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 tap

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

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

• 12 early reflections variations

• 5 reverb tail variations

• 3-band equalizer for the wet signal

• Sample Rate: from 44100 to 192000 Hz.

• Lots of presets

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.

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.

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 support

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.

"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 effect

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

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

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 graph

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 comparisons

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 gain

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.

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

Vendor description

1950 Meets 2019

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.

© 2011-2022 AudioThing Ltd.

Description from the vendor:

Filterjam is a multi-band resonant filter delivering weird ringmod-like filtered sounds. The input signal is divided into 4 bands that are then summed or multiplied together according to the selected mode. Filterjam can be very harsh or gentle, it can add brightness or depth to synth sounds, but it can also completely mangle acoustic sources.

Specifications

• Multi-Band Resonant Filter

• Oversample up to 16x

• Lightweight on CPU

• Preset system with randomizer

System Requirements

Windows 7, 8, 10 2GHz CPU, 4 GB RAM VST2, VST3, AAX, CLAP (64-bit)

OS X 10.9 – macOS 12 2GHz CPU, 4 GB RAM VST2, VST3, AU, AAX, CLAP (64-bit) Universal 2 Binary

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.

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 Range

• 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

• 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

© 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

\

Earth Shattering Kaboom.

ValhallaSpaceModulator is our take on flanging.

Eleven algorithms allow you to get through-zero flanging, barberpole flanging, detuning, doubling, strange echoes, reverbs, and all sorts of effects that defy description.

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

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 signal

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)

Copyright © 2022 NVIDIA Corporation

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

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

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]

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]

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]

Author: Steve Daulton

HQ-Tone is a high quality (no alias) tone generator.

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.

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.

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]

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]

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."

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}

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)

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.

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]

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.\

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.

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 tones

Author: Steven Jones.

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.

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 signal

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

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)

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.

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]

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

Parameters:

1. Max amp sweep speed: [0.01 - 20.0 Hz, default 0.5] - maximum speed of the random amplitude changes

2. Amp sweep depth factor: [1 - 300, default 80] - how extreme the random amplitude changes are

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]

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.

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.

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.

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.

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.

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".

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.

Author: Steve Daulton

Usage:

Type

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.

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.

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.

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.

Author: Steve Daulton.

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)]

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.

Author: Steve Daulton

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.

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.

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.

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.

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]

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]

Author: Steve Daulton

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

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.

Parameters:

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

2. Filter Q (resonance): [0 - 5, default 1]

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]

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]

Author: Steve Daulton

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.

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]

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]

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

2. Filter q (resonance): [0 - 5, default 1]

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.

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 Pass

Authors: Steve Daulton and Bill Wharrie.

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.

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.

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 frequency

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.

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.

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

Vocal Reduction and Isolation 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.

Authors: Will McCown, Steve Daulton, Philip Collier

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

Displayed Results

• Peak level Maximum peak level in the selection

• RMS level The RMS level of the selected audio.

• Noise floor 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.

Author: Steve Daulton

Based on a plugin by Will McCown.

Displayed Results

• Peak level Maximum peak level in the selection

• RMS level The RMS level of the selected audio.

• Noise floor 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.

©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!

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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.

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.

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]

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]

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

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 (%):

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.

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.

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.

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]

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)]

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

Parameters:

1. LFO frequency: [Hz, 0.02 - 20, default 0.1]

2. LFO waveform: [sine, triangle, saw, inverted saw, pulse]

3. Pulse waveform duty cycle: [percent, default 50]

4. LFO starting phase: [degrees, -180 - +180, default 0]

5. Leftmost pan position: [percent, default 5] - 0%=left channel, 50%=center, 100%=right channel

6. Rightmost pan position: [percent, default 95] - 0%=left channel, 50%=center, 100%=right channel

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.

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.