary noise in quieter recordings. 5. **Enhancer Mix Level:** \[-26 to +6 dB - default = -10] - how much of the generated harmonics are mixed into the original audio. 6. **Output:** \[Mix (Normal),Effect Only,Effect Level - default = Mix] - the two "effect" modes let you see and hear the generated harmonics on their own. Use "Effect Level" specifically to test if the Enhancer Drive level is set correctly. Edit > Undo and run Harmonic Enhancer in Mix (Normal) mode to apply the effect.Details
Author: Jvo Studer A tape saturation simulation effect. When used as intended this plugin enhances the apparent loudness of an audio track, adding just a little controlled distortion, similar to the effect of recording on a tape-based recorder with a "hot" signal. It is based on a soft clipping limiter but also "shapes" the high frequency response. The effect is not calibrated for precise modelling of a real tape recorder, but nevertheless should be capable of producing a similar tonal character. 1. **Saturation threshold:** \[-6 to -1 dB. Default = -3 dB]. The level at which the soft clipping begins. 2. **Limiting ratio:** \[1 (soft) to 4 (hard). Default 2]. The "hardness" of the clipping. High values will tend to create a more harsh sound. 3. **High freq. saturation crossover:** \[2000 to 9000 Hz. Default = 4500 Hz)] The crossover frequency where high frequency signals are limited earlier than low frequency signals. 4. **High freq. saturation reduction:** \[-8 to -1 dB. Default = -5 dB] Sets by what amount the limiting of high frequency signals exceeds that of low frequency signals. The more negative this setting, the more that high frequencies will be shaped. 5. **Auto make-up gain:** \[(choice) Off or On. Default = Off] Applies amplification after processing to make up for the lowering of the peak level produced by the effect.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: 1. **Threshold:** sets the 'cutting edge' in a linear volume number from 0.0 to 1.0 The minimum threshold is -90dB, so you can set the threshold slider to 0.0 and listen to 1-bit of a 16-bit recording if you want. The plugin has no memory limits, it can process audio tracks of several hours in length without problems.Details
Author: Edgar-rft Is in principle a similar "soft clipping" effect as [Broadcast Limiter II](#broadcast-limiter-ii), but adds an Exciter to control or intensify the high-range distortion. This function is often desired by musicians to make e.g. electric guitars or drum sets sound more aggressive. Parameters: 1. **Exciter:** controls the high-range distortion in linear numbers from 1 to 10 2. **Threshold:** sets the 'cutting edge' in a linear volume number from 0.0 to 1.0 The plugin has no memory limits, it can process audio tracks of several hours in length without problems.\\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".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: 1. **Leveling Amount (%):** (0 to 100% default 50%) Higher values produce a stronger effect.Details
 Author: Steve Daulton Usage: ### Type Soft Limit (default) progressively reduces the [gain](https://manual.audacityteam.org/man/glossary.html#gain) as the amplitude of the waveform approaches the ["Limit to (dB)"](https://manual.audacityteam.org/man/limiter.html#limit_to) threshold and prevents the waveform from exceeding that level. Hard Limit makes no change to the audio until the peaks reach the "Limit to (dB)" threshold. Where the input level (after applying optional [input gain](https://manual.audacityteam.org/man/limiter.html#Input_Gain)) exceeds the threshold, an equal amount of negative gain is applied so that the peaks never exceed the threshold. Hard Clipping is the simplest method for reducing peaks. It just chops off the peaks at the "Limit to" threshold. Note that [clipping](https://manual.audacityteam.org/man/glossary.html#clipping) causes distortion. Hard clipping may be useful for purposefully introducing distortion on high peaks, for example to add high harmonics to percussive sounds. Excessive use of hard clipping creates a harsh distortion that is usually unpleasant. For heavier use of distortion the "Soft Clipping" option may be preferable. Hard clipping may also be useful for producing synthetic signals for scientific purposes. Soft Clipping works in much the same way as "Hard Clipping", but is less fierce in that it "squashes" the peaks rather than cutting them off flat. Soft clipping starts to reduce the peaks a little below the threshold level and progressively increases its effect as the input level increases such that the threshold is never exceeded. When applied heavily, the effect is similar to a "Fuzz Box" effect. The difference between "Hard" and "Soft" clipping is that "Hard" clipping cuts off the peaks flat, whereas "Soft" clipping rounds the corners where the waveform has been clipped, resulting in a softer sounding distortion. ### Input Gain Amplifies the audio before applying the limiter. As the limiter acts on audio peaks that exceed the [Limit to (dB)](https://manual.audacityteam.org/man/limiter.html#limit_to) threshold, it will clearly have little or no effect on audio tracks in which all of the audio is below the threshold level. In such cases, the audio should be amplified before limiting so that the limiter can work properly. Amplification could be applied using Audacity's [Amplify](https://manual.audacityteam.org/man/amplify.html) effect, or more conveniently using the "Input Gain" controls. For mono tracks, only the "mono/Left" gain control has any affect. For stereo tracks the left and right channel gains may be adjusted independently of each other. ### Limit to (dB) Limits the amplitude (after optional amplifying) to this level. Whichever [type](https://manual.audacityteam.org/man/limiter.html#type) is selected, the limiter prevents the waveform from exceeding this level. (Note that [makeup gain](https://manual.audacityteam.org/man/limiter.html#mugain), if used, is applied to the waveform *after* it has been limited.) ### Hold This applies only to the "Hard Limiter" and "Soft Limiter" settings. It has no effect when using either of the "Clip" settings. In order to catch even the most sudden peak, the limiter "looks ahead" to see when the next peak is coming, and begins to reduce the gain just a little in advance of the peak. The gain level is then held at the reduced level for a short while before being released back to the normal level. Looking ahead and holding the gain level allows the gain to adjust more smoothly and reduces the amount of distortion. The shorter the "Hold" duration, the faster the limiter responds to changes in input level. It is generally desirable for the limiter to respond very rapidly, but responding *too* rapidly will produce distortion, especial when processing low frequency sounds such as a double bass. Normally this control can be left at the default (10 ms) setting. ### Apply makeup gain Amplifies the output (post limiter) close to 0 dB (usually just a little below 0 dB). This is useful when using the limiter to maximize loudness. Basically, a limiter reduces the gain (negative amplification) when the audio exceeds the "threshold" ("Limit to") level. The "Hold" time is how long (in milliseconds) the gain remains at the reduced level before returning back to normal. Usually you would want the gain to return back to normal pretty quickly after the peak has passed because you normally only want to limit the peaks and then return back to normal as quickly as possible. However, there is a problem if the limiter responds too quickly, and this is most noticeable when processing bass instruments. If there is a very low note, say below 100 Hz, then the time from one peak to the next may be longer than the "hold" time. Thus the gain will start to reduce as the peak level rises, then will start to "release" (return to normal gain) between one peak and the next. This rapid "fluttering" of the gain level distorts the waveform, which is usually undesirable. The solution to the problem is simple - just hold the gain at the reduced level for a little while so that the limiter is responding to the overall shape of the note and not the individual waveform peaks.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: 1. **Limit to (dB):** (-10 dB to 0 dB) Sets the maximum peak level. As peaks in the original audio approach this level the gain is reduced so as to prevent the peaks exceeding the set level. 2. **Hold (ms):** (1 to 50 ms Default = 10 ms) Holds the gain at the reduced level after a peak is detected so as to prevent the gain from "riding the waveform" which would cause harmonic distortion. 3. **Make-up Gain (0=No, 1=Yes):** (Default = 1) When enabled (set to 1) the output is amplified by an amount equal to the "Limit" level. If the input audio has a peak level of 0 dB, the peak output level will also be 0 dB. When disabled the peaks are limited only. Shorter *Hold* times allow the peaks to be tracked more accurately and the limiter will respond faster to the dynamics. If there are high levels of very low bass it will be necessary to increase the Hold time to avoid distortion. The default 10 ms hold time is sufficient for frequencies down to 100 Hz without distortion. To cleanly limit high amplitude, very low frequency bass (down to 50 Hz) the Hold should be increased to 20 ms. Setting the hold to 50 ms is sufficient right down to 20 Hz but the delay before the gain level "recovers" is likely to be too slow for most material. **Additional notes:** * **Threshold level:** The **Limit to (dB)** control has a range of -10 dB to 0 dB. The effect is not designed to work beyond this range. When set to 0 dB there will be no change to the audio (though any over 0 dB audio will be clipped). If set below -10 dB the *knee* will be so soft that all of the audio will be compressed, not just the peaks. * **Stereo Tracks:** As is normal for this type of effect, the left/right channels of a stereo track are processed independently. * **Lookahead:** The limiter looks ahead for peaks and will begin to change the gain just before the peak occurs. This ensures that all peaks, no matter how fast they occur, will be caught. The lookahead time is roughly a quarter of the hold time. * **Knee:** The "hardness" of the knee depends on the threshold (Limit To) level. When the threshold is close to zero a hard knee is used but as the threshold is lowered the knee becomes softer so as to provide a smooth transition in gain level even with a very fast attack time. A typical threshold level of around -3 dB will have a relatively "soft knee" so as to avoid unnecessary distortion. That is, the amount of compression (compression ratio) progressively increases as the input gets louder. At the "Limit to" level the compression ratio is infinite (brick wall) which ensures that peaks will not exceed the limit. * **Creative use:** The limiter can be used on its own, or can be used to limit peaks after running a compressor that does not use lookahead (such as the SC4 LADSPA compressor). This can produce "crisper" compression than using a lookahead compressor such as the standard Audacity compressor or Chris's dynamics compressor. * **Over 0 dB input:** The input waveform should not exceed 0 dB. Over 0 dB input signals are treated as *illegal* and will be hard clipped to 0 dB before processing with the limiter. If necessary, the Amplify or Normalize effects should be run before applying this limiter to ensure that the input does not exceed 0 dB.Details
Author: Steve Daulton. The same as [Limiter](#limiter) except that the make up gain control is a multi-choice selection rather than a slider. Parameters: 1. **Limit to (dB):** (-10 dB to 0 dB) 2. **Hold (ms):** (1 ms to 50 ms Default = 10 ms) 3. **Apply Make-up Gain:** \[No, Yes (default)]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: 1. **Select Function:** \[Apply the Noise Gate effect | Test the noise level | View one of the Help screens]. 2. **Stereo Linking:** \[Link Stereo Tracks *(gate audio when both channels fall below the gate threshold)*| Don't Link Stereo *(gate channels independently)*] 3. **Apply Low-Cut filter:** \[No *(Do not apply filter)* | 10Hz 6dB/octave | 20Hz 6dB/octave] *Removes sub-sonic frequencies including DC offset.* 4. **Gate frequencies above:** \[0 kHz to 10 kHz] *Applies the gate only to frequencies above the set level which may be useful for reducing tape hiss, but will also introduce some 'phase shift'. Setting this below 0.1 kHz will switch this feature off.* 5. **Level reduction:** \[-100 dB to 0 dB] *How much the gated sections are reduced in volume. Values below -96 dB 'shut' the gate to produce absolute silence.* 6. **Gate threshold:** \[-96 dB to -6 dB] *When the audio level drops below this threshold the gate will 'close' and the output level will be reduced. When the audio level rises above this threshold the gate will 'open' and the output will return to the same level as the input.* 7. **Attack/Decay:** \[10 to 1000 milliseconds] *How quickly the gate opens and closes. At the minimum (10 ms) the gate will fully open and close almost instantly as the audio level crosses the threshold. At the maximum (1000 ms), the gate will begin to slowly open (fade-in) 1 second before the sound level exceeds the Threshold, and will gradually close (fade-out) after the sound level drops below the Threshold over a period of 1 second.* For more detailed information and usage tips, read the help file included in this [ZIP package](https://wiki.audacityteam.org/w/images/a/a6/NoiseGate.zip), or the help screens included in the plugin.Details
Author: Steve Daulton The effect is like an "upside-down" [Noise Gate](#noise-gate). Whereas a [Noise Gate](#noise-gate) attenuates sounds that are below a specified threshold level, Pop Mute attenuates sounds that are above a specified threshold level. The effect can be used to heavily attenuate loud sounds. It may be useful for rescuing recordings that suffer from loud clicks or pops. Sounds (such as 'pops') that have a peak level above the 'Threshold' level will be lowered to a 'residual' level set by the 'Mute Level'. Be aware that ALL sounds above the threshold will be affected. Take care to avoid selecting loud sounds that should not be muted. The effect 'looks ahead' for peaks so that it can begin to lower the level of the sound smoothly a short time before the peak occurs. This is set by the 'Look ahead' time value. After the peak has passed, the level will smoothly return to normal over a period set by the 'Release time' setting. To attenuate brief clicks, time values of around 5 ms are likely to work well. For larger pops, values of 10 ms or more may sound better. For reverberant sounds such as hand claps, the 'Release time' may be increased so as to catch some of the reverberation. Parameters: 1. **View Help:** \[No | Yes] (default "No") View the built-in help screen. 2. **Threshold:** \[-24 dB to 0 dB] (default -6 dB) This is the level above which sounds are acted on (reduced in level) 3. **Mute Level:** \[-100 dB to 0 dB] (default -24 dB) How much to reduce the peak level by. 4. **Look ahead:** \[1 to 100 milliseconds] (default 10 millisecond) How far to look ahead for the next "pop" or "crackle". 5. **Release time:** \[1 to 1000 milliseconds] (default 10 millisecond) How rapidly to "release" the effect and return to normal volume after the pop has passed.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: 1. **Select function:** \[choices: Apply Effect, View Quick Help, View Examples, View Tips. Default = "Apply Effect"] 2. **Time Units:** \[choices: milliseconds, seconds, minutes, percent. Default = seconds] 3. **Amplification Units:** \[choices: dB or Percent. Default = dB] 4. **Initial Amplification** \[Numeric input. Default = none] 5. **Final Amplification** \[Numeric input. Default = none] 6. **Intermediate Control Points as pairs of time and amplification** \[Pairs of numbers. Default = none] Note: Decimal values must use a dot as the decimal separator.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.Details
Author: Kai Fisher A [Chebyshev filter](http://en.wikipedia.org/wiki/Chebyshev_filter) with options for high-pass or low-pass operation. Type I Chebyshev filters can provide a steeper [roll-off](http://en.wikipedia.org/wiki/Roll-off) than [Butterworth filters](http://en.wikipedia.org/wiki/Butterworth_filter) but at the expense of more [ripple](http://en.wikipedia.org/wiki/Ripple_\(filters\)#Frequency-domain_ripple) in the [passband](http://en.wikipedia.org/wiki/Passband). 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: 1. **Filter Type:** \[choice: Lowpass / Highpass] (default Lowpass) 2. **Order:** \[choice 2 to 30 in steps of 2] (default 6) The higher the "order" number, the steeper the cut-off transition from the passband to [stop band](http://en.wikipedia.org/wiki/Stopband). 3. **Cut-off Frequency:** \[1 to 48000 Hz] (default 1000 Hz). The actual filter frequency is limited to half of the track sample rate (the [Nyquist frequency](http://en.wikipedia.org/wiki/Nyquist_frequency)). For example, if the track sample rate is 44100 Hz, then setting the Cut-off frequency to any value greater than 22050 will produce the same result as setting the frequency to 22050 Hz. 4. **Ripple:** \[0.0 to 3.0 dB] (default 0.05) Lower values will produce less ripple in the passband at the expense of a less steep cut-off. Higher values will produce a steeper cut-off but with more ripple in the passband. The difference in ripple and cut-off slope is likely to be most noticeable with low order filters and may be noticed as a slight boost or ringing in the passband just before the cut-off frequency. When Ripple is set to zero, the passband response is essentially flat and the filter has the characteristics of a Butterworth filter. The high-pass and low-pass filters may be used one after the other to produce a "flat topped" [band-pass](http://en.wikipedia.org/wiki/Band-pass_filter) effect, in which the lower cut-off is provided by the high-pass filter and the upper cut-off provided by the low pass filter. The passband is the frequency band that passes between these two cut-off frequencies.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.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) The name 'comb' filter comes from how it acts on the audio spectrum it's applied to: it looks like a comb with the teeth pointing up. For example, if you set the comb frequency at 1000 Hz, the comb filter emphasizes 1000 Hz as well as 2000, 3000, 4000 Hz and succeeding frequencies. Produces an 'airy' effect, which is more pronounced the higher the comb decay value is set, and resonance is increasingly produced as well. A comb filter can be produced using flanger-like settings on a delay effect, but this filter does not use a delay to get the result, so it does sound somewhat different. Parameters: 1. **Comb frequency:** \[Hz, 20 - 5000, default 440] 2. **Comb decay:** \[0 - 0.1, default 0.025] 3. **Normalization level:** \[0.0 - 1.0, default 0.95]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Parameters: 1. **Center frequency:** \[Hz, 20 - 20000, default 440] 2. **Band width in octaves** \[octaves, 0.1 - 5.0, default 1.0] 3. **Gain:** \[dB, -48.0 - +48.0, default 0.0] 4. **Apply normalization?** \[Default = "no"] 5. **Normalization level:** \[0.0 - 1.0, default 0.95]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 "[British EQ](http://en.wikipedia.org/wiki/British_EQ)". 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: 1. **100 Hz HPF:** (+/- 15 dB) attenuates frequencies below 100 Hz by 12 dB per octave. It may be used to reduce low frequency noise such as microphone popping, stage noise and tape transport rumble. 2. **HF Gain:** sets the gain of the high frequency shelf filter which boosts or cuts high frequencies. Positive values will tend to make the sound "brighter". Negative values will tend to make the sound less bright. 3. **High-Mid Frequency:** (500 Hz to 15 kHz) sets the center frequency of the high-mid band filter. 4. **High-Mid Gain:** (+/- 15 dB) sets the gain of the high-mid band filter. 5. **Low-Mid Frequency:** (35 Hz to 1 kHz) sets the center frequency of the low-mid band filter. 6. **Low-Mid Gain:** (+/- 15 dB) sets the gain of the low-mid band filter. 7. **LF Gain:** (+/- 15 dB) sets the gain of the low frequency shelf filter. Positive values will tend to give the sound more bass and negative values will reduce the bass.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) A high pass filter with Q, or resonance. A high pass filter attenuates frequencies below a given cut-off point. The higher Q is, the more the cut-off frequency will resonate (produce a tone). Applied to white noise, both this filter and the low pass filter with Q can be used to produce wind-like sounds at a constant frequency. See the [high pass filter (LFO)](#undefined) and low pass filter (LFO) for ability to modulate a fixed resonance cut-off frequency. Parameters: 1. **Cut-off frequency:** \[20 - 10000 Hz, default 1000] 2. **Filter Q (resonance):** \[0 - 5, default 1]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) A high pass filter with a low frequency oscillator (LFO). A high pass filter attenuates frequencies below a given cut-off point. The LFO in this plugin modulates the cut-off frequency up and down, like on an electronic synthesizer. Parameters: 1. **LFO frequency:** \[0 - 20 Hz, default 0.2] - defines the speed of the oscillation, higher is faster 2. **Lower cut-off frequency:** \[20 - 20000 Hz, default 160] 3. **Upper cut-off frequency:** \[20 - 20000 Hz, default 2560] 4. **LFO starting phase:** \[-180 to + 180 degrees, default 0]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Parameters: 1. **Center cut-off frequency:** \[20 to 20000 Hz, default 640] 2. **LFO depth (radius):** \[0.0 to 10.0, default 1] - how far (in octaves) from center f the filter sweeps. 3. **LFO frequency:** \[0.0 to 20.0, default 0.2] 4. **LFO starting phase:** \[-180 to + 180 degrees, default 0]Details
Author: Steve Daulton A filter for removing the sound of [mains hum](https://en.wikipedia.org/wiki/Mains_hum) 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 [harmonics](https://en.wikipedia.org/wiki/Harmonic). 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. Parameters: 1. **Select Region:** \[Europe (50Hz) or USA (60Hz), default 50Hz] - Sets the fundamental hum frequency. 2. **Number of odd Harmonics:** \[0 to 200, default 1] - The first harmonic is 50 or 60 Hz depending on the region selected. 3. **Number of even Harmonics:** \[0 to 200, default 0] - The number of even harmonics to filter. 4. **Hum Threshold Level (0 to 100%):** \[0 to 100, default 10] - The signal level, as a percentage of 'full scale' below which the filters are applied.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) A low pass filter with a low frequency oscillator (LFO). A low pass filter attenuates frequencies above a given cut-off point. The LFO in this plugin modulates the cut-off frequency up and down, like on an electronic synthesizer. Parameters: 1. **LFO frequency:** \[0 - 20 Hz, default 0.2] - defines the speed of the oscillation, higher is faster 2. **Lower cut-off frequency:** \[20 - 20000 Hz, default 160] 3. **Upper cut-off frequency:** \[20 - 20000 Hz, default 2560] 4. **LFO starting phase:** \[-180 to + 180 degrees, default 0]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Parameters: 1. **Center cut-off frequency:** \[20 20000 Hz, default 640] 2. **LFO depth (radius):** \[0.0 to 10.0, default 1] - how far (in octaves) from center f the filter sweeps. 3. **LFO frequency:** \[0.0 to 20.0, default 0.2] 4. **LFO starting phase:** \[-180 to + 180 degrees, default 0]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) A low pass filter with Q, or resonance. A low pass filter attenuates frequencies above a given cut-off point. The higher Q is, the more the cut-off frequency will resonate (produce a tone). Applied to white noise, both this filter and the high pass filter with Q can be used to produce wind-like sounds at a constant frequency. See the [low pass filter (LFO)](#low-pass-filter-lfo) and [high pass filter (LFO)](#high-pass-filter-lfo) for ability to modulate a fixed resonance cut-off frequency. 1. **Cut-off frequency:** \[20 - 10000 Hz, default 1000] 2. **Filter q (resonance):** \[0 - 5, default 1]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Select total number of bands (T, from 2 to 30), band number (1 to 30, depending on how many total bands T you chose), and apply gain (-24 to +24 db). Determines width of band depending on total band number T you chose.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: 1. **Center/Cut-off:** \[0 - 10000 Hz, default 100] - sets the static filter frequency 2. **Depth:** \[-10000 - +10000 Hz, default 5000] - sets the negative or positive filter modulation depth 3. **Band Width:** \[50 - 400 Hz, default 100] - controls the resonance, lower values being more resonant 4. **Mode:** \[0="Low" 1="High" 2="Notch" 3="Band" (default)] - sets the filter mode: 0 = "Low pass", 1 = High pass, 2 = Band Reject (cut a notch at the filter frequency), 3 = Band PassDetails
Authors: Steve Daulton and Bill Wharrie. Like its name suggests, a notch filter cuts out a "notch" in the spectrum of your audio. The default frequency (60 Hz) can remove much of the hum that recordings can acquire from 60 Hz mains supply (as used in North and Central America and much of South America). You can set Frequency to 50 Hz to counteract mains hum in other countries. See [chart of mains frequencies by country](http://en.wikipedia.org/wiki/Mains_power_around_the_world#Table_of_mains_voltages_and_frequencies). Filter frequencies above 10000 Hz may be entered by typing the value but are only valid up to half of the sample rate of the audio being processed. Q values outside of the slider range can be entered by typing the values but must be greater than 0.01. Parameters: 1. **Frequency:** \[0 - 10000 Hz, default 60 Hz] 2. **Q:** \[0.1 - 20.00, default 1.00] - determines the width of the notch. Below 1 creates a wider notch, above 1 creates a narrower notch.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: 1. **Frequency (Hz):** \[10 to 10000 Hz, default 1000 Hz] - sets the center frequency of the filter 2. **Width:** \[0 to 10, default 5] - determines the width of the affected frequency band. Greater width settings affect a broader range of frequencies. Smaller width affects a narrower band of frequencies. Numerically the width setting is approximately the half gain width in half octaves, thus the default setting of 5 has a half gain width of approximately 2.5 octaves. 3. **Gain (dB):** \[-15 to +15 dB, default 0 dB (no effect)] - how much the filter center frequency is boosted or attenuated.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Like someone is playing around with the cut-off frequency knob of your low pass filter. Because of the way the random signal is generated, the lower the maximum speed is, the higher the depth factor must be to produce a similar depth of filtering changes. If you generate white noise then apply this effect, you can to some extent simulate constant pitch wind noise. Parameters: 1. **Max filter sweep speed:** \[0.01 - 10.0 Hz, default 0.2] - maximum speed of the random filter cut-off changes 2. **Filter depth factor:** \[1 - 300, default 20] - how extreme the random filter cut-off changes are 3. **Maximum cut-off frequency:** \[20 - 5000 H, default 2000] - the filter's maximum cut-off frequencyDetails
Author: Steve Daulton A filter with low pass, high pass and band pass options with a "resonance" control. Audio filters are commonly designed to have a smooth frequency response that is essentially flat in the pass band then rolls off to a lower level in the stop band, but in some cases it is desirable to use a filter that has a peak and accentuates frequencies close to the defined filter frequency. Such filters are commonly used in sound synthesis to cause "ringing" at specified frequencies. This tends to be most effective with sounds that have complex frequency content, such as noise. Parameters: 1. **Filter frequency:** \[1 to 20000 Hz] (default: 1000 Hz) - The corner frequency of the filter. The frequency must be below the Nyquist Frequency (half the sample rate) or an error message will be displayed. 2. **Resonance (Q):** \[0.1 to 100] (default: 10) - The amount of resonance. Higher values will produce a more pronounced and narrower peak at the corner frequency. Lower values will produce a less prominant peak with values below 0.7 showing no peak at all. 3. **Filter type:** \[choice: Low Pass, High Pass, Band Pass] (default: Low Pass) - Low pass allows frequencies below the corner frequency to pass through the filter and reduces frequencies above the corner. High Pass allows frequencies above the corner to pass and reduces frequencies below the corner. Band Pass reduces frequencies that are below the corner and reduces frequencies that are above the corner, allowing only a band of frequencies around the corner frequency to pass. 4. **Output Gain:** \[-60 to 0 dB] (default -12 dB) - Because the resonance accentuates frequencies around the corner frequency it is often necessary to reduce the output level of this effect. Lower (more negative) values reduce the level more.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: 1. **Filter type:** \[low-shelf / high-shelf / mid-band] - specifies which type of filter 2. **Low frequency cut-off:** \[1 to 10000 Hz] - The corner frequency for the low shelf filter, or the lower corner frequency for the mid-band filter. 3. **High frequency cut-off:** \[0.1 to 20 kHz] - The corner frequency for the high shelf filter, or the upper corner frequency for the mid-band filter. The high frequency cut-off must be less than half the track sample rate. 4. **Filter gain:** \[+/- 30 dB] - how much to boost or cut the filtered audio. Positive values boot and negative values reduce the level.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) An Equalizer (EQ) that can modify one band at a time. Select the **band number** (1 to 10) and **gain** (-24 to +24 dB).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 **rendered**. * If you apply an effect to a selection within a time-stretched clip then Audacity will **split the original clip** so that the selection can be rendered as part of applying the effect. ### Action #### Bandwidth Limited Actions WARNING: The Low Cut and High Cut sliders affect these actions. * **Remove Vocals: to mono:** (default). If the audio is center-panned, removes the vocal range defined by the Low Cut and High Cut sliders, and returns it as a dual-channel mono track. Audio is said to be "*center-panned*" if it is common to both left and right channels. *Cancellation* of center panned audio is achieved by the well known "invert and mix" method. NOTE: This option is usually much quicker than the other actions because the processing is much simpler. * This setting is identical to the **"Vocal Remover** > Remove frequency band" effect in previous versions of Audacity. * The Strength slider is **not** used when "Remove Vocals: to mono:" is selected. * **Remove Vocals:** If the audio is center-panned, removes the vocal range defined by the Low Cut and High Cut sliders, and returns it as a stereo track. * **Isolate Vocals:** If the audio is center-panned, extracts the slider-defined vocal range and returns it as a (dual) mono track. * **Isolate Vocals and Invert:** If the audio is center-panned, extracts the slider-defined vocal range and returns it as an inverted (dual) mono track. HINT: **Inverting** a waveform is the action of flipping the audio samples upside-down, reversing their polarity. The positive samples (above the horizontal zero line in the Audacity Waveform) are moved below the zero line (so becoming negative), and negative samples are made positive. * This option is equivalent to applying the **Isolate Vocals** option and then inverting. * The "and Invert" options may be useful when processing using duplicated tracks. See the examples below. #### Full Spectrum Actions WARNING: The Low Cut and High Cut sliders have no effect with these actions. * **Remove Center: to mono:** Removes all audio (the whole frequency spectrum) that is common to both left and right channels and returns a (dual) mono track. *Cancellation* of center panned audio is achieved by the well known "invert and mix" method. NOTE: This option is usually much quicker than the other actions because the processing is much simpler. * This setting is identical to the "**Vocal Remover** > entire spectrum" effect in previous versions of Audacity. * ***None*** of the sliders are used when "Remove Center: to mono" is selected. * **Remove Center:** Removes all audio (the whole frequency spectrum) that is common to both left and right channels and returns a true stereo track. * **Isolate Center:** If the audio is center-panned, extracts the whole frequency spectrum and returns it as a (dual) mono track. * **Isolate Center and Invert:** If the audio is center-panned, extracts the whole frequency spectrum from and returns it as an inverted (dual) mono track. INFO: This option is equivalent to applying the **Isolate Center** option and then inverting. #### Analysis Action * **Analyze:** Tells you how great the chances are of a successful vocal reduction or isolation. It returns also the average Pan position for the selected audio. See below for an in-depth explanation. NOTE: This option only analyzes the audio. The selected audio is not modified by the effect. ### Strength The shape of the center is not a thin band but rather like a tent with a pole in the middle. This means that there will always audio from other pan positions included. The **Strength** slider modifies the shape of the center. Higher values increase the degree of reduction or isolation. Zero is off (no reduction or isolation). This slider has no effect with the "Remove Center Classic: (Mono)" choice. #### Remove * **Less than 1.0:** Produces a notch with a ***V*** shape like a ship's keel, but also with a sharp point. Use this setting to preserve some amount of the center. * **1.0 - the default:** Produces a notch with a ***V*** shape. The power distribution is preserved. This is the ideal setting for most cases. * **Greater than 1.0:** Produces a notch with a ***U*** shape. This removes some audio adjacent to the center, as well as the center. Note that this will not remove stereo reverberation since it is distributed over the entire pan width. #### Isolate * **Less than 1.0:** Produces a peak like a ***U*** shape upside down, similar to an arch. Only extreme left and right are eliminated. * **1.0 - the default:** Produces a peak like a **''V**'' shape upside down. The power distribution is preserved. You might want a higher value when a lot of side audio leaks in. This is also the recommended setting when working with two tracks (options with **...and Invert**). * **Greater than 1.0:** Produces a peak with an ***A*** shape like the Eiffel Tower (with a sharp point). This will attenuate most of the side energy but might produce artifacts or musical noise. Normalize the audio after using Isolate with a high strength setting. ### Low Cut for Vocals All actions with **Vocals** in the name use this value. The frequencies below will either not be removed or not be included in the isolated audio. It is useful to exclude bass and kick drum. The default value of 120 Hz is good for most lead vocals, or even low male voices. You can enter a higher value around 170 Hz for female voices and about 230 Hz for those of children. NOTE: This control only affects "Actions" that contain the word **Vocals** in their name. ### High Cut for Vocals All actions with **Vocals** in the name use this value. The frequencies above will either not be removed or not be included in the isolated audio. It is useful to exclude high sounds like bells, cymbals or Hi-hat. Note that human sounds like **S** or **Z** can also be very high in frequency - 5000 to 8000 Hz, so listen carefully to the preview. NOTE: This control only affects "Actions" that contain the word **Vocals** in their name. ### Interpreting the analysis results It is recommended to analyze the audio prior to actual processing of the effect. Analysis is very fast compared to the time needed for processing. The two important values are: * **Pan Position:** This gives the average pan position for the whole selected audio. The value of the track pan slider is not included in this calculation. * **Correlation:** This is a measure for the similarity of the two channels. Values of +/-100 mean that the channels are exactly the same, even if one is inverted. A value of 0 means no relationship. It is fruitless to attempt removal or isolation in these cases. The ideal value is normally around +50. Values below 0 are rare and indicate that the stereo width is more than 100%. Inverting one channel makes the correlation positive. NOTE Sometimes the lead vocals are not exactly in the middle. In this case, select a short portion of audio where only the voice is audible and no other instrument. Choose **Analyze**. 1. The correlation should be fairly high (around 100) which would indicate that there is indeed only the voice. 2. Copy the value for the Pan Position. Double-click in the Pan slider to open the adjustment dialog then paste in the value with reversed sign (that is, put a minus sign in front of a positive value and a positive sign in front of a negative value). 3. Mix and render the track. The voice will now be exactly centered and you can remove or isolate it. 4. Do not forget to readjust the selection. ### Limitations * The input must be a true stereo track and not mere (dual) mono. * Stereo Reverberation will not be fully removed because it is independently distributed over the whole stereo image. * Naturally the plug-in does not know what kind of audio is in the center. All is removed or isolated equally, whether vocals, bass or solo instruments. * This effect is quite slow, except for the *"Remove Center Classic: (Mono)"* and *"Analyze"* actions. WARNING: The whole processing is done in memory. Long selections (over half an hour) might therefore cause Audacity to crash. ### Examples This effect can be used in a variety of ways: * Creating a Karaoke version from an original song\ Choose **Remove Vocals** if you want to keep some bass and drum beat. * Creating an acapella version from an original song\ Choose **Isolate Vocals** with a high Strength value. The result will probably still have some music in it. It is therefore better suited for a remix with a different accompaniment. Pure vocal versions have to be post-processed with other tools. * Alternative to "Auto Duck" with podcasts\ Duplicate the track and choose **Isolate Center and Invert** on the second track. Record your voice. Silence the second track where you do not speak and the original music should be heard. On the same track, make fades at the boundaries (where the speech starts or stops). * Removing excessive stereo reverb from a vocal or instrument track.\ Use **Isolate** to make a single vocal or instrument track "dry" again. * Removing random system noise from a two channel recording of a mono source.\ use **Isolate** to remove random noise produced by the audio interface, the cables or the computer itself. * Removing single words or phrases\ You can remove offending content from a song or movie dialog by selecting it and choosing **Remove Vocals**. Mask it with other sounds, if necessary. * Amplifying dialogs in a movie.\ Duplicate the track, isolate the vocals and adjust the gain. * Converting stereo files into 3.1, 5.1 and other multi-channel formats.\ Duplicate the track and choose "Isolate Center and Invert" on the second track. Render all to a new track. Delete the first track, invert the second track and split the third track. Rearrange the tracks accordingly. Thus you end up with "Front Left", "Center" and "Front Right" eventually. * Extracting instrument solos\ Isolate the center if the solo is there, otherwise remove the center, split to mono and delete the superfluous track. * Blending between original, karaoke version and vocals only\ Duplicate the track and choose **Isolate Vocals and Invert** on the second track. Choose the envelope tool or fade effects to gradually change from one mode to the other. Silencing the first track results in the vocals being isolated. Silencing the second track produces the original audio. Playing both tracks together removes the vocals. * Correcting the pan position\ Analyze the whole track and copy the value for the pan position. Paste the value into the pan dialog and reverse the sign, that is, set a minus sign in front of positive values and vice versa. * Measuring similarity between two mono tracks\ Make the two tracks stereo and choose **Analyze**. The correlation value is the measure for the similarity in percent. * Applying a brick wall filter to mono tracks\ Duplicate a mono track, make it stereo and choose **Isolate Vocals** for band pass and **Remove Vocals** for band stop filtering. Adjust the frequencies accordingly.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Simulates two tape decks playing identical tapes, but out of synchronization with each other. Written to produce an effect I heard in the late 1970s: I recorded then played identical audio on two mono tape decks. There was an amazing "whooshing" effect as one tape deck "caught up" with and passed what the other tape deck was playing. This plugin allows the "whooshing" to go back and forth. Different effects are made using mono-sounding vs. "true" stereo audio. The plugin can produce some interesting stereo effects, though note that due to the greater "cross-talk" of speakers, listening in speakers and headphones will sound different. Stereo flanger-like effects can be made by (for example) applying dualtapedecks.ny to audio, applying Stereo Butterfly (static) with a spread value of zero (sounds mono after applying), then applying dualtapedecks.ny a second time with the same settings as the first time. This plugin will work on mono audio as well, but the only effect will be rising and falling changes in pitch and tempo. Parameters: 1. **LFO frequency:** \[Hz, 0.001 to 25.000] 2. **Starting phase:** \[degrees, -180 to +180, default 0] 3. **Phase difference:** \[degrees, 0 to 360, default 180] 4. **Depth:** \[0.001 to 2.000] - The larger depth is, the more pronounced the pitch and tempo shift become until there is a noticeable warble.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Unlike a regular flanger (which cycles up and down repeatedly), this plugin creates a single high-frequency flange, and you can set the position of that high-frequency point, anywhere between the start and end of the selection and beyond. The plugin works by mixing the original selected audio with a slightly shrunk (shorter) version of itself. Works on mono and stereo audio. Parameters: 1. **High frequency flange position:** \[percent, - 100 to + 200, default 0] - the position in the signal where the high frequency portion of the flange is heard. If set to 0%, the high frequency will be at the start of the selection; at 50% in the exact middle; at 100%, at the end. If you set this value below 0% or above 100%, you won't hear the highest flange frequency peak but will hear a falling or rising flange effect, as if the peak lay outside the start or end of the selection. 2. **Time decrease:** \[milliseconds, - 0.1 to 200, default 5.0] - how much the length of the original selection is decreased. 3. **Wet level:** \[percent, 1 to 99, default 50] - the 'wet' signal is the shortened signal, and the wet level is how much (in percent) to mix with the dry (unaltered) audio. 4. **Wet: inverted or positive:** \[0=inverted, 1=positive, default positive]Details
Author: Steve Daulton A variable tremolo plugin customized with controls for length and fade in/out speed of each pulse. The modulation frequency (speed) and depth transform gradually from the initial settings to the final settings. A modified square wave is used as the modulation waveform. The effect is typically applied to a single tone to create [isochronic tones](http://en.wikipedia.org/wiki/Isochronic_tones). The author supplies the plugin only as a demonstration of audio processing without endorsing or claiming any relevance to the theory or practice of [brainwave entrainment](http://en.wikipedia.org/wiki/Brainwave_entrainment). Parameters: 1. **Pulse Width:** \[percent, 0 - 100, default 40] - How long each pulse will be "on". Higher values will make the sound "on" for longer. 50% gives a "square pulse" where the sound will be on for half the time. 2. **Fade Time:** \[percent, 0 - 100, default 15] - adjusts the fade in and fade out speed of the pulse. Higher values produce a more gradual fade in and out of the pulses. At 100% the fade in/out times will be half of the pulse width. At 0% there will be no fade. 3. **Initial Modulation Frequency:** \[Hz, 1 - 20, default 7.0] 4. **Final Modulation Frequency:** \[Hz, 1 - 20, default 2.0] 5. **Initial Modulation Depth:** \[percent, 0 - 100, default 100] 6. **Final Modulation Depth:** \[percent, 0 - 100, default 100]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Similar to Random Panning, this time playing around with the volume knob. Because of the way the random signal is generated, the lower the maximum speed is, the higher the depth factor must be to produce a similar depth of amplitude changes. Parameters: 1. **Max amp sweep speed:** \[0.01 - 20.0 Hz, default 0.5] - maximum speed of the random amplitude changes 2. **Amp sweep depth factor:** \[1 - 300, default 80] - how extreme the random amplitude changes areDetails
Author: [David R.Sky](https://audionyq.com/david_r_sky) Randomly modulates the pitch of your audio. As with other randomly-controlled effects, the the lower the maximum speed is, the higher the depth factor must be to produce a similar depth of random changes. This effect works on mono and stereo audio. In stereo, each channel has different random pitch modulation applied. "Max pitch mod depth" can be explained thus: at higher warping depth settings, pitch mod depth should be made higher, otherwise there will be momentary periods without pitch changes. With lower warping depth settings this does not happen, and the effect can be re-applied repeatedly to give further random pitch changes. Parameters: 1. **Warping depth:** \[0.001 - 2.000, default 0.100] - controls the number of pitch changes that occur 2. **Max sweep speed:** \[0.01 - 20.0 Hz, default 0.50] - maximum speed of the random pitch changes, higher values increase "warbling" effect 3. **Sweep depth factor:** \[1 - 300, default 80] - how far 4. **Max pitch mod depth:** \[0.01 - 3.00, default 0.50]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) A ring modulator is a tremolo effect, but instead of using an LFO to amplitude modulate audio, an audio signal is used. The result is a combination of the sum of and the difference between the two input signal frequencies e.g., two sine waves of 440 Hz and 660 Hz produce a result of 220 Hz (difference) and 1100 Hz (sum). This plugin also allows use of triangle, sawtooth and pulse waveforms, so the results are the sums and differences between the harmonics of the modulating signal and harmonics of the signal being modulated. Parameters: 1. **Modulation frequency:** \[Hz, 20 to 5000, default 500] 2. **Amount:** \[percent, 0 to 100, default 100] 3. **Waveform:** \[0=sine, 1=triangle, 2=sawtooth, 3=pulse, default sine] 4. **Pulse bias:** \[percent, -100 to +100, default 0] - if the pulse waveform is selected, bias is the pulse width. The default 0 gives a "square" wave. Lower values give a narrower positive signal, higher values a wider positive signal.Details
Author: Steve Daulton This plugin produces a "Tremolo" effect in which the frequency and depth of the tremolo varies from an initial setting to a final setting. The effect makes the loudness "wobble" and you can set the speed of the wobble and the amount that it wobbles for both the start of the selection and the end of the selection. Parameters: 1. **Tremolo Shape:** \[sine,triangle,sawtooth,inverse sawtooth,square. default = "sine"] 1. Sine: The volume rises and falls smoothly up and down. 2. Triangle: The volume alternates between rising at a constant rate and falling at a constant rate. 3. Sawtooth: The volume rises abruptly but falls gradually. 4. Inverse Sawtooth: The volume rises gradually but falls abruptly. 5. Square: The volume jumps abruptly between higher and lower levels. 2. **Starting Phase:** \[0 to 360 degrees, default = 90] - This sets the starting point for the tremolo cycle. At 90 degrees the tremolo starts at the higher level. At 0 degrees the tremolo starts at the lower level. 3. **Initial Tremolo Frequency:** \[1 to 20 Hz - default = 4] - The speed of the tremolo effect at the start of the selection. 4. **Final Tremolo Frequency:** \[1 to 20 Hz - default = 12] - The speed of the tremolo effect at the end of the selection. 5. **Initial Tremolo Amount:** \[0 to 100 % - default = 40] - How much the initial volume varies as a percentage of the original level. 6. **Final Tremolo Amount:** \[0 to 100 % - default = 40] - How much the final volume varies as a percentage of the original level.Details
Author: Steve Daulton This effect applies vibrato (a regular pulsating change of pitch) to the selected audio. The speed at which the sound pulsates can be varied over time by selecting an initial speed and a final speed. The vibrato speed will smoothly transition from the initial speed to the final speed. For a constant speed vibrato, set the Initial and Final speed settings to the same value. The depth of the vibrato (amount of pitch variance) can varied over time by setting initial, mid point and final depth values. The vibrato depth begins at the initial value and smoothly transitions to the mid-point value, then to the final value. The time at which the mid-point depth is achieved can be adjusted. When "Mid point position" is set to 50%, the mid-point depth is reached half way through the selection. Smaller "Mid point position" values shift the mid-point earlier, so that the transition in depth from "Initial" to "Mid point" occurs more quickly, and the transition from "Mid point" to "Final" occurs more slowly. Similarly, larger values shift the mid-point later. The maximum vibrato depth (100%) is equivalent to a pitch deviation of +/- 1 whole tones. At 50%, the deviation is +/- 1 semitone. Parameters: 1. **Initial speed:** \[0 to 10, default 1] The initial vibrato speed (wobbles per second). 2. **Final speed:** \[0 to 10, default 5] The vibrato speed at the end of the selection. 3. **Initial depth:** \[0 to 100%, default 10%] The amount of pitch variance at the start of the selection. 4. **Mid point depth:** \[0 to 100%, default 100%] The amount of pitch variance at the mid-point. 5. **Final depth:** \[0 to 100%, default 0%] The amount of pitch variance at the end of the selection. 6. **Mid point position:** \[0 to 100%, default 30%] How far through the selection that the mid-point depth is reached.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Developed from the previous sequencers 1a and 1b (these can't be recommended due to the interface being too tall, and distortion problems). You can turn any short piece of imported, recorded or generated audio into a (repeated) sequence of notes based on chosen tempo, beats per sequence and semitone values. Any sound can be used (a guitar pluck or bell sound, cat meow, someone saying "hello", in fact - any sound). Comes with an 8-note default sequence already programmed in. Can also pan stereo audio, and transpose successive sequences. To generate a rest, use r, n, (), or a blank svp input line. Parameters: 1. **Tempo:** \[beats per minute (default 210), beats per sequence (default 8), starting offset (beats) (default 0)] 2. **Pan stereo selection:** \[0 = no 1=yes (default)] 3. **Timing randomization:** \[0 - 100 plus or minus percent, default 0] 4. **Number of repeated randomized sequences:** \[0 - 8, default 0] 5. **Overall transpose value:** \[(default 0), then for successive measures (default 0 0 5 5 0 0 -5 -5)] 6. **Sequences to generate:** \[1 - 96, default 4] 7. **1st Semitone, volume, pan value\[s]:** \[default (0 1 0) (4 .5 .2) (7 .5 .8) (2 1 .5)] 8. **2nd SVP value\[s]:** \[default (12 .5 .8) (7 .5 .2) (4 1 1) r] 9. **3rd, 4th. 5th and 6th SVP value\[s]:** \[filled in by user]Details
Author: Steve Daulton Similar to the [Change Speed](https://manual.audacityteam.org/man/change_speed.html) effect, this effect changes the speed of the selected audio, thus changing both the tempo and pitch. Possible uses include slowing down music to aid transcription, where the amount of pitch change is known. Parameters: 1. **Semitones change:** \[-24 to +24 semitones]Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Extracts audio from a selected area without using a mouse or cursor keys. Audacity has a Selection Toolbar providing a screen-reader friendly display of selection start time and duration which you could use for similar purpose, this plugin provides a solution. It also has an easy option to extract a percentage of the selected audio. For example, selecting 50% "start percent" and 100% "end percent" will leave you with only the last half of your selection. Parameters: 1. **Time or percent?** 2. **Start time:** \[seconds, maximum 600] 3. **End time:** \[seconds, maximum 600] 4. **Start percent:** 5. **End percent:**Details
Author: Steve Daulton This effect inserts silence at the start of the audio selection. The length of inserted silence is equal to the length of the selection. This effect may be useful in situations where it is necessary to insert a lot of silences because Audacity (2.0.1 and later) allow keyboard shortcuts to be set for effects (see:Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) Previously called "Audio chunker". Splits audio into desired number of segments by inserting silences of specified duration (up to 10.0 seconds). You can also specify fade-in and fade-out lengths for each segment. By setting number of segments to ' 1 ' you can use audio splitter to apply a specified fade-in and fade-out to a single length of audio in one action. Parameters: 1. **Number of audio segments:** \[1 to 120, default 10] 2. **Silence duration between segments \[seconds]:** \[0.01 to 10, default 1] 3. **Fade-in/fade-out length \[milliseconds; 0 = no fade]:** \[0 to 500, default 20]Details
Author: Steve Daulton A speed change effect in which the initial and final speed may be set independently. Like Audacity's [Change Speed](https://manual.audacityteam.org/man/change_speed.html) effect, this effect changes both the tempo and pitch of the audio. **Accessibility:** This effect may be useful as an accessible alternative to Audacity's [Time Tracks](https://manual.audacityteam.org/man/time_tracks.html). Parameters: 1. **Initial speed (0.1 to 10 x):** \[0.1 to 10, default 1.0] Sets the initial speed. 2. **Final speed (0.1 to 10 x):** \[0.1 to 10, default 1.0] Sets the final speed. The speed controls set the initial and final speed as a multiple of the original speed. A setting of 1.0 means 1x the original speed, in other words, unchanged. 2x speed is double speed. 0.5x speed is half speed. The speed changes linearly between the 'Initial' and 'Final' settings.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) For those who are confused by Audacity's "Change Speed" effect where to make the tempo twice as slow you apply a -50% change and to make it twice as fast apply a 100% change, try this plugin. Its default settings multiply the tempo by 0.5, making the tempo twice as slow (dividing by 2.0 has the same effect). To make the tempo twice as fast, simply multiply by 2.0 (or divide by 0.5). The default setting (the opposite of Audacity's "Change Speed" default) might be handy for example to return tapes dubbed at 2x speed to normal speed. Parameters: 1. **Tempo change factor:** \[0.1 to 8.0, default 0.5] 2. **Multiply or divide:** \[0=multiply (default), 1=divide] - multiplies or divides by the tempo change number.Details
Author: [David R.Sky](https://audionyq.com/david_r_sky) A plugin for performing the same task as the Time Shift Tool in Audacity, without using a mouse. The effect works thus: if the shift value is positive, silence is inserted before the selection. If the shift value is negative, audio is removed from the start of the selection. If your selected track is mono, set the value in the left/mono edit field, if your track is stereo, set the values in both the left/mono and right channel shift value fields. Setting slightly different values in both fields when you have a stereo track can be used for special effect. Note that positive shifts can lead to the audio being truncated at the right edge when shifting a stereo track. To avoid this, split the stereo track using the Track Drop-down Menu, select each track in turn and apply the shift you want using the Mono/left channel shift field. If the audio starts after time zero and is preceded by empty space, convert the empty space to silence with Project > Quick Mix in legacy Audacity up to and including version 1.3.13. In current Audacity, select the track, press J, then press SHIFT + HOME, then Generate > Silence. This prevents truncation of the audio providing there is sufficient silence at the start of the track. Parameters: 1. **Mono/left channel shift:** \[-1000 - +1000, default 0] 2. **Right channel shift:** \[-1000 - +1000, default 0] 3. **\[0=milliseconds 1=seconds]** (default is milliseconds)Details
Author: Steve Daulton This plugin can trim audio from the start and/or end of a track (or track selection), or pad the start and/or end with silence, or any combination of the two. The effect can be used in a [Macro](https://manual.audacityteam.org/man/macros.html) for batch processing. Parameters: 1. **Time units:** \[samples, milliseconds, seconds (default), minutes] This selection sets the time units for the following two controls. 2. **Trim / Extend start by:** \[-100 to +100, default 0] How much audio to trim (delete) from the start of the track selection, or how much silence to add to the start of the selection. Move the slider to the left or type a negative value to remove audio. Move the slider to the right or type a positive value to add silence. 3. **Trim / Extend end by:** \[-100 to +100, default 0] How much audio to trim (delete) from the end of the track selection, or how much silence to add to the end of the selection. Move the slider to the left or type a negative value to remove audio. Move the slider to the right or type a positive value to add silence. 4. **Error message control:** \[Show errors (default) / Hide errors] In the event of an error, a message will normally be shown to indicate what the error is. For example, if you attempt to delete a longer duration than the selection, an error message will indicate that the operation cannot be done. However, when using this effect in a Macro for batch processing, the error message will cause the Macro to stop until the error message is acknowledged. That behavior may not be wanted, so "Hide errors" may be selected to suppress error messages. When "Hide errors" is enabled, if an error is encountered, the plugin will silently skip the current track and continue with any remaining tracks.Details
Author: Steve Daulton This plugin is for trimming excess silence from the start and/or end of tracks or audio clips. The effect can be used in a [Macro](https://manual.audacityteam.org/man/macros.html) for batch processing. Parameters: 1. **Silence Threshold (dB):** \[-100 to 0, default -48] Audio below this level at the start or end of the track may be trimmed. 2. **Silence to leave at start (0 to 30 seconds):** \[0 to 30 seconds, default 0] When trimming silence from the start of a track or audio clip, the silence will not be trimmed shorter than this length. Note that if the silence at the start of the track is already shorter than this length, it will not be trimmed and will not be extended. 3. **Silence to leave at end (0 to 30 seconds):** \[0 to 30 seconds, default 0] When trimming silence from the end of a track or audio clip, the silence will not be trimmed shorter than this length. Note that if the silence at the end of the track is already shorter than this length, it will not be trimmed and will not be extended. 4. **Preview** is not available for this effect. 5. This plugin requires the audio to be loaded into RAM. To avoid running out of RAM (which would cause Audacity to freeze or crash), the plugin limits the maximum size of the selected audio to 1 GB (about 47 minutes for a stereo track at 44.1 kHz). Additional information is provided in the plugin code comments.Details
Authors: [David R.Sky](https://audionyq.com/david_r_sky), Roger B. Dannenberg Simulates unplugging and plugging in a turntable while it's powered, and related effects such as speeding it up before unplugging it. Improved over the previous turntablewarp.ny - this version lets you warp both mono and stereo audio. The default settings simulate unplugging a turntable - the audio starts to slow down halfway through the selection area, and drops to 40% of original volume by the end. **Parameters:** 1. **Initial step:** \[-36 - +36 semitones, default 0] - the resulting pitch change at the start of the selection, defined as number of semitones above or below the original pitch (1 step = 1 semitone, so 12 steps = 1 octave.) 2. **Initial amplitude:** \[0 - 100 percent, default 100] - the resulting volume at the start of the selection, related to the original amplitude (so 100% = no change). 3. **Change time:** \[0 - 100 percent, default 50] - the point in time in the selection (set internally to the original pitch and volume) at which the warping values change to reach and move away from this point. What this means is that depending on your start and end step and amplitude values, you can design warps that slow down then speed up; speed up then slow down; speed up to a particular pitch then remain there; slowly speed up then quickly speed up. 4. **End step:** \[-36 - +36 semitones, default -12] - the resulting pitch change at the end of the audio. compared to the original pitch. 5. **End amplitude:** \[0 - 100 percent, default 40] - the resulting volume at the end of the selection, compared to the original volume.Details
Author: Steve Daulton Based on a plugin by Will McCown. **Displayed Results** * **Peak leve**l\ Maximum peak level in the selection * **RMS level**\ The RMS level of the selected audio. * **Noise floo**r\ The RMS level of the quietest 500 milliseconds in the selection * **Warnings**: *(These are only displayed when applicable.)* * Warning: ACX require 44100 Hz sample rate. * Warning: ACX require running time no longer than 120 minutes * Noise floor: -inf dB Warning (too low - Dead silence sounds unnatural.)\ ACX require constant, low level "room tone" rather than an unnaturally silent background noise level. **Limitations** * This tool is intended only as an aid in achieving ACX acceptance. Even straight passes from this tool is NO guarantee of ACX acceptance. * The Noise Floor measurement is taken from the quietest half second of audio found in the selection. If one part of the selection is quieter than the rest, you will get a false value. * Minimum selection length is 1/2 second. * Maximum selection length is about 2.14 billion samples (13.5 hours at a sample rate of 44100 Hz) Also beware that some noise sources are worse than others, and noise such as the 1000 Hz whine that often happens in USB audio interfaces may result in an ACX rejection even though it is below the -60 dBFS noise floor requirement.Details
©2020 MAAT Incorporated, All rights reserved. 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!* \\Details
Author: Edgar-rft. **Parameters** 1. **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. 2. **Minimum Distance: \[samples]:** \[1 to 1000, (default 100)] The minimum distance between labels (in samples). If audio is clipped there may be many samples in succession at the maximum amplitude. This setting avoids labeling every successive sample by setting a minimum distance between labels. **Limitations** * This effect can be very slow on long selections. * If the audio is clipped and "Place labels at: all equally loud peaks" is selected, there may be an extremely large number of labels created. The "Minimum Distance" setting is useful to reduce the number of labels. * Peaks that appear to be at the maximum amplitude will not be labeled unless they are **exactly** at the maximum amplitude.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** 1. **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. 2. **Analyse first (seconds):** \[0.1 to 1. Default 0.2] At the default setting the first 0.2 seconds of the selection will be analyzed. **Limitations** This plugin is intended to detect single notes - you may get strange results if you try to analyze chords. Extremely high frequencies may not be detected very accurately, especially if the sample rate is not very high. The plugin will often detect very high frequencies better if the sample rate is 96000 Hz rather than 44100 Hz. The plugin should usually be able to detect pitches of single notes to within a few percent of the actual frequency. Don't expect the frequency in Hz to be exact.. **Advanced usage tips** * For detecting very low frequencies (less than a few hundred Hz) the plugin should be set to the low frequency range (20 to 1000 Hz). * For detecting very high frequencies (several kHz) the plugin should be set to the high frequency range (1 kHz to 10 kHz). * For measuring synthesized tones and other electronic signals, the most accurate measure of frequency in Audacity is to use "Plot Spectrum" and set the "Size" setting to a high value.| Service | |
|---|---|
| /files/kyOu7TaKbseCgFyXYjwv | ACX, Audible |
| /files/IlgyUWIbrIWi2hVRTKuA | Amazon Music, Alexa |
| /files/75tO4S7IDXTppAgRDpQ1 | Apple Music |
| /files/CrHLRA0qigfjVHs7exwt | Club Play |
| /files/9G4001Jr911ue2balPSJ | Deezer |
| /files/XhiN0QxI8YEkYAu2tub0 | Netflix |
| /files/3MWBPZd4vdykxxRYTicb | (Podcasts) |
| /files/gec6OGfrWI4926lFcM0H | Sony related |
| /files/8yQQSH2tCqD7ipm91LuR | Soundcloud |
| /files/1BXPgAf7GbAsXPh8Rj3K | Spotify |
| /files/RPsrLxq6OUOUufcWqdG1 | Spotify |
| /files/tPCAWxBxgYg4EFd5izPf | Tidal |
| /files/x1diHim7fd2K79ELQ0GZ | YouTube |
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 leve**l\ Maximum peak level in the selection * **RMS level**\ The RMS level of the selected audio. * **Noise floo**r\ The RMS level of the quietest 500 milliseconds in the selection * **Warnings**: *(These are only displayed when applicable.)* **Limitations** * These tools provide useful guidance, but they do not guarantee that the services in question will reject or modify the uploaded sounds anyway. This particularly goes for ACX, which has additional quality standards not measured by this tool. * The Noise Floor measurement is taken from the quietest half second of audio found in the selection. If one part of the selection is quieter than the rest, you will get a false value. Also note that ACX requires a very quiet noise floor to be present. * Minimum selection length is 1/2 second. * Maximum selection length is about 2.14 billion samples (13.5 hours at a sample rate of 44100 Hz) Also beware that some noise sources are worse than others, and noise such as the 1000 Hz whine found in some USB audio interfaces may result in an ACX rejection even though it is below the -60 dBFS noise floor requirement.Details
*Additional info, such as developer name, license and copyright info, a short "how to use" section or link to the documentation page, or a screenshot of the plugin - preferably of it working withing Audacity.* \[screenshot, if applicable]- Analyzing selected track audio.
- Durations are relative to the selection length.
- Code iterates through each selected track.
- Generating audio.
- Durations are absolute (1 "unit" of time = 1 second).
- Code runs once only regardless of number of tracks.
- Processing selected track audio.
- Durations are relative to the selection length.
- Code iterates through each selected track.
- Nyquist Macros / Other.
- Nyquist cannot directly modify the project.
- Additional global variables to pass additional information from Audacity to Nyquist.
New optional plugin headers:
Sets tracenable (LISP) or sal-traceback (SAL) and displays debug window if there is anything to show.
This may be useful when showing debug info is integral to what the plugin does, or when debugging a script.
| | ;debugflags notrace |Disables tracenable (LISP) / sal-traceback (SAL). This prevents the debug window from opening on error, unless the Debug button has been pressed.
If the Debug button has not been pressed and there is an error, the error message will be sent to Audacity's log. Note that disabling tracenable limits the debug output to only the error message with little or no additional debugging information.
| | ;debugflags compiler |Set sal-compiler-debug to 'true'.
Output from the SAL compiler is printed to the debug output. This may be viewed in the Debug window if enabled (for example, by clicking the Debug button), otherwise in the Audacity log.
| | ;debugflags nocompiler |Set sal-compiler-debug to 'false'.
Disables compiler messages from SAL. Only debug and error messages are printed to the debug output.
| These flags may be used in conjunction with the [debugbutton](#debugbutton) header to provide a variety of behaviors. * **debugbutton true, debugflags trace**: Debug window opens if debug button pressed or if Nyquist printed to the debug window. * **debugbutton false, debugflags trace**: Debug button disabled, but debug window opens if Nyquist printed to the debug window. * **debugbutton true, debugflags notrace**: Debug window opens only when Debug button is pressed. If Debug button has NOT been pressed, any error messages that arise are routed to the [Audacity log](https://manual.audacityteam.org/man/help_menu_diagnostics.html#show_log). * **debugflags compiler, debugflags trace**: (When using SAL syntax) The Debug window always opens to display the output from the SAL compiler. * **debugflags compiler, debugflags notrace**: (When using SAL syntax) The Debug window only opens to display the output from the SAL compiler if Debug button is clicked, otherwise it is printed to the Audacity log. * **debugflags nocompiler, debugflags trace**: (When using SAL syntax) The Debug window always opens to display debug and/or error messages. Output from the SAL compiler is suppressed. ### **helpfile** A [Help button](http://manual.audacityteam.org/man/help_button.html) may be added to the plugin [GUI](https://en.wikipedia.org/wiki/Graphical_user_interface) by giving the relative path and name of a help file. As the button can only be created if the plugin has a GUI, this header should not be used in plugins that do not have a GUI. ``` ;helpfile path to file ``` The file path is relative to the plugin search path. Normally the help file would be placed in the same location as the plugin when it is installed. To support HTML files with images and/or media, the help file and its resources may be in a folder, and the folder included in the file path. For example, if the help file is called "my\_effect.html" and it includes images, then the html file and images may be placed in a folder called "my\_effect\_help", and the helpfile header would be: ``` ;helpfile "my_effect_help/my_effect.html" ``` {% hint style="info" %} If the help file is not found, the ? button will not appear. {% endhint %} ### **manpage** This is primarily intended for use by plugins that are shipped with Audacity. It is similar to the [helpfile](#helpfile) header, except that it looks for the help file in the search path for Audacity's manual. As with "*;helpfile*" it should only be used in plugins that have a GUI. For more information, see [Location of Manual](http://manual.audacityteam.org/man/interface_preferences.html) ``` ;manpage URL ``` ### **maxlen** Specifies the maximum number of samples to be processed in "process" or "analyze" type plugins. This can help the progress bar if length that will be processed is determined by the plugin rather than by the length of selection. It could also be used as a fail-safe for plugins that are specifically designed for short selections. This example limits the number of samples to 1 million: ``` ;maxlen 1000000 ``` When the MAXLEN header is used, the Nyquist global variable LEN is the maximum of "length of selection in samples" and "value set by MAXLEN". {% hint style="info" %} In "process" type effects, this may cause the selected audio to be truncated to the specified number of samples. In such cases it may be best to throw an error if "maxlen" is exceeded, for example: {% endhint %} ```lisp ;maxlen 1000000 (defun isok () ;; Return true if selection is less than "maxlen". (let ((start (get '*selection* 'start)) (end (get '*selection* 'end))) (<= (truncate (* *sound-srate* (- end start))) len))) ``` ### **mergeclips** Allows Nyquist plugins to override Audacity's default "clip merge" behaviour. By default, when effects (including generator effects) are applied across one or more clip boundaries and the returned audio is a different length from the original selection, Audacity will add "split lines" at the ends of the returned audio. In all other cases, the returned audio is "merged" into the current audio. This option only applies when the plugin is applied across clip boundaries (including across "split lines"). #### Clip Merge Options * -1 Automatic clip merge behaviour (default) ``` ;mergeclips -1 ``` * 0 Don't merge clips. Effects that are applied across clip boundaries will not be merged into the existing audio (there will be split lines at the ends of the returned audio) whether the returned audio is the same length as the original selection or not. ``` ;mergeclips 0 ``` * 1 Always merge clips. The returned audio will always be merged into the existing audio (no split lines added). ``` ;mergeclips 1 ``` See also "restoresplits". ### **preview** Provides options for previewing the effect. Multiple preview options may be defined to achieve the desired behaviour. Preview Options * enabled (default). Preview is enabled. ``` ;preview enabled ``` * true Same as "enabled". ``` ;preview true ``` * disabled Preview is disabled. If Audacity is unable to provide a meaningful preview, then preview should be disabled. This may be required for effects that affect specific time regions within the selection. ``` ;preview disabled ``` * false Same as "disabled". ``` ;preview false ``` * linear Provides an optimisation for previewing multiple tracks by mixing the selected tracks before applying the Nyquist code. This optimisation is disabled by default. ``` ;preview linear ``` * selection When previewing, the Nyquist code is applied to the entire selection (not just the length that will be previewed). Audacity's "Preview" then plays the first few seconds of the processed audio. This may be required for effects that vary over the duration of the selection. ``` ;preview selection ``` ### **restoresplits** Allows Nyquist plugins to override Audacity's default "split restore" behaviour. By default, when effects (including generator effects) are applied across one or more clip boundaries, Audacity will restore "split lines" at the position of the original clip boundaries. This option only applies when the plugin is applied across clip boundaries (including across "split lines"). Restore Splits Options * 1 Splits at clip boundaries are restored (default) ``` ;restoresplits 1 ``` * 0 Splits at clip boundaries are not restored (clips are joined). ``` ;restoresplits 0 ``` Note: Nyquist plugins are not currently able to distinguish between silence and "empty space" within the selection, so gaps between audio clips will be treated as if the empty space is an additional "silent" audio clip. See also "mergeclips". ## Obsolete plugin headers {% hint style="warning" %} These headers are obsolete and no longer used by Audacity. They should not be used in new plugins. They may still be found in some old plugins, but are now treated by Audacity as ordinary "comments" and simply disregarded. {% endhint %} ### **action** Description shown in progress window. ``` ;action "text" ``` ### **categories** Specifies an LV2 category for the plugin. ``` ;categories "text" ``` ### **info** A single line of text to be displayed at the top border of the plugin window. For multiple lines of text, a two-character sequence "\n" may be used within "text" to create a line break. ``` ;info "text" ``` ## Control headers If the plugin includes at least one properly formed "Control" header, Audacity will launch the plugin with a GUI. Each valid "Control" header adds a widget to the GUI. If there are no "Control" headers, Audacity will attempt to run the effect without a GUI. There can be several "control" lines in the plugin header. Add one for each widget to appear in the dialog box. The ;control headers should normally be the final headers as they define variables used by the Nyquist code. ``` ;control parameters ``` # Widgets Reference A [Widget](https://en.wikipedia.org/wiki/Widget_\(GUI\)) is an element of a graphical user interface ([GUI](https://en.wikipedia.org/wiki/Graphical_user_interface)), such as a button or slider, which facilitates user interaction. Audacity supplies a number of widget types for Nyquist plugins, which are ultimately derived from the [WxWidgets](https://www.wxwidgets.org/) toolkit. Fortunately, Nyquist plugin developers are largely spared from the complexities of WxWidgets, and can simply select which widgets they need by adding the appropriate "header" to the top of the plugin script. The layout of a Nyquist plugin GUI is a simple list of widgets, one above the other. ### Slider Widgetexample of a slider
example of numeric text
example of a text input
example of multiple choice input
example of time input
example of file buttons
example of text
Nyquist prompt example
Original signal before command
Resulting signal
resulting waveform
File button widget example
| Scripting Id | Action | Parameters | Description |
|---|---|---|---|
| Save: | Save Project | none | Saves the current Audacity project .AUP3 file. |
| SaveAs: | Save Project As... | none | Same as "Save Project" above, but allows you to save a copy of an open project to a different name or location |
| SaveCopy: | Save Lossless Copy of Project... | none | Saves the current Audacity project .AUP3 file. |
| SaveCompressed: | Save Compressed Copy of Project... | none | Saves in the audacity .aup3 project file format, but compressed (Suitable for mailing) |
When Audacity is playing, recording or paused, sets the left boundary of a potential selection by moving the cursor to the current position of the green playback cursor (or red recording cursor).
Otherwise, opens the "Set Left Selection Boundary" dialog for adjusting the time position of the left-hand selection boundary. If there is no selection, moving the time digits backwards creates a selection ending at the former cursor position, and moving the time digits forwards provides a way to move the cursor forwards to an exact point.
| | SetRightSelection: | Right at Playback Position | *none* |When Audacity is playing, recording or paused, sets the right boundary of the selection, thus drawing the selection from the cursor position to the current position of the green playback cursor (or red recording cursor).
Otherwise, opens the "Set Right Selection Boundary" dialog for adjusting the time position of the right-hand selection boundary. If there is no selection, moving the time digits forwards creates a selection starting at the former cursor position, and moving the time digits backwards provides a way to move the cursor backwards to an exact point.
| | SelTrackStartToCursor: | Track Start to Cursor | *none* | Selects a region in the selected track(s) from the start of the track to the cursor position. | | SelCursorToTrackEnd: | Cursor to Track End | *none* | Selects a region in the selected track(s) from the cursor position to the end of the track. | | SelTrackStartToEnd: | Track Start to End | *none* | Selects a region in the selected track(s) from the start of the track to the end of the track. | | SelSave: | Store Selection | *none* | Stores the end points of a selection for later reuse. | | SelRestore: | Retrieve Selection | *none* | Retrieves the end points of a previously stored selection. | ### Select: Spectral For making a selection of a frequency range. | Scripting Id | Action | Parameters | Description | | ------------------------ | -------------------------- | ---------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- | | ToggleSpectralSelection: | Toggle Spectral Selection | *none* | Changes between selecting a time range and selecting the last selected spectral selection in that time range. This command toggles the spectral selection even if not in Spectrogram view, but you must be in Spectrogram view to use the spectral selection in one of the Spectral edit effects. | | NextHigherPeakFrequency: | Next Higher Peak Frequency | *none* | When in Spectrogram view, snaps the center frequency to the next higher frequency peak, moving the spectral selection upwards. | | NextLowerPeakFrequency: | Next Lower Peak Frequency | *none* | When in Spectrogram views snaps the center frequency to the next lower frequency peak, moving the spectral selection downwards. | ### Select: Clip Boundaries For modifying a selection, taking account of clips. | Scripting Id | Action | Parameters | Description | | ---------------------------- | -------------------------------- | ---------- | ------------------------------------------------------------------------------------------ | | SelPrevClipBoundaryToCursor: | Previous Clip Boundary to Cursor | *none* | Selects from the current cursor position back to the right-hand edge of the previous clip. | | SelCursorToNextClipBoundary: | Cursor to Next Clip Boundary | *none* | Selects from the current cursor position forward to the left-hand edge of the next clip. | | SelPrevClip: | Previous Clip | *none* | Moves the selection to the previous clip. | | SelNextClip: | Next Clip | *none* | Moves the selection to the next clip. | ## View Menu 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. | Scripting Id | Action | Parameters | Description | | --------------- | ---------------------- | ---------- | --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- | | UndoHistory: | History... | *none* | Brings up the History window which can then be left open while using Audacity normally. History lists all undoable actions performed in the current project, including importing. | | Karaoke: | Karaoke... | *none* | Brings up the Karaoke window, which displays the labels in a "bouncing ball" scrolling display | | MixerBoard: | Mixer Board... | *none* | Mixer Board is an alternative view to the audio tracks in the main tracks window. Analogous to a hardware mixer board, each audio track is displayed in a Track Strip. | | ShowExtraMenus: | Extra Menus (on/off) | *none* | Shows extra menus with many extra less-used commands. | | ShowClipping: | Show Clipping (on/off) | *none* | Option to show or not show audio that is too loud (in red) on the wave form. | ### View: Zoom Zoom in/out on the horizontal axis. Show more detail or show a longer length of time. | Scripting Id | Action | Parameters | Description | | -------------- | ------------------------- | ---------- | -------------------------------------------------------------------------------------------------- | | ZoomIn: | Zoom In | *none* | Zooms in on the horizontal axis of the audio displaying more detail over a shorter length of time. | | ZoomNormal: | Zoom Normal | *none* | Zooms to the default view which displays about one inch per second. | | ZoomOut: | Zoom Out | *none* | Zooms out displaying less detail over a greater length of time. | | ZoomSel: | Zoom to Selection | *none* | Zooms in or out so that the selected audio fills the width of the window. | | ZoomToggle: | Zoom Toggle | *none* | Changes the zoom back and forth between two preset levels. | | AdvancedVZoom: | Advanced Vertical Zooming | *none* | Enable for left-click gestures in the vertical scale to control zooming. | ### View: Track Size Controls the sizes of tracks. | Scripting Id | Action | Parameters | Description | | ------------------ | ----------------------- | ---------- | ----------------------------------------------------------------------------- | | FitInWindow: | Fit to Width | *none* | Zooms out until the entire project just fits in the window. | | FitV: | Fit to Height | *none* | Adjusts the height of all the tracks until they fit in the project window. | | CollapseAllTracks: | Collapse All Tracks | *none* | Collapses all tracks to take up the minimum amount of space. | | ExpandAllTracks: | Expand Collapsed Tracks | *none* | Expands all collapsed tracks to their original size before the last collapse. | ### View: Skip to Move forward/backwards through the audio | Scripting Id | Action | Parameters | Description | | ------------- | --------------- | ---------- | ---------------------------------------------------------------------------------------------------- | | SkipSelStart: | Selection Start | *none* | When there is a selection, moves the cursor to the start of the selection and removes the selection. | | SkipSelEnd: | Selection End | *none* | When there is a selection, moves the cursor to the end of the selection and removes the selection. | ### View: Toolbars Toolbars can be used to determine which of the Audacity toolbars are displayed. By default all toolbars are shown except Spectral Selection and Scrub | Scripting Id | Action | Parameters | Description | | ------------------------ | -------------------------- | ---------- | ---------------------------------------------------------------------------------------------------------------------------------- | | ResetToolbars: | Reset Toolbars | *none* | Using this command positions all toolbars in default location and size as they were when Audacity was first installed | | ShowTransportTB: | Transport Toolbar | *none* | Controls playback and recording and skips to start or end of project when neither playing or recording | | ShowToolsTB: | Tools Toolbar | *none* | Chooses various tools for selection, volume adjustment, zooming and time-shifting of audio | | ShowRecordMeterTB: | Recording Meter Toolbar | *none* | Displays recording levels and toggles input monitoring when not recording | | ShowPlayMeterTB: | Playback Meter Toolbar | *none* | Displays playback levels | | ShowMixerTB: | Mixer Toolbar | *none* | Adjusts the recording and playback volumes of the devices currently selected in Device Toolbar | | ShowEditTB: | Edit Toolbar | *none* | Cut, copy, paste, trim audio, silence audio, undo, redo, zoom tools | | ShowTranscriptionTB: | Play-at-Speed Toolbar | *none* | Plays audio at a slower or faster speed than normal, affecting pitch | | ShowScrubbingTB: | Scrub Toolbar | *none* | Controls playback and recording and skips to start or end of project when neither playing or recording | | ShowDeviceTB: | Device Toolbar | *none* | Selects audio host, recording device, number of recording channels and playback device | | ShowSelectionTB: | Selection Toolbar | *none* | Controls the sample rate of the project, snapping to the selection format and adjusts cursor and region position by keyboard input | | ShowSpectralSelectionTB: | Spectral Selection Toolbar | *none* | Displays and lets you adjust the current spectral (frequency) selection without having to be in Spectrogram view | ## Transport Menu Transport Menu commands let you play or stop, loop play, scrub play or record (including timed and sound activated recordings). | Scripting Id | Action | Parameters | Description | | -------------- | -------------------- | ---------- | --------------------------------------------------------------------------------------------------------------------------- | | RescanDevices: | Rescan Audio Devices | *none* | Rescan for audio devices connected to your computer, and update the playback and recording dropdown menus in Device Toolbar | ### Transport: Playing These commands control playback in Audacity. You can Start, Stop or Pause playback of the audio in your project. | Scripting Id | Action | Parameters | Description | | --------------- | ------------------------ | ---------- | --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- | | PlayStop: | Play/Stop | *none* | Starts and stops playback or stops a recording (stopping does not change the restart position). Therefore using any play or record command after stopping with "Play/Stop" will start playback or recording from the same Timeline position it last started from. You can also assign separate shortcuts for [Play](broken://pages/kGgWQRN8abzBJbdLCXqA) and [Stop](broken://pages/kGgWQRN8abzBJbdLCXqA). | | PlayStopSelect: | Play/Stop and Set Cursor | *none* | Starts playback like "Play/Stop", but stopping playback sets the restart position to the stop point. When stopped, this command is the same as "Play/Stop". When playing, this command stops playback and moves the cursor (or the start of the selection) to the position where playback stopped. | | Pause: | Pause | *none* | Temporarily pauses playing or recording without losing your place. | ### 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 | | ---------------- | --------------------- | ---------- | ------------------------------------------------------------------------------------------------------------------- | | Record1stChoice: | Record | *none* | Starts recording at the end of the currently selected track(s). | | Record2ndChoice: | Record New Track | *none* | Recording begins on a new track at either the current cursor location or at the beginning of the current selection. | | TimerRecord: | Timer Record... | *none* | Brings up the Timer Record dialog. | | PunchAndRoll: | Punch and Roll Record | *none* | Re-record over audio, with a pre-roll of audio that comes before. | | Pause: | Pause | *none* | Temporarily pauses playing or recording without losing your place. | ### 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 | | ----------------- | ----------- | ---------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------ | | Scrub: | Scrub | *none* | 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. | | Seek: | Seek | *none* | Seeking is similar to Scrubbing except that it is playback with skips, similar to using the seek button on a CD player. | | ToggleScrubRuler: | Scrub Ruler | *none* | Shows (or hides) the scrub ruler, which is just below the timeline. | ### 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 | | --------------------- | ---------------------- | ---------- | ----------------------------------------------------------------------------------------------------------- | | CursSelStart: | Selection Start | *none* | Moves the left edge of the current selection to the center of the screen, without changing the zoom level. | | CursSelEnd: | Selection End | *none* | Moves the right edge of the current selection to the center of the screen, without changing the zoom level. | | CursTrackStart: | Track Start | *none* | Moves the cursor to the start of the selected track. | | CursTrackEnd: | Track End | *none* | Moves the cursor to the end of the selected track. | | CursPrevClipBoundary: | Previous Clip Boundary | *none* | Moves the cursor position back to the right-hand edge of the previous clip | | CursNextClipBoundary: | Next Clip Boundary | *none* | Moves the cursor position forward to the left-hand edge of the next clip | | CursProjectStart: | Project Start | *none* | Moves the cursor to the beginning of the project. | | CursProjectEnd: | Project End | *none* | Moves the cursor to the end of the project. | ### Transport: Transport Options This submenu lets you manage and set various options for transport (playing and recording) in Audacity | Scripting Id | Action | Parameters | Description | | --------------------- | ---------------------------------- | ---------- | ------------------------------------------------------------------------------------------------------------------------------------------------- | | SoundActivationLevel: | Sound Activation Level... | *none* | Sets the activation level above which Sound Activated Recording will record. | | SoundActivation: | Sound Activated Recording (on/off) | *none* | Toggles on and off the Sound Activated Recording option. | | PinnedHead: | Pinned Play/Record Head (on/off) | *none* | You can change Audacity to play and record with a fixed head pinned to the Timeline. You can adjust the position of the fixed head by dragging it | | Overdub: | Overdub (on/off) | *none* | Toggles on and off the Overdub option. | | SWPlaythrough: | Software Playthrough (on/off) | *none* | Toggles on and off the Software Playthrough option. | ## Tracks Menu Tracks Menu provides commands for creating and removing tracks, applying operations to selected tracks such as mixing, resampling or converting from stereo to mono, and lets you add or edit labels. | Scripting Id | Action | Parameters | Description | | ------------- | ------------------------- | ---------- | --------------------------------------------------------------------------------------------------------------------------------------- | | Resample: | Resample... | *none* | Allows you to resample the selected track(s) to a new sample rate for use in the project | | RemoveTracks: | Remove Tracks | *none* | Removes the selected track(s) from the project. Even if only part of a track is selected, the entire track is removed. | | SyncLock: | Sync-Lock Tracks (on/off) | *none* | Ensures that length changes occurring anywhere in a defined group of tracks also take place in all audio or label tracks in that group. | ### Tracks: Add New Adds a new track | Scripting Id | Action | Parameters | Description | | --------------- | ------------ | ---------- | ---------------------------------------------------------------------------------------------- | | NewMonoTrack: | Mono Track | *none* | Creates a new empty mono audio track. | | NewStereoTrack: | Stereo Track | *none* | Adds an empty stereo track to the project | | NewLabelTrack: | Label Track | *none* | Adds an empty label track to the project | | NewTimeTrack: | Time Track | *none* | Adds an empty time track to the project. Time tracks are used to speed up and slow down audio. | ### Tracks: Mix Mixes down selected tracks to mono or stereo tracks | Scripting Id | Action | Parameters | Description | | ----------------------- | --------------------------- | ---------- | -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- | | Stereo to Mono: | Mix Stereo Down to Mono | *none* | Converts the selected stereo track(s) into the same number of mono tracks, combining left and right channels equally by averaging the volume of both channels. | | MixAndRender: | Mix and Render | *none* | Mixes down all selected tracks to a single mono or stereo track, rendering to the waveform all real-time transformations that had been applied (such as track *gain*, panning, amplitude envelopes or a change in project rate). | | MixAndRenderToNewTrack: | Mix and Render to New Track | *none* | Same as Tracks > Mix and Render except that the original tracks are preserved rather than being replaced by the resulting "Mix" track. | ### Tracks: Mute/Unmute Mutes or unmutes audio tracks in the project | Scripting Id | Action | Parameters | Description | | ---------------- | ----------------- | ---------- | ------------------------------------------------------------------------------------------------------------------------------- | | MuteAllTracks: | Mute All Tracks | *none* | Mutes all the audio tracks in the project as if you had used the mute buttons from the Track Control Panel on each track. | | UnmuteAllTracks: | Unmute All Tracks | *none* | Unmutes all the audio tracks in the project as if you had released the mute buttons from the Track Control Panel on each track. | | MuteTracks: | Mute Tracks | *none* | Mutes the selected tracks. | | UnmuteTracks: | Unmute Tracks | *none* | Unmutes the selected tracks. | ### Tracks: Pan Pans left right or center audio tracks in the project | Scripting Id | Action | Parameters | Description | | ------------ | ------ | ---------- | ------------------------------------ | | PanLeft: | Left | *none* | Pan selected audio to left speaker | | PanRight: | Right | *none* | Pan selected audio centrally. | | PanCenter: | Center | *none* | Pan selected audio to right speaker. | ### Tracks: Align Tracks Commands that provide an automatic way of aligning selected tracks with the cursor, the selection, or with the start of the project. | Scripting Id | Action | Parameters | Description | | ------------------------ | ----------------------------------- | ---------- | ---------------------------------------------------------------------------------------------------------------- | | Align\_EndToEnd: | Align End to End | *none* | Aligns the selected tracks one after the other, based on their top-to-bottom order in the project window. | | Align\_Together: | Align Together | *none* | Align the selected tracks so that they start at the same (averaged) start time. | | Align\_StartToZero: | Start to Zero | *none* | Aligns the start of selected tracks with the start of the project. | | Align\_StartToSelStart: | Start to Cursor/Selection Start | *none* | Aligns the start of selected tracks with the current cursor position or with the start of the current selection. | | Align\_StartToSelEnd: | Start to Selection End | *none* | Aligns the start of selected tracks with the end of the current selection. | | Align\_EndToSelStart: | End to Cursor/Selection Start | *none* | Aligns the end of selected tracks with the current cursor position or with the start of the current selection. | | Align\_EndToSelEnd: | End to Selection End | *none* | Aligns the end of selected tracks with the end of the current selection. | | MoveSelectionWithTracks: | Move Selection with Tracks (on/off) | *none* | Toggles on/off the selection moving with the realigned tracks, or staying put. | ### Tracks: Sort Tracks 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 | | ------------ | ------------- | ---------- | ----------------------------------- | | SortByTime: | By Start Time | *none* | Sort tracks in order of start time. | | SortByName: | By Name | *none* | Sort tracks in order by name. | ## Generate Menu Generate Menu lets you create audio containing tones, noise or silence. | Scripting Id | Action | Parameters | Description | | ----------------- | -------------- | ---------- | --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- | | ManageGenerators: | Plugin Manager | *none* | Selecting this option from the Effect Menu (or the Generate Menu or Analyze Menu) takes you to a dialog where you can enable or disable particular Effects, Generators and Analyzers in Audacity. Even if you do not add any third-party plugins, you can use this to make the Effect menu shorter or longer as required. For details see Plugin Manager. | | Built-in: | Built-in | *none* | Shows the list of available Audacity built-in effects but only if the user has effects "Grouped by Type" in Effects Preferences. | | Nyquist: | Nyquist | *none* | Shows the list of available Nyquist effects but only if the user has effects "Grouped by Type" in Effects Preferences. | ### 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 | | ------------ | ------------- | ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ | ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- | | Chirp: | Chirp... |double StartFreq, (default:440)
double EndFreq, (default:1320)
double StartAmp, (default:0.8)
double EndAmp, (default:0.1)
enum Waveform, (default:Sine)
- Sine
- Square
- Sawtooth
- Square, no alias
enum Interpolation, (default:Linear)
- Linear
- Logarithmic
string Sequence, (default:audacity)
double Duty Cycle, (default:55)
double Amplitude, (default:0.8)
enum Type, (default:White)
- White
- Pink
- Brownian
double Amplitude, (default:0.8)
double Frequency, (default:440)
double Amplitude, (default:0.8)
enum Waveform, (default:Sine)
- Sine
- Square
- Sawtooth
- Square, no alias
enum Interpolation, (default:Linear)
- Linear
- Logarithmic
int pitch, (default:0)
enum fade, (default:Abrupt)
- Abrupt
- Gradual
double dur, (default:0)
double tempo, (default:0)
int timesig, (default:0)
double swing, (default:0)
int bars, (default:0)
double click-track-dur, (default:0)
double offset, (default:0)
enum click-type, (default:Metronome)
- Metronome
- Ping (short)
- Ping (long)
- Cowbell
- ResonantNoise
- NoiseClick
- Drip (short)
- Drip (long)
int high, (default:0)
int low, (default:0)
double freq, (default:0)
double decay, (default:0)
double cf, (default:0)
double bw, (default:0)
double noise, (default:0)
double gain, (default:0)
float Ratio, (default:0.9)
bool AllowClipping, (default:False)
double DuckAmountDb, (default:-12)
double InnerFadeDownLen, (default:0)
double InnerFadeUpLen, (default:0)
double OuterFadeDownLen, (default:0.5)
double OuterFadeUpLen, (default:0.5)
double ThresholdDb, (default:-30)
double MaximumPause, (default:1)
double Bass, (default:0)
double Treble, (default:0)
double Gain, (default:0)
bool Link Sliders, (default:False)
double Percentage, (default:0)
bool SBSMS, (default:False)
double Percentage, (default:0)
double Percentage, (default:0)
bool SBSMS, (default:False)
int Threshold, (default:200)
int Width, (default:20)
double Threshold, (default:-12)
double NoiseFloor, (default:-40)
double Ratio, (default:2)
double AttackTime, (default:0.2)
double ReleaseTime, (default:1)
bool Normalize, (default:True)
bool UsePeak, (default:False)
enum Type, (default:Hard Clipping)
- Hard Clipping
- Soft Clipping
- Soft Overdrive
- Medium Overdrive
- Hard Overdrive
- Cubic Curve (odd harmonics)
- Even Harmonics
- Expand and Compress
- Leveller
- Rectifier Distortion
- Hard Limiter 1413
bool DC Block, (default:False)
double Threshold dB, (default:-6)
double Noise Floor, (default:-70)
double Parameter 1, (default:50)
double Parameter 2, (default:50)
int Repeats, (default:1)
float Delay, (default:1)
float Decay, (default:0.5)
Applies a linear fade-out to the selected audio - the rapidity of the fade-out depends entirely on the length of the selection it is applied to. For a more customizable logarithmic fade, use the Envelope Tool on the Tools Toolbar.
size\_t FilterLength, (default:8191)
bool InterpolateLin, (default:False)
enum InterpolationMethod, (default:B-spline)
- B-spline
- Cosine
- Cubic
double f0, (default:0)
double v0, (default:0)
size\_t FilterLength, (default:8191)
bool InterpolateLin, (default:False)
enum InterpolationMethod, (default:B-spline)
- B-spline
- Cosine
- Cubic
double f0, (default:0)
double v0, (default:0)
bool StereoIndependent, (default:False)
double LUFSLevel, (default:-23)
double RMSLevel, (default:-20)
bool DualMono, (default:True)
int NormalizeTo, (default:0)
This effect is ideal for reducing constant background noise such as fans, tape noise, or hums. It will not work very well for removing talking or music in the background. More details here
This effect is not currently available from scripting.
double PeakLevel, (default:-1)
bool ApplyGain, (default:True)
bool RemoveDcOffset, (default:True)
bool StereoIndependent, (default:False)
float Stretch Factor, (default:10)
float Time Resolution, (default:0.25)
int Stages, (default:2)
int DryWet, (default:128)
double Freq, (default:0.4)
double Phase, (default:0)
int Depth, (default:100)
int Feedback, (default:0)
double Gain, (default:-6)
int Count, (default:1)
double RoomSize, (default:75)
double Delay, (default:10)
double Reverberance, (default:50)
double HfDamping, (default:50)
double ToneLow, (default:100)
double ToneHigh, (default:100)
double WetGain, (default:-1)
double DryGain, (default:-1)
double StereoWidth, (default:100)
bool WetOnly, (default:False)
double RatePercentChangeStart, (default:0)
double RatePercentChangeEnd, (default:0)
double PitchHalfStepsStart, (default:0)
double PitchHalfStepsEnd, (default:0)
double PitchPercentChangeStart, (default:0)
double PitchPercentChangeEnd, (default:0)
double Threshold, (default:-20)
enum Action, (default:Truncate Detected Silence)
- Truncate Detected Silence
- Compress Excess Silence
double Minimum, (default:0.5)
double Truncate, (default:0.5)
double Compress, (default:50)
bool Independent, (default:False)
double Freq, (default:1.5)
double Phase, (default:0)
int Depth, (default:70)
double Resonance, (default:2.5)
int Offset, (default:30)
double Gain, (default:-6)
enum type, (default:Up)
- Up
- Down
- SCurveUp
- SCurveDown
double curve, (default:0)
enum units, (default:Percent)
- Percent
- dB
double gain0, (default:0)
double gain1, (default:0)
enum preset, (default:None)
- None
- LinearIn
- LinearOut
- ExponentialIn
- ExponentialOut
- LogarithmicIn
- LogarithmicOut
- RoundedIn
- RoundedOut
- CosineIn
- CosineOut
- SCurveIn
- SCurveOut
double threshold, (default:0)
double gain, (default:0)
enum type, (default:ConstantGain)
- ConstantGain
- ConstantPower1
- ConstantPower2
- CustomCurve
double curve, (default:0)
enum direction, (default:Automatic)
- Automatic
- OutIn
- InOut
enum delay-type, (default:Regular)
- Regular
- BouncingBall
- ReverseBouncingBall
double dgain, (default:0)
double delay, (default:0)
enum pitch-type, (default:PitchTempo)
- PitchTempo
- LQPitchShift
double shift, (default:0)
int number, (default:0)
enum constrain, (default:Yes)
- Yes
- No
double frequency, (default:0)
enum rolloff, (default:dB6)
- dB6
- dB12
- dB24
- dB36
- dB48
enum type, (default:SoftLimit)
- SoftLimit
- HardLimit
- SoftClip
- HardClip
double gain-L, (default:0)
double gain-R, (default:0)
double thresh, (default:0)
double hold, (default:0)
enum makeup, (default:No)
- No
- Yes
double frequency, (default:0)
enum rolloff, (default:dB6)
- dB6
- dB12
- dB24
- dB36
- dB48
double frequency, (default:0)
double q, (default:0)
double control-gain, (default:0)
double control-gain, (default:0)
enum wave, (default:Sine)
- Sine
- Triangle
- Sawtooth
- InverseSawtooth
- Square
int phase, (default:0)
int wet, (default:0)
double lfo, (default:0)
enum action, (default:RemoveToMono)
- RemoveToMono
- Remove
- Isolate
- IsolateInvert
- RemoveCenterToMono
- RemoveCenter
- IsolateCenter
- IsolateCenterInvert
- Analyze
double strength, (default:0)
double low-transition, (default:0)
double high-transition, (default:0)
double dst, (default:0)
enum mst, (default:BothChannels)
- BothChannels
- RightOnly
int bands, (default:0)
double track-vl, (default:0)
double noise-vl, (default:0)
double radar-vl, (default:0)
double radar-f, (default:0)
int Duty Cycle Start, (default:3)
int Duty Cycle End, (default:3)
int thresval, (default:0)
double threshold, (default:0)
enum measurement, (default:peak)
- peak
- avg
- rms
double sil-dur, (default:0)
double snd-dur, (default:0)
enum type, (default:before)
- before
- after
- around
- between
double pre-offset, (default:0)
double post-offset, (default:0)
string text, (default:"")
string Path, (default:)
enum CaptureWhat, (default:Window)
- Window
- FullWindow
- WindowPlus
- Fullscreen
- Toolbars
- Effects
- Scriptables
- Preferences
- Selectionbar
- SpectralSelection
- Timer
- Tools
- Transport
- Mixer
- Meter
- PlayMeter
- RecordMeter
- Edit
- Device
- Scrub
- Play-at-Speed
- Trackpanel
- Ruler
- Tracks
- FirstTrack
- FirstTwoTracks
- FirstThreeTracks
- FirstFourTracks
- SecondTrack
- TracksPlus
- FirstTrackPlus
- AllTracks
- AllTracksPlus
enum Background, (default:None)
- Blue
- White
- None
bool ToTop, (default:True)
string Command, (default:)
int Version, (default:3)
string files, (default:)
enum overwrite, (default:Disallow)
- Disallow
- Allow
enum mode, (default:Both)
- Both
- Number
- Interval
int totalnum, (default:0)
double interval, (default:0)
double region, (default:0)
enum adjust, (default:No)
- No
- Yes
string labeltext, (default:)
enum zeros, (default:TextOnly)
- TextOnly
- OneBefore
- TwoBefore
- ThreeBefore
- OneAfter
- TwoAfter
- ThreeAfter
int firstnum, (default:0)
enum verbose, (default:Details)
- Details
- Warnings
- None
int number, (default:0)
enum units, (default:dB)
- dB
- Linear
string filename, (default:)
enum fileformat, (default:None)
- None
- Count
- Time
enum header, (default:None)
- None
- Minimal
- Standard
- All
string optext, (default:)
enum channel-layout, (default:SameLine)
- SameLine
- Alternate
- LFirst
enum messages, (default:Yes)
- Yes
- Errors
- None
string filename, (default:)
enum bad-data, (default:ThrowError)
- ThrowError
- ReadAsZero
double Start, (default:unchanged)
double End, (default:unchanged)
enum RelativeTo, (default:unchanged)
- ProjectStart
- Project
- ProjectEnd
- SelectionStart
- Selection
- SelectionEnd
double High, (default:unchanged)
double Low, (default:unchanged)
double Track, (default:unchanged)
double TrackCount, (default:unchanged)
enum Mode, (default:unchanged)
- Set
- Add
- Remove
string Name, (default:unchanged)
bool Selected, (default:unchanged)
bool Focused, (default:unchanged)
bool Mute, (default:unchanged)
bool Solo, (default:unchanged)
double Gain, (default:unchanged)
double Pan, (default:unchanged)
int Height, (default:unchanged)
enum Display, (default:unchanged)
- Waveform
- Spectrogram
- Multi-view
enum Scale, (default:unchanged)
- Linear
- dB
enum Color, (default:unchanged)
- Color0
- Color1
- Color2
- Color3
enum VZoom, (default:unchanged)
- Reset
- Times2
- HalfWave
double VZoomHigh, (default:unchanged)
double VZoomLow, (default:unchanged)
bool SpecPrefs, (default:unchanged)
bool SpectralSel, (default:unchanged)
enum Scheme, (default:unchanged)
- Color (default)
- Color (classic)
- Grayscale
- Inverse Grayscale
string Name, (default:)
string Name, (default:)
string Value, (default:)
bool Reload, (default:False)
double At, (default:unchanged)
enum Color, (default:unchanged)
- Color0
- Color1
- Color2
- Color3
double Start, (default:unchanged)
double Time, (default:unchanged)
double Value, (default:unchanged)
bool Delete, (default:unchanged)
int Label, (default:0)
string Text, (default:unchanged)
double Start, (default:unchanged)
double End, (default:unchanged)
bool Selected, (default:unchanged)
string Name, (default:unchanged)
double Rate, (default:unchanged)
int X, (default:unchanged)
int Y, (default:unchanged)
int Width, (default:unchanged)
int Height, (default:unchanged)
double Start, (default:unchanged)
double End, (default:unchanged)
enum RelativeTo, (default:unchanged)
- ProjectStart
- Project
- ProjectEnd
- SelectionStart
- Selection
- SelectionEnd
double High, (default:unchanged)
double Low, (default:unchanged)
double Track, (default:unchanged)
double TrackCount, (default:unchanged)
enum Mode, (default:unchanged)
- Set
- Add
- Remove
string Name, (default:unchanged)
bool Selected, (default:unchanged)
bool Focused, (default:unchanged)
bool Mute, (default:unchanged)
bool Solo, (default:unchanged)
double Gain, (default:unchanged)
double Pan, (default:unchanged)
int Height, (default:unchanged)
enum Display, (default:unchanged)
- Waveform
- Spectrogram
enum Scale, (default:unchanged)
- Linear
- dB
enum Color, (default:unchanged)
- Color0
- Color1
- Color2
- Color3
enum VZoom, (default:unchanged)
- Reset
- Times2
- HalfWave
double VZoomHigh, (default:unchanged)
double VZoomLow, (default:unchanged)
bool SpecPrefs, (default:unchanged)
bool SpectralSel, (default:unchanged)
bool GrayScale, (default:unchanged)
enum Type, (default:Commands)
- Commands
- Menus
- Preferences
- Tracks
- Clips
- Envelopes
- Labels
- Boxes
enum Format, (default:JSON)
- JSON
- LISP
- Brief
string Text, (default:Some message)
string Command, (default:Help)
enum Format, (default:JSON)
- JSON
- LISP
- Brief
string Filename, (default:)
string Filename, (default:exported.wav)
int NumChannels, (default:1)
string Filename, (default:test.aup3)
bool AddToHistory, (default:false)
string Filename, (default:name.aup3)
bool AddToHistory, (default:False)
bool Compress, (default:False)
int Id, (default:unchanged)
string Window, (default:unchanged)
double FromX, (default:unchanged)
double FromY, (default:unchanged)
double ToX, (default:unchanged)
double ToY, (default:unchanged)
enum RelativeTo, (default:unchanged)
- Panel
- App
- Track0
- Track1
float Threshold, (default:0)
Screenshot
Figure 1. Bass Boost curve: 3dB at 500 Hz
Figure 2. Treble Cut curve: -13.7 dB at 10 kHz
Figure 3. Combined Bass Boost and Treble Cut curve
Figure 4. Combined Bass Boost, Treble Cut and 50 Hz LF Shelving curve
@ 50Hz
\[dB]
@10kHz
\[dB]
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. \[5] 250 Hz or 300 Hz seem to be a question of personal listening preference, as explained in [Individual fine tuning](https://github.com/audacity/audacity-support/blob/plugins/additional-resources/eq-curves/broken-reference/README.md) \[6] Robert C. Moyer, Evolution of a Recording Curve; in: Audio Engineering, vol.37, no.7, July 1953; pp 19-22, 53-54. Roll-off is frequently listed as between 10 and 12 dB, but this “flattening off” to ca 10 dB at 10 kHz was an intentional high cut which must not be compensated for in playback. Thus a setting of 13.7 dB is correct!| EQ Curve file | Description |
|---|---|
| /files/hKGDCtTybM57vHXL4Ncl | 500-FLAT 500N-0 was used by British Columbia, EMI, His Master’s Voice, MGM and Parlophone between 1931 and 1953. Later releases have modified treble. |
| /files/n8t9CdYHJYHjTmtMLt4p | American 78 – 250N-6/250N-8: Common setting for many American shellacs. This curve here is a compromise between both varieties with -7 dB roll-off. |
| /files/1NORIqNgRYx7XViwhoM5 | BBC 2dB/octave: used by smaller British labels (Aco, Broadcast, Linguaphone, Vocalion, …) from 1926 to 1933, which had their recordings mastered by BBC with the Marconi system. |
| /files/90VmSN9sD1rAzfn94Ulq | Blumlein300 - 300N-0: A British traditional for Gramophone Company, Decca, Columbia and EMI (1930s – 1944). |
| /files/iXvNqfmYK1wixuEy9sc6 | Columbia 78 – 300N-16 is the right one for American CBS-Columbia shellacs (1938 - 1948). |
| /files/1hngKyG3gIbAPp44xsnA | Decca 78 – 300N-5.5: For Decca and London shellacs featuring the ffrr (full frequency range recording) system. Here in the version of Copeland/The British Library Sound Archive |
| /files/scxKmk1CZqfVQL4hvceL | European 78 - 250N-0 is a common setting for European shellacs (1926 – ca. 1944), especially for Columbia and His Master’s Voice produced by EMI (UK), Cetra and Cetra-Soria. |
| /files/Wnovg51vl1URl1AhMPEW | Telefunken 400N-0: used by European Ultraphon, Supraphon and Turicaphon from 1929. Also used by Telefunken – after the takeover of Ultraphon – until mid 1950s. |
| /files/iT3DLziOhUtGV4jds35B | Western Electric: Very early Columbia and Victor recordings (1926) used a bass turnover frequency of 250 or 300 Hz but their treble is described as “flat”. The perceived treble amplification was possibly only the result of resonant peaks of the early Western Electric condenser microphones used in recording. The above download is an experimental replay EQ curve for this microphone / pre-emphasis combination. Additional background information is given in this PDF. |
| EQ curve files | Description |
|---|---|
| /files/s6xY3CRTNgtgdiELY7wB | AES – 400N-12.3: Intended by AES (Audio Engineering Society) as a replay standard for many American shellacs of the 1930s and 1940s. Also used by many record producers as a recording curve for N78 and M33 between 1951 and 1958. Also to replay Capitol and Capitol-Cetra recordings with “Capitol curve” 400N-12.7 (1951 – 1955). |
| /files/JulTcUhEzXKBe7VQciWA | Columbia LP – 500C-16: For Columbia and many other labels, mostly 33⅓ LPs (M33). |
| /files/1KKNn2hJYkFNmDh5T8fz | London LP – 500C-10.5: Used for British Decca and for London / Decca releases in the US featuring the ffrr (full frequency range recording) system. Mostly 33⅓ LPs (1949 – 1956) |
| /files/fjvJ8H04OZUBjDttxoWM | NAB – 500B-16: A widely adopted standard of NAB / NARTB (National Association of Radio and Television Broadcasters) requiring ca. 6 dB more bass boost than Columbia LP |
| /files/vL3cAtBXL2gyqluZvCMV | RCA 45 – 800N-13.7: RCA’s curve for their 45 rpm discs (1949 – August 1952). Possibly identical with Technichord’s “Technichord curve” 800N-12 already used since 1938 for their 78s. |
| /files/KtYHlVO5HhxtMAq2sLGB | RIAA = RCA New Orthophonic – 500R-13.7 |
| /files/FCNUAFbjBcDIlqi0CMBW | RCA Old Orthophonic – 500N-13.7: RCA’s curve for 33⅓ LPs (1950 – August 1952) and for LPs mastered by RCA for other labels. Possibly identical with “MGM curve” 500N-12 used by MGM. |
turnover
bass \[Hz]
bass
shelf
treble roll-off
\[dB @ 10kHz]
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! * \[MGM] ... used a special “MGM curve” 500N-12. Play back with “RCA Old Orthophonic” 500N-13.7! * \[R-D] ... Roll-off is frequently listed as between 10 and 12 dB. But this “flattening off” to ca 10 dB at 10 kHz was an intentional high cut which must not be compensated for in playback. Thus a setting of 13.7 dB is correct! * \[R-C] ... RCA recommends replaying with “Old Ortho.”. High frequencies were cut off more deliberately at 8,500 Hz (with no effect on playback EQ!) than in later years. \[cf. Moyer] * \[R-B] ... From c. 1931 to 1938 high frequencies were even cut off at 5,500 Hz! The bass curve is “subject to some question, however,” since it was extensively modified by electronic filters and mechanical damping. But RCA found 500 Hz to be the best average characteristic and used this in re-recording pre-1938 masters. So in principle: “Old Ortho.” for playback again! \[cf. Moyer] * \[TCH] ... Technichord used its own “Technichord curve” 800N-12. Play back with “RCA 45” 800N-13.7! * \[W\.E.] ... Very early Columbia and Victor recordings (1926) used a bass turnover frequency of 250 or 300 Hz but their treble is described as “flat”. The perceived treble amplification was possibly only the result of resonant peaks of the early Western Electric condenser microphones used in recording. Background information is given in [this PDF](http://audacity.xmission.eu/Early_Western_Electric_Characteristic.pdf). An experimental replay EQ curve for this microphone / pre-emphasis combination can be downloaded here.turnover
bass \[Hz]
bass
shelf
treble roll-off
\[dB @ 10kHz]